diff --git a/sflphone-gtk/src/configurationmanager-glue.h b/sflphone-gtk/src/configurationmanager-glue.h
index ba2d46b61d762461b715f06ee93ec4d07afd6ede..1f3810ccc8a47fdc5c9ef8fb363a27dd39823a32 100644
--- a/sflphone-gtk/src/configurationmanager-glue.h
+++ b/sflphone-gtk/src/configurationmanager-glue.h
@@ -1,7 +1,6 @@
/* Generated by dbus-binding-tool; do not edit! */
-#include <glib/gtypes.h>
-#include <glib/gerror.h>
+#include <glib.h>
#include <dbus/dbus-glib.h>
G_BEGIN_DECLS
@@ -2001,6 +2000,156 @@ org_sflphone_SFLphone_ConfigurationManager_get_sip_port_async (DBusGProxy *proxy
stuff->userdata = userdata;
return dbus_g_proxy_begin_call (proxy, "getSipPort", org_sflphone_SFLphone_ConfigurationManager_get_sip_port_async_callback, stuff, g_free, G_TYPE_INVALID);
}
+static
+#ifdef G_HAVE_INLINE
+inline
+#endif
+gboolean
+org_sflphone_SFLphone_ConfigurationManager_set_stun_server (DBusGProxy *proxy, const char * IN_server, GError **error)
+
+{
+ return dbus_g_proxy_call (proxy, "setStunServer", error, G_TYPE_STRING, IN_server, G_TYPE_INVALID, G_TYPE_INVALID);
+}
+
+typedef void (*org_sflphone_SFLphone_ConfigurationManager_set_stun_server_reply) (DBusGProxy *proxy, GError *error, gpointer userdata);
+
+static void
+org_sflphone_SFLphone_ConfigurationManager_set_stun_server_async_callback (DBusGProxy *proxy, DBusGProxyCall *call, void *user_data)
+{
+ DBusGAsyncData *data = (DBusGAsyncData*) user_data;
+ GError *error = NULL;
+ dbus_g_proxy_end_call (proxy, call, &error, G_TYPE_INVALID);
+ (*(org_sflphone_SFLphone_ConfigurationManager_set_stun_server_reply)data->cb) (proxy, error, data->userdata);
+ return;
+}
+
+static
+#ifdef G_HAVE_INLINE
+inline
+#endif
+DBusGProxyCall*
+org_sflphone_SFLphone_ConfigurationManager_set_stun_server_async (DBusGProxy *proxy, const char * IN_server, org_sflphone_SFLphone_ConfigurationManager_set_stun_server_reply callback, gpointer userdata)
+
+{
+ DBusGAsyncData *stuff;
+ stuff = g_new (DBusGAsyncData, 1);
+ stuff->cb = G_CALLBACK (callback);
+ stuff->userdata = userdata;
+ return dbus_g_proxy_begin_call (proxy, "setStunServer", org_sflphone_SFLphone_ConfigurationManager_set_stun_server_async_callback, stuff, g_free, G_TYPE_STRING, IN_server, G_TYPE_INVALID);
+}
+static
+#ifdef G_HAVE_INLINE
+inline
+#endif
+gboolean
+org_sflphone_SFLphone_ConfigurationManager_get_stun_server (DBusGProxy *proxy, char ** OUT_server, GError **error)
+
+{
+ return dbus_g_proxy_call (proxy, "getStunServer", error, G_TYPE_INVALID, G_TYPE_STRING, OUT_server, G_TYPE_INVALID);
+}
+
+typedef void (*org_sflphone_SFLphone_ConfigurationManager_get_stun_server_reply) (DBusGProxy *proxy, char * OUT_server, GError *error, gpointer userdata);
+
+static void
+org_sflphone_SFLphone_ConfigurationManager_get_stun_server_async_callback (DBusGProxy *proxy, DBusGProxyCall *call, void *user_data)
+{
+ DBusGAsyncData *data = (DBusGAsyncData*) user_data;
+ GError *error = NULL;
+ char * OUT_server;
+ dbus_g_proxy_end_call (proxy, call, &error, G_TYPE_STRING, &OUT_server, G_TYPE_INVALID);
+ (*(org_sflphone_SFLphone_ConfigurationManager_get_stun_server_reply)data->cb) (proxy, OUT_server, error, data->userdata);
+ return;
+}
+
+static
+#ifdef G_HAVE_INLINE
+inline
+#endif
+DBusGProxyCall*
+org_sflphone_SFLphone_ConfigurationManager_get_stun_server_async (DBusGProxy *proxy, org_sflphone_SFLphone_ConfigurationManager_get_stun_server_reply callback, gpointer userdata)
+
+{
+ DBusGAsyncData *stuff;
+ stuff = g_new (DBusGAsyncData, 1);
+ stuff->cb = G_CALLBACK (callback);
+ stuff->userdata = userdata;
+ return dbus_g_proxy_begin_call (proxy, "getStunServer", org_sflphone_SFLphone_ConfigurationManager_get_stun_server_async_callback, stuff, g_free, G_TYPE_INVALID);
+}
+static
+#ifdef G_HAVE_INLINE
+inline
+#endif
+gboolean
+org_sflphone_SFLphone_ConfigurationManager_enable_stun (DBusGProxy *proxy, GError **error)
+
+{
+ return dbus_g_proxy_call (proxy, "enableStun", error, G_TYPE_INVALID, G_TYPE_INVALID);
+}
+
+typedef void (*org_sflphone_SFLphone_ConfigurationManager_enable_stun_reply) (DBusGProxy *proxy, GError *error, gpointer userdata);
+
+static void
+org_sflphone_SFLphone_ConfigurationManager_enable_stun_async_callback (DBusGProxy *proxy, DBusGProxyCall *call, void *user_data)
+{
+ DBusGAsyncData *data = (DBusGAsyncData*) user_data;
+ GError *error = NULL;
+ dbus_g_proxy_end_call (proxy, call, &error, G_TYPE_INVALID);
+ (*(org_sflphone_SFLphone_ConfigurationManager_enable_stun_reply)data->cb) (proxy, error, data->userdata);
+ return;
+}
+
+static
+#ifdef G_HAVE_INLINE
+inline
+#endif
+DBusGProxyCall*
+org_sflphone_SFLphone_ConfigurationManager_enable_stun_async (DBusGProxy *proxy, org_sflphone_SFLphone_ConfigurationManager_enable_stun_reply callback, gpointer userdata)
+
+{
+ DBusGAsyncData *stuff;
+ stuff = g_new (DBusGAsyncData, 1);
+ stuff->cb = G_CALLBACK (callback);
+ stuff->userdata = userdata;
+ return dbus_g_proxy_begin_call (proxy, "enableStun", org_sflphone_SFLphone_ConfigurationManager_enable_stun_async_callback, stuff, g_free, G_TYPE_INVALID);
+}
+static
+#ifdef G_HAVE_INLINE
+inline
+#endif
+gboolean
+org_sflphone_SFLphone_ConfigurationManager_is_stun_enabled (DBusGProxy *proxy, gint* OUT_state, GError **error)
+
+{
+ return dbus_g_proxy_call (proxy, "isStunEnabled", error, G_TYPE_INVALID, G_TYPE_INT, OUT_state, G_TYPE_INVALID);
+}
+
+typedef void (*org_sflphone_SFLphone_ConfigurationManager_is_stun_enabled_reply) (DBusGProxy *proxy, gint OUT_state, GError *error, gpointer userdata);
+
+static void
+org_sflphone_SFLphone_ConfigurationManager_is_stun_enabled_async_callback (DBusGProxy *proxy, DBusGProxyCall *call, void *user_data)
+{
+ DBusGAsyncData *data = (DBusGAsyncData*) user_data;
+ GError *error = NULL;
+ gint OUT_state;
+ dbus_g_proxy_end_call (proxy, call, &error, G_TYPE_INT, &OUT_state, G_TYPE_INVALID);
+ (*(org_sflphone_SFLphone_ConfigurationManager_is_stun_enabled_reply)data->cb) (proxy, OUT_state, error, data->userdata);
+ return;
+}
+
+static
+#ifdef G_HAVE_INLINE
+inline
+#endif
+DBusGProxyCall*
+org_sflphone_SFLphone_ConfigurationManager_is_stun_enabled_async (DBusGProxy *proxy, org_sflphone_SFLphone_ConfigurationManager_is_stun_enabled_reply callback, gpointer userdata)
+
+{
+ DBusGAsyncData *stuff;
+ stuff = g_new (DBusGAsyncData, 1);
+ stuff->cb = G_CALLBACK (callback);
+ stuff->userdata = userdata;
+ return dbus_g_proxy_begin_call (proxy, "isStunEnabled", org_sflphone_SFLphone_ConfigurationManager_is_stun_enabled_async_callback, stuff, g_free, G_TYPE_INVALID);
+}
#endif /* defined DBUS_GLIB_CLIENT_WRAPPERS_org_sflphone_SFLphone_ConfigurationManager */
G_END_DECLS
diff --git a/sflphone-gtk/src/configwindow.c b/sflphone-gtk/src/configwindow.c
index be137c73cf4bc7c7e1f9257351c5e5c3eacc656e..a15290e3bfa511942377ca14416d7bd5397fe148 100644
--- a/sflphone-gtk/src/configwindow.c
+++ b/sflphone-gtk/src/configwindow.c
@@ -109,7 +109,7 @@ config_window_fill_account_list()
* Delete an account
*/
static void
-delete_account(GtkWidget *widget, gpointer data UNUSED)
+delete_account(GtkWidget *widget UNUSED, gpointer data UNUSED)
{
if(selectedAccount)
{
@@ -449,24 +449,15 @@ create_accounts_tab()
void stun_state( void )
{
- guint i, size;
- gchar * stun_enabled = "FALSE";
- account_t * account;
+ guint stun_enabled = 0;
gboolean stunActive = (gboolean)gtk_toggle_button_get_active( GTK_TOGGLE_BUTTON( stunEnable ));
gtk_widget_set_sensitive( GTK_WIDGET( stunServer ) , stunActive );
+
// Check if we actually change the state
- size = account_list_get_size();
- for(i=0; i<size; i++)
- {
- account = account_list_get_nth(i);
- if( strcmp(g_hash_table_lookup(account->properties, ACCOUNT_TYPE), "SIP" ) == 0 )
- {
- stun_enabled = g_hash_table_lookup(account->properties, ACCOUNT_SIP_STUN_ENABLED);
- break;
- }
- }
- if( (stunActive && strcmp(stun_enabled, "FALSE")==0) || (!stunActive && strcmp(stun_enabled, "TRUE")==0) )
+ stun_enabled = dbus_stun_is_enabled();
+
+ if( (stunActive && stun_enabled ==0 ) || (!stunActive && stun_enabled ==1))
{
gtk_widget_set_sensitive( GTK_WIDGET( applyButton ) , TRUE );
}
@@ -477,24 +468,8 @@ void stun_state( void )
void update_registration( void )
{
- guint nb_accounts, i;
- account_t *current;
-
- nb_accounts = account_list_get_size();
-
- for(i=0; i<nb_accounts;i++){
-
- current = account_list_get_nth(i);
- // If SIP account, then set the new value
- if( strcmp(g_hash_table_lookup(current->properties, ACCOUNT_TYPE), "SIP" ) == 0 )
- {
- g_hash_table_replace(current->properties, g_strdup(ACCOUNT_SIP_STUN_SERVER),
- g_strdup((gchar *)gtk_entry_get_text(GTK_ENTRY(stunServer))));
- g_hash_table_replace(current->properties, g_strdup(ACCOUNT_SIP_STUN_ENABLED),
- g_strdup(gtk_toggle_button_get_active(GTK_TOGGLE_BUTTON(stunEnable)) ? "TRUE": "FALSE"));
- dbus_set_account_details(current);
- }
- }
+ dbus_set_stun_server((gchar *)gtk_entry_get_text(GTK_ENTRY(stunServer)));
+ dbus_enable_stun();
gtk_widget_set_sensitive(GTK_WIDGET( applyButton ) , FALSE );
}
@@ -505,23 +480,11 @@ GtkWidget* create_stun_tab()
gchar * stun_server= "stun.fwdnet.net:3478";
gchar * stun_enabled = "FALSE";
GtkWidget * label;
- account_t * account;
- guint i, size;
- // All SIP accounts are supposed to have the same STUN configuration
- // So let's take the first SIP account we find
- size = account_list_get_size();
- for(i=0; i<size; i++)
- {
- account = account_list_get_nth(i);
- if( strcmp(g_hash_table_lookup(account->properties, ACCOUNT_TYPE), "SIP" ) == 0 )
- {
- stun_enabled = g_hash_table_lookup(account->properties, ACCOUNT_SIP_STUN_ENABLED);
- stun_server = g_hash_table_lookup(account->properties, ACCOUNT_SIP_STUN_SERVER);
- break;
- }
- }
-
+ /* Retrieve the STUN configuration */
+ stun_enabled = (dbus_stun_is_enabled()==1)?"TRUE":"FALSE";
+ stun_server = dbus_get_stun_server();
+
tableNat = gtk_table_new ( 3, 2 , FALSE/* homogeneous */);
// NAT detection code section
diff --git a/sflphone-gtk/src/dbus.c b/sflphone-gtk/src/dbus.c
index 7ed21b2118a4241b4dffd5ef73e120a12848ffe4..aa2edb87b9cb287bc1924d1fc5f0ba3a901321b4 100644
--- a/sflphone-gtk/src/dbus.c
+++ b/sflphone-gtk/src/dbus.c
@@ -39,6 +39,7 @@ DBusGConnection * connection;
DBusGProxy * callManagerProxy;
DBusGProxy * configurationManagerProxy;
DBusGProxy * instanceProxy;
+DBusGProxy * nameOwnerProxy;
static void
incoming_call_cb (DBusGProxy *proxy UNUSED,
@@ -172,18 +173,32 @@ error_alert(DBusGProxy *proxy UNUSED,
sflphone_throw_exception( errCode );
}
+
+static void nameOwnerChanged(DBusGProxy *proxy, char *name, char *old_owner, char *new_owner, gpointer data )
+{
+
+ g_print("******************************************************************\n");
+ g_print("Owner name of the service %s changed from %s to %s\n", name, old_owner, new_owner);
+ g_print("******************************************************************\n");
+
+ if (strcmp(name, "org.sflphone.SFLphone")!=0) return;
+
+}
+
gboolean
dbus_connect ()
{
GError *error = NULL;
connection = NULL;
+ instanceProxy = NULL;
+ nameOwnerProxy = NULL;
g_type_init ();
connection = dbus_g_bus_get (DBUS_BUS_SESSION, &error);
- if (connection == NULL)
+ if (error)
{
g_printerr ("Failed to open connection to bus: %s\n",
error->message);
@@ -191,13 +206,32 @@ dbus_connect ()
return FALSE;
}
+
+ nameOwnerProxy = dbus_g_proxy_new_for_name( connection,
+ DBUS_SERVICE_DBUS,
+ DBUS_PATH_DBUS,
+ DBUS_INTERFACE_DBUS);
+
+ if( nameOwnerProxy==NULL)
+ {
+ g_printerr ("Failed to get proxy to NameOwner\n");
+ return FALSE;
+ }
+
+ dbus_g_proxy_add_signal( nameOwnerProxy, "NameOwnerChanged",
+ G_TYPE_STRING, G_TYPE_STRING, G_TYPE_STRING, G_TYPE_INVALID);
+ dbus_g_proxy_connect_signal (nameOwnerProxy, "NameOwnerChanged",
+ G_CALLBACK (nameOwnerChanged), NULL, NULL);
+
+
/* Create a proxy object for the "bus driver" (name "org.freedesktop.DBus") */
instanceProxy = dbus_g_proxy_new_for_name (connection,
"org.sflphone.SFLphone",
"/org/sflphone/SFLphone/Instance",
"org.sflphone.SFLphone.Instance");
- if (!instanceProxy)
+
+ if (instanceProxy==NULL)
{
g_printerr ("Failed to get proxy to Instance\n");
return FALSE;
@@ -210,7 +244,8 @@ dbus_connect ()
"org.sflphone.SFLphone",
"/org/sflphone/SFLphone/CallManager",
"org.sflphone.SFLphone.CallManager");
- if (!callManagerProxy)
+
+ if (callManagerProxy==NULL)
{
g_printerr ("Failed to get proxy to CallManagers\n");
return FALSE;
@@ -256,6 +291,7 @@ dbus_connect ()
"org.sflphone.SFLphone",
"/org/sflphone/SFLphone/ConfigurationManager",
"org.sflphone.SFLphone.ConfigurationManager");
+
if (!configurationManagerProxy)
{
g_printerr ("Failed to get proxy to ConfigurationManager\n");
@@ -1505,3 +1541,58 @@ dbus_get_sip_port( void )
return (guint)portNum;
}
+gchar* dbus_get_stun_server (void)
+{
+ GError* error = NULL;
+ gchar* server;
+ org_sflphone_SFLphone_ConfigurationManager_get_stun_server(
+ configurationManagerProxy,
+ &server,
+ &error);
+ if(error)
+ {
+ g_error_free(error);
+ }
+ return server;
+}
+
+void dbus_set_stun_server( gchar* server)
+{
+ GError* error = NULL;
+ org_sflphone_SFLphone_ConfigurationManager_set_stun_server(
+ configurationManagerProxy,
+ server,
+ &error);
+ if(error)
+ {
+ g_error_free(error);
+ }
+}
+
+guint dbus_stun_is_enabled (void)
+{
+ GError* error = NULL;
+ guint stun;
+ org_sflphone_SFLphone_ConfigurationManager_is_stun_enabled(
+ configurationManagerProxy,
+ &stun,
+ &error);
+ if(error)
+ {
+ g_error_free(error);
+ }
+ return stun;
+}
+
+void dbus_enable_stun (void)
+{
+
+ GError* error = NULL;
+ org_sflphone_SFLphone_ConfigurationManager_enable_stun(
+ configurationManagerProxy,
+ &error);
+ if(error)
+ {
+ g_error_free(error);
+ }
+}
diff --git a/sflphone-gtk/src/dbus.h b/sflphone-gtk/src/dbus.h
index 4989443f06897300ebcd52e6cea87c6721cbcc58..f621cd04b10d75cf9bd98792b34bfa8d368f9432 100644
--- a/sflphone-gtk/src/dbus.h
+++ b/sflphone-gtk/src/dbus.h
@@ -429,4 +429,10 @@ void dbus_set_sip_port(const guint portNum);
guint dbus_get_sip_port();
+gchar* dbus_get_stun_server (void);
+void dbus_set_stun_server( gchar* server);
+
+guint dbus_stun_is_enabled (void);
+void dbus_enable_stun (void);
+
#endif
diff --git a/src/Makefile.am b/src/Makefile.am
index 5be031174cdf639b7967fbb297021f6256af06e5..311be41e02e478b31e549bd5a1a2e9c29cc6b65f 100644
--- a/src/Makefile.am
+++ b/src/Makefile.am
@@ -39,8 +39,7 @@ sflphoned_SOURCES = \
call.cpp \
account.cpp \
sipcall.cpp \
- $(IAXSOURCES) \
- useragent.cpp
+ $(IAXSOURCES)
sflphoned_CXXFLAGS = \
-DPREFIX=\"$(prefix)\" -DPROGSHAREDIR=\"${datadir}/sflphone\" \
@@ -62,6 +61,8 @@ sflphoned_LDADD = \
@PULSEAUDIO_LIBS@ \
@SAMPLERATE_LIBS@
+#sflphoned_LDFLAGS=-pg
+
noinst_LTLIBRARIES = libsflphone.la
noinst_HEADERS = \
@@ -78,8 +79,7 @@ noinst_HEADERS = \
accountcreator.h \
sipvoiplink.h \
call.h \
- sipcall.h \
- useragent.h
+ sipcall.h
libsflphone_la_LIBADD = \
$(src)/libs/stund/libstun.la \
diff --git a/src/account.cpp b/src/account.cpp
index 5bc6ba36ebb8325da825c8e93d3e1fdfa30dc1a7..b50fc71fc49d5648d875ac8bc4cb99758b82a9d3 100644
--- a/src/account.cpp
+++ b/src/account.cpp
@@ -22,9 +22,10 @@
#include "account.h"
#include "manager.h"
-Account::Account(const AccountID& accountID) :
- _accountID(accountID), _link(NULL), _enabled(false)
+Account::Account(const AccountID& accountID, std::string type) :
+ _accountID(accountID), _link(NULL), _enabled(false), _type(type)
{
+ setRegistrationState(Unregistered);
}
Account::~Account()
@@ -46,3 +47,9 @@ void Account::loadConfig()
#endif
}
+void Account::setRegistrationState( RegistrationState state ) {
+ _registrationState = state;
+
+ // Notify the client
+ Manager::instance().connectionStatusNotification( );
+}
diff --git a/src/account.h b/src/account.h
index 571f7220315facc986953b1c7a9cc9a2cb2f7b89..faf33dc162725e39c3dd4ad23985c0bb207b83c3 100644
--- a/src/account.h
+++ b/src/account.h
@@ -38,6 +38,19 @@ class VoIPLink;
typedef std::string AccountID;
+/** Contains all the state an Voip can be in */
+typedef enum RegistrationState {
+ Unregistered,
+ Trying,
+ Registered,
+ Error,
+ ErrorAuth ,
+ ErrorNetwork ,
+ ErrorHost,
+ ErrorExistStun,
+ ErrorConfStun
+} RegistrationState;
+
#define AccountNULL ""
// Common account parameters
@@ -59,7 +72,7 @@ class Account{
public:
- Account(const AccountID& accountID);
+ Account(const AccountID& accountID, std::string type);
/**
* Virtual destructor
@@ -106,10 +119,28 @@ class Account{
* Get the registration state of the specified link
* @return RegistrationState The registration state of underlying VoIPLink
*/
- VoIPLink::RegistrationState getRegistrationState() { return _link->getRegistrationState(); }
+ inline RegistrationState getRegistrationState() { return _registrationState; }
- private:
+ void setRegistrationState( RegistrationState state );
+
+ /* inline functions */
+ /* They should be treated like macro definitions by the C++ compiler */
+ inline std::string getUsername( void ) { return _username; }
+ inline void setUsername( std::string username) { _username = username; }
+
+ inline std::string getHostname( void ) { return _hostname; }
+ inline void setHostname( std::string hostname) { _hostname = hostname; }
+
+ inline std::string getPassword( void ) { return _password; }
+ inline void setPassword( std::string password ) { _password = password; }
+
+ inline std::string getAlias( void ) { return _alias; }
+ inline void setAlias( std::string alias ) { _alias = alias; }
+ inline std::string getType( void ) { return _type; }
+ inline void setType( std::string type ) { _type = type; }
+
+ private:
// copy constructor
Account(const Account& rh);
@@ -122,6 +153,32 @@ class Account{
*/
AccountID _accountID;
+ /**
+ * Account login information: username
+ */
+ std::string _username;
+
+ /**
+ * Account login information: hostname
+ */
+ std::string _hostname;
+
+ /**
+ * Account login information: password
+ */
+ std::string _password;
+
+ /**
+ * Account login information: Alias
+ */
+ std::string _alias;
+
+ /*
+ * The account type
+ * IAX2 or SIP
+ */
+ std::string _type;
+
/**
* Voice over IP Link contains a listener thread and calls
*/
@@ -134,6 +191,11 @@ class Account{
*/
bool _enabled;
+ /*
+ * The registration state of the account
+ */
+ RegistrationState _registrationState;
+
};
#endif
diff --git a/src/audio/Makefile.am b/src/audio/Makefile.am
index 19cf33ee8124858805aa0f2bd72f73a7c1ec69e8..73659fe66c2bec7b756b3db7049ca65d14f8ef4c 100644
--- a/src/audio/Makefile.am
+++ b/src/audio/Makefile.am
@@ -30,6 +30,7 @@ libaudio_la_SOURCES = \
dtmf.cpp \
tone.cpp \
alsalayer.cpp \
+ audiolayer.cpp \
pulselayer.cpp \
audiodevice.cpp \
dtmfgenerator.cpp \
diff --git a/src/audio/alsalayer.cpp b/src/audio/alsalayer.cpp
index acd933c9d071b4d82e1a02c7206f07ab069a279d..5409c855963e18002f45ebb8ab787a98afef0f83 100644
--- a/src/audio/alsalayer.cpp
+++ b/src/audio/alsalayer.cpp
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2008 Savoir-Faire Linux inc.
+ * Copyright (C) 2008 2009 Savoir-Faire Linux inc.
* Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
*
* This program is free software; you can redistribute it and/or modify
@@ -19,13 +19,7 @@
#include "alsalayer.h"
-void* ringtoneThreadEntry( void *ptr);
-
-static pthread_t ringtone_thread;
-bool ringtone_thread_is_running;
-
-pthread_mutex_t mut = PTHREAD_MUTEX_INITIALIZER;
-pthread_cond_t cond = PTHREAD_COND_INITIALIZER;
+int framesPerBufferAlsa = 2048;
// Constructor
AlsaLayer::AlsaLayer( ManagerImpl* manager )
@@ -34,70 +28,63 @@ pthread_cond_t cond = PTHREAD_COND_INITIALIZER;
, _CaptureHandle(NULL)
, _periodSize()
, _audioPlugin()
- , _inChannel()
- , _outChannel()
- , _defaultVolume(100)
- , IDSoundCards()
+ , IDSoundCards()
+ , _is_prepared_playback (false)
+ , _is_running_playback (false)
+ , _is_open_playback (false)
+ , _is_prepared_capture (false)
+ , _is_running_capture (false)
+ , _is_open_capture (false)
+ , _trigger_request (false)
+
{
_debug(" Constructor of AlsaLayer called\n");
-
- // The flag to stop the ringtone thread loop
- ringtone_thread_is_running = false;
+ /* Instanciate the audio thread */
+ _audioThread = new AudioThread (this);
}
// Destructor
AlsaLayer::~AlsaLayer (void)
{
- _debugAlsa("Close ALSA streams\n");
- closeCaptureStream();
- closePlaybackStream();
- deviceClosed = true;
-
- ringtone_thread_is_running = false;
- pthread_join(ringtone_thread, NULL);
+ _debug("Destructor of AlsaLayer called\n");
+ closeLayer();
}
void
AlsaLayer::closeLayer()
{
_debugAlsa("Close ALSA streams\n");
+
+ if (_audioThread)
+ {
+ _debug("Try to stop audio thread\n");
+ delete _audioThread; _audioThread=NULL;
+ }
+
closeCaptureStream();
closePlaybackStream();
- deviceClosed = true;
-
- ringtone_thread_is_running = false;
}
bool
AlsaLayer::openDevice (int indexIn, int indexOut, int sampleRate, int frameSize, int stream , std::string plugin)
{
-
- // We don't accept that the audio plugin is changed during a conversation
- if( _talk ){
- _debug("can't switch audio plugin when talking\n. Please hang up and try again...\n");
- return false;
- }
-
- if(deviceClosed == false)
+ /* Close the devices before open it */
+ if (stream == SFL_PCM_BOTH && is_capture_open() == true && is_playback_open() == true)
{
- if( stream == SFL_PCM_CAPTURE )
- closeCaptureStream();
- else if( stream == SFL_PCM_PLAYBACK)
- closePlaybackStream();
- else
- {
- closeCaptureStream();
- closePlaybackStream();
- }
+ closeCaptureStream();
+ closePlaybackStream();
}
+ else if ((stream == SFL_PCM_CAPTURE || stream == SFL_PCM_BOTH) && is_capture_open() == true)
+ closeCaptureStream ();
+ else if ((stream == SFL_PCM_PLAYBACK || stream == SFL_PCM_BOTH) && is_playback_open () == true)
+ closePlaybackStream ();
+
_indexIn = indexIn;
_indexOut = indexOut;
_sampleRate = sampleRate;
_frameSize = frameSize;
_audioPlugin = plugin;
- _inChannel = 1;
- _outChannel = 1;
_debugAlsa(" Setting AlsaLayer: device in=%2d, out=%2d\n", _indexIn, _indexOut);
_debugAlsa(" : alsa plugin=%s\n", _audioPlugin.c_str());
@@ -105,17 +92,7 @@ AlsaLayer::openDevice (int indexIn, int indexOut, int sampleRate, int frameSize,
_debugAlsa(" : sample rate=%5d, format=%s\n", _sampleRate, SFLDataFormatString);
ost::MutexLock lock( _mutex );
-
- /*void **hint;
- int r = snd_device_name_hint(-1, "pcm", &hint);
-
- while( *hint ){
- printf("%s\n", snd_device_name_get_hint(*hint, "DESC"));
- ++hint;
- }*/
-
-
std::string pcmp = buildDeviceTopo( plugin , indexOut , 0);
std::string pcmc = buildDeviceTopo( PCM_PLUGHW , indexIn , 0);
@@ -125,142 +102,40 @@ AlsaLayer::openDevice (int indexIn, int indexOut, int sampleRate, int frameSize,
void
AlsaLayer::startStream(void)
{
- if( _CaptureHandle && _PlaybackHandle )
- {
- _talk = true ;
- _debugAlsa(" Start stream\n");
- int err;
- //ost::MutexLock lock( _mutex );
- snd_pcm_prepare( _CaptureHandle );
- snd_pcm_start( _CaptureHandle ) ;
-
- snd_pcm_prepare( _PlaybackHandle );
- if( (err = snd_pcm_start( _PlaybackHandle)) < 0 ) _debugAlsa(" Cannot start (%s)\n", snd_strerror(err));
- }
+ _debug ("Start ALSA streams\n");
+ prepareCaptureStream ();
+ startCaptureStream ();
+ startPlaybackStream ();
}
void
AlsaLayer::stopStream(void)
{
- if( _CaptureHandle && _PlaybackHandle )
- {
- //ost::MutexLock lock( _mutex );
- _debugAlsa(" Stop Stream\n ");
- _talk = false;
- snd_pcm_drop( _CaptureHandle );
- snd_pcm_prepare( _CaptureHandle );
- snd_pcm_drop( _PlaybackHandle );
- snd_pcm_prepare( _PlaybackHandle );
- _urgentBuffer.flush();
- }
-}
-
-
-void* ringtoneThreadEntry( void *ptr )
-{
- while( ringtone_thread_is_running )
- {
- ( ( AlsaLayer *) ptr) -> playTones();
- //sleep(0.1);
- }
- /*
- pthread_mutex_lock(&mut);
- while( ((AlsaLayer*)ptr)->_manager->getTelephoneTone() == NULL )
- {
- pthread_cond_wait(&cond, &mut);
- }
- ( AlsaLayer *) ptr -> playTones();
- pthread_mutex_unlock(&mut);*/
- return 0;
-}
-
-
- void
-AlsaLayer::fillHWBuffer( void)
-{
+ _debug ("Stop ALSA streams\n");
+ stopCaptureStream ();
+ //stopPlaybackStream ();
- unsigned char* data;
- int pcmreturn, l1, l2;
- short s1, s2;
- int periodSize = 128 ;
- int frames = periodSize >> 2 ;
- _debug("frames = %d\n", frames);
-
- data = (unsigned char*)malloc(periodSize);
- for(l1 = 0; l1 < 100; l1++) {
- for(l2 = 0; l2 < frames; l2++) {
- s1 = 0;
- s2 = 0;
- data[4*l2] = (unsigned char)s1;
- data[4*l2+1] = s1 >> 8;
- data[4*l2+2] = (unsigned char)s2;
- data[4*l2+3] = s2 >> 8;
- }
- while ((pcmreturn = snd_pcm_writei(_PlaybackHandle, data, frames)) < 0) {
- snd_pcm_prepare(_PlaybackHandle);
- //_debugAlsa("< Buffer Underrun >\n");
- }
- }
+ /* Flush the ring buffers */
+ flushUrgent();
+ flushMain();
}
- bool
-AlsaLayer::isStreamActive (void)
-{
- ost::MutexLock lock( _mutex );
- return (isPlaybackActive() && isCaptureActive());
-}
-
- int
-AlsaLayer::playSamples(void* buffer, int toCopy, bool isTalking)
-{
- //ost::MutexLock lock( _mutex );
- if( isTalking )
- _talk = true;
- if ( _PlaybackHandle ){
- write( adjustVolume( buffer , toCopy , SFL_PCM_PLAYBACK ) , toCopy );
- }
- return 0;
-}
-
- int
-AlsaLayer::putUrgent(void* buffer, int toCopy)
-{
- int nbBytes = 0;
-
- if ( _PlaybackHandle ){
- //fillHWBuffer();
- int a = _urgentBuffer.AvailForPut();
- if( a >= toCopy ){
- nbBytes = _urgentBuffer.Put( buffer , toCopy , _defaultVolume );
- } else {
- nbBytes = _urgentBuffer.Put( buffer , a , _defaultVolume ) ;
- }
- }
-
- return nbBytes;
-}
-
-void AlsaLayer::trigger_thread(void)
-{
- _debug("Wake up the ringtone thread\n");
- pthread_cond_broadcast(&cond);
-}
-
-
int
AlsaLayer::canGetMic()
{
int avail;
- if ( _CaptureHandle ) {
- avail = snd_pcm_avail_update( _CaptureHandle );
- //printf("%d\n", avail );
- if(avail > 0)
- return avail;
- else
- return 0;
+
+ if (! _CaptureHandle )
+ return 0;
+
+ avail = snd_pcm_avail_update( _CaptureHandle );
+
+ if( avail == -EPIPE ){
+ stop_capture ();
+ return 0;
}
else
- return 0;
+ return ((avail < 0)? 0:avail);
}
int
@@ -275,75 +150,77 @@ AlsaLayer::getMic(void *buffer, int toCopy)
return res ;
}
-
- bool
-AlsaLayer::isStreamStopped (void)
-{
- ost::MutexLock lock( _mutex );
- return !(isStreamActive());
-}
-
-
-void AlsaLayer::reducePulseAppsVolume( void ){}
-void AlsaLayer::restorePulseAppsVolume( void ){}
-
//////////////////////////////////////////////////////////////////////////////////////////////
///////////////// ALSA PRIVATE FUNCTIONS ////////////////////////////////////////////////
////////////////////////////////////////////////////////////////////////////////////////////
+
+void AlsaLayer::stopCaptureStream (void)
+{
+ if(_CaptureHandle){
+ snd_pcm_drop (_CaptureHandle);
+ stop_capture ();
+ }
+}
+void AlsaLayer::closeCaptureStream (void)
+{
+ if( is_capture_prepared() == true && is_capture_running() == true )
+ stopCaptureStream ();
+ if (is_capture_open())
+ snd_pcm_close (_CaptureHandle);
+ close_capture ();
+}
- void
-AlsaLayer::playTones( void )
+void AlsaLayer::startCaptureStream (void)
{
- int frames;
- int maxBytes;
+ if(_CaptureHandle){
+ snd_pcm_start (_CaptureHandle);
+ start_capture();
+ }
+}
- pthread_mutex_lock(&mut);
- while(!_manager-> getTelephoneTone() && !_manager->getTelephoneFile())
- {
- _debug("Make the ringtone thread wait\n");
- pthread_cond_wait(&cond, &mut);
- }
-
- //frames = _periodSize ;
- frames = 940 ;
- maxBytes = frames * sizeof(SFLDataFormat) ;
- SFLDataFormat* out = (SFLDataFormat*)malloc(maxBytes * sizeof(SFLDataFormat));
- if( _talk ) {}
- else {
- AudioLoop *tone = _manager -> getTelephoneTone();
- if( tone != 0 ){
- tone -> getNext( out , frames , _manager->getSpkrVolume() );
- write( out , maxBytes );
- }
- else if( ( tone=_manager->getTelephoneFile() ) != 0 ){
- tone ->getNext( out , frames , _manager->getSpkrVolume() );
- write( out , maxBytes );
- }
+void AlsaLayer::prepareCaptureStream (void)
+{
+ if (is_capture_open() ) {
+ if(snd_pcm_prepare (_CaptureHandle) < 0) _debug("Error preparing the device\n");
+ prepare_capture ();
}
- // free the temporary data buffer
- free( out ); out = 0;
- pthread_mutex_unlock(&mut);
}
+void AlsaLayer::stopPlaybackStream (void)
+{
+ if( _PlaybackHandle){
+ snd_pcm_drop (_PlaybackHandle);
+ stop_playback ();
+ }
+}
-bool
-AlsaLayer::isPlaybackActive(void) {
- ost::MutexLock guard( _mutex );
- if( _PlaybackHandle )
- return (snd_pcm_state(_PlaybackHandle) == SND_PCM_STATE_RUNNING ? true : false);
- else
- return false;
+
+void AlsaLayer::closePlaybackStream (void)
+{
+ if( is_playback_prepared() == true && is_playback_running() == true )
+ stopPlaybackStream ();
+ if (is_playback_open())
+ snd_pcm_close (_PlaybackHandle);
+
+ close_playback ();
}
-bool
-AlsaLayer::isCaptureActive(void) {
- ost::MutexLock guard( _mutex );
- if( _CaptureHandle )
- return (snd_pcm_state( _CaptureHandle) == SND_PCM_STATE_RUNNING ? true : false);
- else
- return false;
+void AlsaLayer::startPlaybackStream (void)
+{
+ if( _PlaybackHandle){
+ snd_pcm_start (_PlaybackHandle);
+ start_playback();
+ }
+}
+
+void AlsaLayer::preparePlaybackStream (void)
+{
+ if(is_playback_open()){
+ if( snd_pcm_prepare (_PlaybackHandle) < 0) _debug("Error preparing the device\n");
+ prepare_playback ();
+ }
}
bool AlsaLayer::alsa_set_params( snd_pcm_t *pcm_handle, int type, int rate ){
@@ -444,23 +321,8 @@ bool AlsaLayer::alsa_set_params( snd_pcm_t *pcm_handle, int type, int rate ){
return false;
}
- if( type == 1 ){
- /*if( (err = snd_async_add_pcm_handler( &_AsyncHandler, pcm_handle , AlsaCallBack, this ) < 0)){
- _debugAlsa(" Unable to install the async callback handler (%s)\n", snd_strerror(err));
- return false;
- }*/
-
- // So the loop could start when the ringtone thread entry function is reached
- ringtone_thread_is_running = true;
- if( pthread_create(&ringtone_thread, NULL, ringtoneThreadEntry, this) != 0 )
- {
- _debug("Unable to start the ringtone posix thread\n");
- return false;
- }
- }
snd_pcm_sw_params_free( swparams );
- deviceClosed = false;
return true;
}
@@ -477,13 +339,17 @@ AlsaLayer::open_device(std::string pcm_p, std::string pcm_c, int flag)
if((err = snd_pcm_open(&_PlaybackHandle, pcm_p.c_str(), SND_PCM_STREAM_PLAYBACK, 0 )) < 0){
_debugAlsa("Error while opening playback device %s\n", pcm_p.c_str());
setErrorMessage( ALSA_PLAYBACK_DEVICE );
+ close_playback ();
return false;
}
if(!alsa_set_params( _PlaybackHandle, 1, getSampleRate() )){
_debug("playback failed\n");
snd_pcm_close( _PlaybackHandle );
+ close_playback ();
return false;
}
+
+ open_playback ();
}
if(flag == SFL_PCM_BOTH || flag == SFL_PCM_CAPTURE)
@@ -491,20 +357,23 @@ AlsaLayer::open_device(std::string pcm_p, std::string pcm_c, int flag)
if( (err = snd_pcm_open(&_CaptureHandle, pcm_c.c_str(), SND_PCM_STREAM_CAPTURE, 0)) < 0){
_debugAlsa("Error while opening capture device %s\n", pcm_c.c_str());
setErrorMessage( ALSA_CAPTURE_DEVICE );
+ close_capture ();
return false;
}
if(!alsa_set_params( _CaptureHandle, 0, 8000 /*getSampleRate()*/ )){
_debug("capture failed\n");
snd_pcm_close( _CaptureHandle );
+ close_capture ();
return false;
}
- if( (err = snd_pcm_start( _CaptureHandle ))<0) {
- _debugAlsa( "snd_pcm_start failed\n");
- }
+
+ open_capture ();
+
+ startCaptureStream ();
}
- //TODO something, really!
- _talk = false;
+ /* Start the secondary audio thread for callbacks */
+ _audioThread->start();
return true;
}
@@ -513,33 +382,34 @@ AlsaLayer::open_device(std::string pcm_p, std::string pcm_c, int flag)
int
AlsaLayer::write(void* buffer, int length)
{
- //if(snd_pcm_state( _PlaybackHandle ) == SND_PCM_STATE_XRUN)
- //handle_xrun_playback();
- //_debugAlsa("avail = %d - toWrite = %d\n" , snd_pcm_avail_update( _PlaybackHandle ) , length / 2);
- snd_pcm_uframes_t frames = snd_pcm_bytes_to_frames( _PlaybackHandle, length);
- int err = snd_pcm_writei( _PlaybackHandle , buffer , frames );
- switch(err) {
- case -EAGAIN:
- _debugAlsa("EAGAIN (%s)\n", snd_strerror( err ));
- snd_pcm_resume( _PlaybackHandle );
- break;
- case -EPIPE:
- _debugAlsa(" UNDERRUN (%s)\n", snd_strerror(err));
- handle_xrun_playback();
- snd_pcm_writei( _PlaybackHandle , buffer , frames );
- break;
- case -ESTRPIPE:
- _debugAlsa(" ESTRPIPE(%s)\n", snd_strerror(err));
- snd_pcm_resume( _PlaybackHandle );
- break;
- case -EBADFD:
- _debugAlsa(" (%s)\n", snd_strerror( err ));
- break;
+
+ if (_trigger_request == true)
+ {
+ _trigger_request = false;
+ startPlaybackStream ();
}
- if( ( err >=0 ) && ( err < (int)frames ) )
- _debugAlsa("Short write : %d out of %d\n", err , (int)frames);
+ snd_pcm_uframes_t frames = snd_pcm_bytes_to_frames( _PlaybackHandle, length);
+ int err;
+ if ((err=snd_pcm_writei( _PlaybackHandle , buffer , frames ))<0)
+ {
+ switch(err) {
+ case -EPIPE:
+ case -ESTRPIPE:
+ case -EIO:
+ //_debugAlsa(" XRUN playback ignored (%s)\n", snd_strerror(err));
+ handle_xrun_playback();
+ if (snd_pcm_writei( _PlaybackHandle , buffer , frames )<0)
+ _debugAlsa ("XRUN handling failed\n");
+ _trigger_request = true;
+ break;
+ default:
+ //_debugAlsa ("Write error unknown - dropping frames **********************************: %s\n", snd_strerror(err));
+ stopPlaybackStream ();
+ break;
+ }
+ }
return ( err > 0 )? err : 0 ;
}
@@ -547,29 +417,32 @@ AlsaLayer::write(void* buffer, int length)
int
AlsaLayer::read( void* buffer, int toCopy)
{
- if(deviceClosed || _CaptureHandle == NULL)
- return 0;
+ ost::MutexLock lock( _mutex );
+
+ int samples;
- int err;
if(snd_pcm_state( _CaptureHandle ) == SND_PCM_STATE_XRUN)
{
- snd_pcm_prepare( _CaptureHandle );
- snd_pcm_start( _CaptureHandle );
+ _debug("xrun caught before start\n");
+ prepareCaptureStream ();
+ startCaptureStream ();
}
+
snd_pcm_uframes_t frames = snd_pcm_bytes_to_frames( _CaptureHandle, toCopy );
- if(( err = snd_pcm_readi( _CaptureHandle, buffer, frames)) < 0 ) {
- switch(err){
- case EPERM:
- _debugAlsa(" Capture EPERM (%s)\n", snd_strerror(err));
- snd_pcm_prepare( _CaptureHandle);
- snd_pcm_start( _CaptureHandle );
- break;
- case -EAGAIN:
- _debugAlsa(" Capture EAGAIN (%s)\n", snd_strerror(err));
- break;
+ if(( samples = snd_pcm_readi( _CaptureHandle, buffer, frames)) < 0 ) {
+ switch (samples)
+ {
case -EPIPE:
- _debugAlsa(" Capture EPIPE (%s)\n", snd_strerror(err));
+ case -ESTRPIPE:
+ case -EIO:
+ _debugAlsa(" XRUN capture ignored (%s)\n", snd_strerror(samples));
handle_xrun_capture();
+ samples = snd_pcm_readi( _CaptureHandle, buffer, frames);
+ if (samples<0) samples=0;
+ break;
+ default:
+ //_debugAlsa ("Error when capturing data ***********************************************: %s\n", snd_strerror(samples));
+ stopCaptureStream();
break;
}
return 0;
@@ -577,14 +450,6 @@ AlsaLayer::read( void* buffer, int toCopy)
return toCopy;
-}
-
- int
-AlsaLayer::putInCache( char code UNUSED,
- void *buffer UNUSED,
- int toCopy UNUSED )
-{
- return 1;
}
void
@@ -596,9 +461,9 @@ AlsaLayer::handle_xrun_capture( void )
int res = snd_pcm_status( _CaptureHandle, status );
if( res <= 0){
if(snd_pcm_status_get_state(status) == SND_PCM_STATE_XRUN ){
- snd_pcm_drop( _CaptureHandle );
- snd_pcm_prepare( _CaptureHandle );
- snd_pcm_start( _CaptureHandle );
+ stopCaptureStream ();
+ prepareCaptureStream ();
+ startCaptureStream ();
}
}
else
@@ -618,9 +483,9 @@ AlsaLayer::handle_xrun_playback( void )
state = snd_pcm_status_get_state( status );
if( state == SND_PCM_STATE_XRUN )
{
- snd_pcm_drop( _PlaybackHandle );
- snd_pcm_prepare( _PlaybackHandle );
- //snd_pcm_start( _PlaybackHandle );
+ stopPlaybackStream ();
+ preparePlaybackStream ();
+ _trigger_request = true;
}
}
}
@@ -695,25 +560,7 @@ AlsaLayer::getSoundCardsInfo( int stream )
return cards_id;
}
- void
-AlsaLayer::closeCaptureStream( void)
-{
- if(_CaptureHandle){
- snd_pcm_drop( _CaptureHandle );
- snd_pcm_close( _CaptureHandle );
- _CaptureHandle = 0;
- }
-}
- void
-AlsaLayer::closePlaybackStream( void)
-{
- if(_PlaybackHandle){
- snd_pcm_drop( _PlaybackHandle );
- snd_pcm_close( _PlaybackHandle );
- _PlaybackHandle = 0;
- }
-}
bool
AlsaLayer::soundCardIndexExist( int card , int stream )
@@ -749,8 +596,68 @@ AlsaLayer::soundCardGetIndex( std::string description )
return 0;
}
- void*
-AlsaLayer::adjustVolume( void* buffer , int len, int stream )
+void AlsaLayer::audioCallback (void)
+{
+
+ int toGet, toPut, urgentAvail, normalAvail, micAvailPut, maxBytes;
+ unsigned short spkrVolume, micVolume;
+ AudioLoop *tone;
+
+ SFLDataFormat *out;
+
+ spkrVolume = _manager->getSpkrVolume();
+ micVolume = _manager->getMicVolume();
+
+ // AvailForGet tell the number of chars inside the buffer
+ // framePerBuffer are the number of data for one channel (left)
+ urgentAvail = _urgentRingBuffer.AvailForGet();
+ if (urgentAvail > 0) {
+ // Urgent data (dtmf, incoming call signal) come first.
+ toGet = (urgentAvail < (int)(framesPerBufferAlsa * sizeof(SFLDataFormat))) ? urgentAvail : framesPerBufferAlsa * sizeof(SFLDataFormat);
+ out = (SFLDataFormat*)malloc(toGet * sizeof(SFLDataFormat) );
+ _urgentRingBuffer.Get(out, toGet, spkrVolume);
+ /* Play the sound */
+ write( out , toGet );
+ free(out); out=0;
+ // Consume the regular one as well (same amount of bytes)
+ _voiceRingBuffer.Discard(toGet);
+ } else {
+ tone = _manager->getTelephoneTone();
+ toGet = 940 ;
+ maxBytes = toGet * sizeof(SFLDataFormat) ;
+ if ( tone != 0) {
+ out = (SFLDataFormat*)malloc(maxBytes * sizeof(SFLDataFormat));
+ tone->getNext(out, toGet, spkrVolume);
+ write (out , maxBytes);
+ } else if ( (tone=_manager->getTelephoneFile()) != 0 ) {
+ out = (SFLDataFormat*)malloc(maxBytes * sizeof(SFLDataFormat) );
+ tone->getNext(out, toGet, spkrVolume);
+ write (out , maxBytes);
+ } else {
+ // If nothing urgent, play the regular sound samples
+ normalAvail = _voiceRingBuffer.AvailForGet();
+ toGet = (normalAvail < (int)(framesPerBufferAlsa * sizeof(SFLDataFormat))) ? normalAvail : framesPerBufferAlsa * sizeof(SFLDataFormat);
+ out = (SFLDataFormat*)malloc(framesPerBufferAlsa * sizeof(SFLDataFormat));
+
+ if (toGet) {
+ _voiceRingBuffer.Get(out, toGet, spkrVolume);
+ write (out, toGet);
+ } else {
+ bzero(out, framesPerBufferAlsa * sizeof(SFLDataFormat));
+ }
+ }
+ free(out); out=0;
+ }
+
+ // Additionally handle the mic's audio stream
+ //micAvailPut = _micRingBuffer.AvailForPut();
+ //toPut = (micAvailPut <= (int)(framesPerBufferAlsa * sizeof(SFLDataFormat))) ? micAvailPut : framesPerBufferAlsa * sizeof(SFLDataFormat);
+ //_debug("AL: Nb sample: %d char, [0]=%f [1]=%f [2]=%f\n", toPut, in[0], in[1], in[2]);
+ //_micRingBuffer.Put(in, toPut, micVolume);
+}
+
+
+void* AlsaLayer::adjustVolume( void* buffer , int len, int stream )
{
int vol, i, size;
SFLDataFormat *src = NULL;
@@ -768,4 +675,3 @@ AlsaLayer::adjustVolume( void* buffer , int len, int stream )
}
return src ;
}
-
diff --git a/src/audio/alsalayer.h b/src/audio/alsalayer.h
index c9dfa016d2cf8d61d68df15d815d7cc8f13eb088..875e0cc14124b19963b50e7fbc947e9ba3e7ad65 100644
--- a/src/audio/alsalayer.h
+++ b/src/audio/alsalayer.h
@@ -21,8 +21,8 @@
#define _ALSA_LAYER_H
#include "audiolayer.h"
-
-#include <pthread.h>
+#include "eventthread.h"
+#include <alsa/asoundlib.h>
class RingBuffer;
class ManagerImpl;
@@ -48,8 +48,6 @@ class AlsaLayer : public AudioLayer {
*/
~AlsaLayer(void);
- void trigger_thread(void);
-
void closeLayer( void );
/**
@@ -81,52 +79,6 @@ class AlsaLayer : public AudioLayer {
*/
void stopStream(void);
- /**
- * Check if the playback is running
- * @return true if the state of the playback handle equals SND_PCM_STATE_RUNNING
- * false otherwise
- */
- bool isPlaybackActive( void );
-
- /**
- * Check if the capture is running
- * @return true if the state of the capture handle equals SND_PCM_STATE_RUNNING
- * false otherwise
- */
- bool isCaptureActive( void );
-
- /**
- * Check if both capture and playback are running
- * @return true if capture and playback are running
- * false otherwise
- */
- bool isStreamActive(void);
-
- /**
- * Check if both capture and playback are stopped
- * @return true if capture and playback are stopped
- * false otherwise
- */
- bool isStreamStopped(void);
-
- /**
- * Send samples to the audio device.
- * @param buffer The buffer containing the data to be played ( voice and DTMF )
- * @param toCopy The number of samples, in bytes
- * @param isTalking If whether or not the conversation is running
- * @return int The number of bytes played
- */
- int playSamples(void* buffer, int toCopy, bool isTalking);
-
- /**
- * Send a chunk of data to the hardware buffer to start the playback
- * Copy data in the urgent buffer.
- * @param buffer The buffer containing the data to be played ( ringtones )
- * @param toCopy The size of the buffer
- * @return int The number of bytes copied in the urgent buffer
- */
- int putUrgent(void* buffer, int toCopy);
-
/**
* Query the capture device for number of bytes available in the hardware ring buffer
* @return int The number of bytes available
@@ -185,34 +137,9 @@ class AlsaLayer : public AudioLayer {
*/
std::string getAudioPlugin( void ) { return _audioPlugin; }
- /**
- * UNUSED in ALSA layer
- */
- int putInCache( char code, void *buffer, int toCopy );
-
-
- /**
- * UNUSED in ALSA layer
- */
- void reducePulseAppsVolume( void );
-
- /**
- * UNUSED in ALSA layer
- */
- void restorePulseAppsVolume( void );
-
- /**
- * UNUSED in ALSA layer
- */
- void setPlaybackVolume( UNUSED int volume ){}
- void setCaptureVolume( UNUSED int volume ){}
-
- /**
- * Callback used for asynchronous playback.
- * Write tones buffer to the alsa internal ring buffer.
- */
- void playTones( void );
+ void audioCallback (void);
+ bool isCaptureActive (void) { return is_capture_running (); }
private:
@@ -222,29 +149,44 @@ class AlsaLayer : public AudioLayer {
// Assignment Operator
AlsaLayer& operator=( const AlsaLayer& rh);
+ bool _is_prepared_playback;
+ bool _is_prepared_capture;
+ bool _is_running_capture;
+ bool _is_running_playback;
+ bool _is_open_playback;
+ bool _is_open_capture;
+ bool _trigger_request;
+
+ bool is_playback_prepared (void) { return _is_prepared_playback; }
+ bool is_capture_prepared (void) { return _is_prepared_capture; }
+ void prepare_playback (void) { _is_prepared_playback = true; }
+ void prepare_capture (void) { _is_prepared_capture = true; }
+ bool is_capture_running (void) { return _is_running_capture; }
+ bool is_playback_running (void) { return _is_running_playback; }
+ void start_playback (void) { _is_running_playback = true; }
+ void stop_playback (void) { _is_running_playback = false; _is_prepared_playback = false; }
+ void start_capture (void) { _is_running_capture = true; }
+ void stop_capture (void) { _is_running_capture = false; _is_prepared_capture = false; }
+ void close_playback (void) { _is_open_playback = false; }
+ void close_capture (void) { _is_open_capture = false; }
+ void open_playback (void) { _is_open_playback = true; }
+ void open_capture (void) { _is_open_capture = true; }
+ bool is_capture_open (void) { return _is_open_capture; }
+ bool is_playback_open (void) { return _is_open_playback; }
+
/**
* Drop the pending frames and close the capture device
* ALSA Library API
*/
void closeCaptureStream( void );
+ void stopCaptureStream( void );
+ void startCaptureStream( void );
+ void prepareCaptureStream( void );
- /**
- * Drop the pending frames and close the playback device
- * ALSA Library API
- */
void closePlaybackStream( void );
-
- /**
- * Fill the alsa internal ring buffer with chunks of data
- */
- void fillHWBuffer( void) ;
-
- /**
- * Callback used for asynchronous playback.
- * Called when a certain amount of data is written ot the device
- * @param pcm_callback The callback pointer
- */
- //static void AlsaCallBack( snd_async_handler_t* pcm_callback);
+ void stopPlaybackStream( void );
+ void startPlaybackStream( void );
+ void preparePlaybackStream( void );
/**
* Open the specified device.
@@ -280,6 +222,8 @@ class AlsaLayer : public AudioLayer {
*/
int read( void* buffer, int toCopy);
+
+
/**
* Recover from XRUN state for capture
* ALSA Library API
@@ -292,7 +236,9 @@ class AlsaLayer : public AudioLayer {
*/
void handle_xrun_playback( void );
- /**
+ void* adjustVolume( void* buffer , int len, int stream );
+
+/**
* Handles to manipulate playback stream
* ALSA Library API
*/
@@ -309,38 +255,16 @@ class AlsaLayer : public AudioLayer {
*/
snd_pcm_uframes_t _periodSize;
- /**
- * Volume is controlled by the application. Data buffer are modified here to adjust to the right volume selected by the user on the main interface
- * @param buffer The buffer to adjust
- * @param len The number of bytes
- * @param stream The stream mode ( PLAYBACK - CAPTURE )
- */
- void * adjustVolume( void * , int , int);
-
/**
* name of the alsa audio plugin used
*/
std::string _audioPlugin;
- /**
- * Input channel (mic) should be 1 mono
- */
- unsigned int _inChannel;
-
- /**
- * Output channel (stereo) should be 1 mono
- */
- unsigned int _outChannel;
-
- /**
- * Default volume for incoming RTP and Urgent sounds.
- */
- unsigned short _defaultVolume; // 100
-
-
/** Vector to manage all soundcard index - description association of the system */
std::vector<HwIDPair> IDSoundCards;
+ AudioThread *_audioThread;
+
};
#endif // _ALSA_LAYER_H_
diff --git a/src/audio/audiolayer.cpp b/src/audio/audiolayer.cpp
new file mode 100644
index 0000000000000000000000000000000000000000..3eea4392c737b51c3f954b438ca3dac7b57a30b9
--- /dev/null
+++ b/src/audio/audiolayer.cpp
@@ -0,0 +1,69 @@
+/*
+ * Copyright (C) 2009 Savoir-Faire Linux inc.
+ * Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ */
+
+#include "audiolayer.h"
+
+void AudioLayer::flushMain (void)
+{
+ ost::MutexLock guard(_mutex);
+ _voiceRingBuffer.flush();
+}
+
+void AudioLayer::flushUrgent (void)
+{
+ ost::MutexLock guard(_mutex);
+ _urgentRingBuffer.flush();
+}
+
+void AudioLayer::flushMic (void)
+{
+ ost::MutexLock guard(_mutex);
+ _micRingBuffer.flush();
+}
+
+int AudioLayer::putUrgent(void* buffer, int toCopy)
+{
+ int a;
+
+ ost::MutexLock guard(_mutex);
+ a = _urgentRingBuffer.AvailForPut();
+ if ( a >= toCopy ) {
+ return _urgentRingBuffer.Put(buffer, toCopy, _defaultVolume);
+ } else {
+ return _urgentRingBuffer.Put(buffer, a, _defaultVolume);
+ }
+ return 0;
+}
+
+int AudioLayer::putMain (void *buffer, int toCopy)
+{
+ int a;
+
+ ost::MutexLock guard(_mutex);
+ a = _voiceRingBuffer.AvailForPut();
+ if ( a >= toCopy ) {
+ return _voiceRingBuffer.Put(buffer, toCopy, _defaultVolume);
+ } else {
+ _debug("Chopping sound, Ouch! RingBuffer full ?\n");
+ return _voiceRingBuffer.Put(buffer, a, _defaultVolume);
+ }
+ return 0;
+}
+
+
diff --git a/src/audio/audiolayer.h b/src/audio/audiolayer.h
index 658ac3825deaad2d14fec6c31e547d60648d4ed3..679738b68c7fc15242eff85fd5e52c6068687f2f 100644
--- a/src/audio/audiolayer.h
+++ b/src/audio/audiolayer.h
@@ -22,24 +22,16 @@
#ifndef _AUDIO_LAYER_H
#define _AUDIO_LAYER_H
-#include "../global.h"
-#include "../manager.h"
+#include "global.h"
#include "audiodevice.h"
#include "ringbuffer.h"
+#include "manager.h"
#include <cc++/thread.h> // for ost::Mutex
-#include <vector>
-#include <alsa/asoundlib.h>
-#include <pulse/pulseaudio.h>
-
-#include <iostream>
-#include <istream>
-#include <sstream>
#define FRAME_PER_BUFFER 160
-
/**
* @file audiolayer.h
* @brief Main sound class. Manages the data transfers between the application and the hardware.
@@ -47,282 +39,225 @@
class AudioLayer {
- private:
-
- //copy constructor
- AudioLayer(const AudioLayer& rh);
-
- // assignment operator
- AudioLayer& operator=(const AudioLayer& rh);
-
- public:
- /**
- * Constructor
- * @param manager An instance of managerimpl
- */
- AudioLayer( ManagerImpl* manager , int type )
- : _layerType( type )
- , _manager(manager)
- , _urgentBuffer( SIZEBUF )
- , _talk ( false )
- , deviceClosed ( true )
- , _indexIn ( 0 )
- , _indexOut ( 0 )
- , _sampleRate ( 0 )
- , _frameSize ( 0 )
- , _inChannel( 1 )
- , _outChannel ( 1 )
- , _errorMessage ( 0 )
- , _mutex ()
+ private:
+
+ //copy constructor
+ AudioLayer(const AudioLayer& rh);
+
+ // assignment operator
+ AudioLayer& operator=(const AudioLayer& rh);
+
+ public:
+ /**
+ * Constructor
+ * @param manager An instance of managerimpl
+ */
+ AudioLayer( ManagerImpl* manager , int type )
+ : _layerType( type )
+ , _manager(manager)
+ , _voiceRingBuffer( SIZEBUF )
+ , _urgentRingBuffer( SIZEBUF)
+ , _micRingBuffer( SIZEBUF )
+ , _defaultVolume(100)
+ , _indexIn ( 0 )
+ , _indexOut ( 0 )
+ , _sampleRate ( 0 )
+ , _frameSize ( 0 )
+ , _inChannel( 1 )
+ , _outChannel ( 1 )
+ , _errorMessage ( 0 )
+ , _mutex ()
{
}
-
- /**
- * Destructor
- */
- ~AudioLayer(void){}
-
- virtual void closeLayer( void ) = 0;
-
- virtual void trigger_thread(void)=0;
-
- /**
- * Check if no devices are opened, otherwise close them.
- * Then open the specified devices by calling the private functions open_device
- * @param indexIn The number of the card choosen for capture
- * @param indexOut The number of the card choosen for playback
- * @param sampleRate The sample rate
- * @param frameSize The frame size
- * @param stream To indicate which kind of stream you want to open
- * SFL_PCM_CAPTURE
- * SFL_PCM_PLAYBACK
- * SFL_PCM_BOTH
- * @param plugin The alsa plugin ( dmix , default , front , surround , ...)
- */
- virtual bool openDevice(int indexIn, int indexOut, int sampleRate, int frameSize, int stream , std::string plugin) = 0;
-
- /**
- * Start the capture stream and prepare the playback stream.
- * The playback starts accordingly to its threshold
- * ALSA Library API
- */
- virtual void startStream(void) = 0;
-
- /**
- * Stop the playback and capture streams.
- * Drops the pending frames and put the capture and playback handles to PREPARED state
- * ALSA Library API
- */
- virtual void stopStream(void) = 0;
-
-
- /**
- * Check if the capture is running
- * @return true if the state of the capture handle equals SND_PCM_STATE_RUNNING
- * false otherwise
- */
- virtual bool isCaptureActive( void ) = 0;
-
- /**
- * Send samples to the audio device.
- * @param buffer The buffer containing the data to be played ( voice and DTMF )
- * @param toCopy The number of samples, in bytes
- * @param isTalking If whether or not the conversation is running
- * @return int The number of bytes played
- */
- virtual int playSamples(void* buffer, int toCopy, bool isTalking) = 0;
-
- /**
- * Send a chunk of data to the hardware buffer to start the playback
- * Copy data in the urgent buffer.
- * @param buffer The buffer containing the data to be played ( ringtones )
- * @param toCopy The size of the buffer
- * @return int The number of bytes copied in the urgent buffer
- */
- virtual int putUrgent(void* buffer, int toCopy) = 0;
-
- virtual int putInCache(char code, void* buffer, int toCopy) = 0;
-
- /**
- * Query the capture device for number of bytes available in the hardware ring buffer
- * @return int The number of bytes available
- */
- virtual int canGetMic() = 0;
-
- /**
- * Get data from the capture device
- * @param buffer The buffer for data
- * @param toCopy The number of bytes to get
- * @return int The number of bytes acquired ( 0 if an error occured)
- */
- virtual int getMic(void * buffer, int toCopy) = 0;
-
- /**
- * Scan the sound card available on the system
- * @param stream To indicate whether we are looking for capture devices or playback devices
- * SFL_PCM_CAPTURE
- * SFL_PCM_PLAYBACK
- * SFL_PCM_BOTH
- * @return std::vector<std::string> The vector containing the string description of the card
- */
- virtual std::vector<std::string> getSoundCardsInfo( int stream ) = 0;
-
- /**
- * Check if the given index corresponds to an existing sound card and supports the specified streaming mode
- * @param card An index
- * @param stream The stream mode
- * SFL_PCM_CAPTURE
- * SFL_PCM_PLAYBACK
- * SFL_PCM_BOTH
- * @return bool True if it exists and supports the mode
- * false otherwise
- */
- virtual bool soundCardIndexExist( int card , int stream ) = 0;
-
- /**
- * An index is associated with its string description
- * @param description The string description
- * @return int Its index
- */
- virtual int soundCardGetIndex( std::string description ) = 0;
-
- /**
- * Get the current audio plugin.
- * @return std::string The name of the audio plugin
- */
- virtual std::string getAudioPlugin( void ) = 0;
-
- virtual void reducePulseAppsVolume( void ) = 0;
- virtual void restorePulseAppsVolume( void ) = 0;
-
- virtual void setPlaybackVolume( int volume ) = 0;
- virtual void setCaptureVolume( int volume ) = 0;
-
- /**
- * Write accessor to the error state
- * @param error The error code
- * Could be: ALSA_PLAYBACK_DEVICE
- * ALSA_CAPTURE_DEVICE
- */
- void setErrorMessage(const int& error) { _errorMessage = error; }
-
- /**
- * Read accessor to the error state
- * @return int The error code
- */
- int getErrorMessage() { return _errorMessage; }
-
- /**
- * Get the index of the audio card for capture
- * @return int The index of the card used for capture
- * 0 for the first available card on the system, 1 ...
- */
- int getIndexIn() { return _indexIn; }
-
- /**
- * Get the index of the audio card for playback
- * @return int The index of the card used for playback
- * 0 for the first available card on the system, 1 ...
- */
- int getIndexOut() { return _indexOut; }
-
- /**
- * Get the sample rate of the audio layer
- * @return unsigned int The sample rate
- * default: 44100 HZ
- */
- unsigned int getSampleRate() { return _sampleRate; }
-
- /**
- * Get the frame size of the audio layer
- * @return unsigned int The frame size
- * default: 20 ms
- */
- unsigned int getFrameSize() { return _frameSize; }
-
- /**
- * Get the current state. Conversation or not
- * @return bool true if playSamples has been called
- * false otherwise
- */
- bool getCurrentState( void ) { return _talk; }
-
- int getLayerType( void ) { return _layerType; }
-
- protected:
-
- int _layerType;
-
- /**
- * Drop the pending frames and close the capture device
- */
- virtual void closeCaptureStream( void ) = 0;
-
- /**
- * Drop the pending frames and close the playback device
- */
- virtual void closePlaybackStream( void ) = 0;
-
- /** Augment coupling, reduce indirect access */
- ManagerImpl* _manager;
-
- /**
- * Urgent ring buffer used for ringtones
- */
- RingBuffer _urgentBuffer;
-
- /**
- * Determine if both endpoints hang up.
- * true if conversation is running
- * false otherwise
- */
- bool _talk;
-
- /**
- * Enable to determine if the devices are opened or not
- * true if the devices are closed
- * false otherwise
- */
- bool deviceClosed;
-
- /**
- * Number of audio cards on which capture stream has been opened
- */
- int _indexIn;
-
- /**
- * Number of audio cards on which playback stream has been opened
- */
- int _indexOut;
-
- /**
- * Sample Rate SFLphone should send sound data to the sound card
- * The value can be set in the user config file- now: 44100HZ
- */
- unsigned int _sampleRate;
-
- /**
- * Length of the sound frame we capture or read in ms
- * The value can be set in the user config file - now: 20ms
- */
- unsigned int _frameSize;
-
- /**
- * Input channel (mic) should be 1 mono
- */
- unsigned int _inChannel;
-
- /**
- * Output channel (stereo) should be 1 mono
- */
- unsigned int _outChannel;
-
- /** Contains the current error code */
- int _errorMessage;
-
- ost::Mutex _mutex;
+ /**
+ * Destructor
+ */
+ virtual ~AudioLayer(void) {}
+
+ virtual void closeLayer( void ) = 0;
+
+ /**
+ * Check if no devices are opened, otherwise close them.
+ * Then open the specified devices by calling the private functions open_device
+ * @param indexIn The number of the card choosen for capture
+ * @param indexOut The number of the card choosen for playback
+ * @param sampleRate The sample rate
+ * @param frameSize The frame size
+ * @param stream To indicate which kind of stream you want to open
+ * SFL_PCM_CAPTURE
+ * SFL_PCM_PLAYBACK
+ * SFL_PCM_BOTH
+ * @param plugin The alsa plugin ( dmix , default , front , surround , ...)
+ */
+ virtual bool openDevice(int indexIn, int indexOut, int sampleRate, int frameSize, int stream , std::string plugin) = 0;
+
+ /**
+ * Start the capture stream and prepare the playback stream.
+ * The playback starts accordingly to its threshold
+ * ALSA Library API
+ */
+ virtual void startStream(void) = 0;
+
+ /**
+ * Stop the playback and capture streams.
+ * Drops the pending frames and put the capture and playback handles to PREPARED state
+ * ALSA Library API
+ */
+ virtual void stopStream(void) = 0;
+
+ /**
+ * Query the capture device for number of bytes available in the hardware ring buffer
+ * @return int The number of bytes available
+ */
+ virtual int canGetMic() = 0;
+
+ /**
+ * Get data from the capture device
+ * @param buffer The buffer for data
+ * @param toCopy The number of bytes to get
+ * @return int The number of bytes acquired ( 0 if an error occured)
+ */
+ virtual int getMic(void * buffer, int toCopy) = 0;
+
+ /**
+ * Send a chunk of data to the hardware buffer to start the playback
+ * Copy data in the urgent buffer.
+ * @param buffer The buffer containing the data to be played ( ringtones )
+ * @param toCopy The size of the buffer
+ * @return int The number of bytes copied in the urgent buffer
+ */
+ int putUrgent(void* buffer, int toCopy);
+
+ /**
+ * Put voice data in the main sound buffer
+ * @param buffer The buffer containing the voice data ()
+ * @param toCopy The size of the buffer
+ * @return int The number of bytes copied
+ */
+ int putMain(void* buffer, int toCopy);
+
+ void flushMain (void);
+
+ void flushUrgent (void);
+
+ /**
+ * Flush the mic ringbuffer
+ */
+ void flushMic();
+
+ virtual bool isCaptureActive (void) = 0;
+
+ /**
+ * Write accessor to the error state
+ * @param error The error code
+ * Could be: ALSA_PLAYBACK_DEVICE
+ * ALSA_CAPTURE_DEVICE
+ */
+ void setErrorMessage(const int& error) { _errorMessage = error; }
+
+ /**
+ * Read accessor to the error state
+ * @return int The error code
+ */
+ int getErrorMessage() { return _errorMessage; }
+
+ /**
+ * Get the index of the audio card for capture
+ * @return int The index of the card used for capture
+ * 0 for the first available card on the system, 1 ...
+ */
+ int getIndexIn() { return _indexIn; }
+
+ /**
+ * Get the index of the audio card for playback
+ * @return int The index of the card used for playback
+ * 0 for the first available card on the system, 1 ...
+ */
+ int getIndexOut() { return _indexOut; }
+
+ /**
+ * Get the sample rate of the audio layer
+ * @return unsigned int The sample rate
+ * default: 44100 HZ
+ */
+ unsigned int getSampleRate() { return _sampleRate; }
+
+ /**
+ * Get the frame size of the audio layer
+ * @return unsigned int The frame size
+ * default: 20 ms
+ */
+ unsigned int getFrameSize() { return _frameSize; }
+
+ int getLayerType( void ) { return _layerType; }
+
+ /**
+ * Default volume for incoming RTP and Urgent sounds.
+ */
+ unsigned short _defaultVolume; // 100
+
+ protected:
+
+ int _layerType;
+
+ /**
+ * Drop the pending frames and close the capture device
+ */
+ virtual void closeCaptureStream( void ) = 0;
+
+ /**
+ * Drop the pending frames and close the playback device
+ */
+ virtual void closePlaybackStream( void ) = 0;
+
+ /** Augment coupling, reduce indirect access */
+ ManagerImpl* _manager;
+
+ /**
+ * Urgent ring buffer used for ringtones
+ */
+ RingBuffer _urgentRingBuffer;
+ RingBuffer _voiceRingBuffer;
+ RingBuffer _micRingBuffer;
+
+ /**
+ * Number of audio cards on which capture stream has been opened
+ */
+ int _indexIn;
+
+ /**
+ * Number of audio cards on which playback stream has been opened
+ */
+ int _indexOut;
+
+ /**
+ * Sample Rate SFLphone should send sound data to the sound card
+ * The value can be set in the user config file- now: 44100HZ
+ */
+ unsigned int _sampleRate;
+
+ /**
+ * Length of the sound frame we capture or read in ms
+ * The value can be set in the user config file - now: 20ms
+ */
+ unsigned int _frameSize;
+
+ /**
+ * Input channel (mic) should be 1 mono
+ */
+ unsigned int _inChannel;
+
+ /**
+ * Output channel (stereo) should be 1 mono
+ */
+ unsigned int _outChannel;
+
+ /** Contains the current error code */
+ int _errorMessage;
+
+ ost::Mutex _mutex;
};
#endif // _AUDIO_LAYER_H_
diff --git a/src/audio/audiortp.cpp b/src/audio/audiortp.cpp
index c2f3ec87673b1d73b58029c8215bd0cfc3a24b43..ed25feec437a4373e2dc787453a348e3e9bb510a 100644
--- a/src/audio/audiortp.cpp
+++ b/src/audio/audiortp.cpp
@@ -52,21 +52,22 @@ AudioRtp::~AudioRtp (void) {
int
AudioRtp::createNewSession (SIPCall *ca) {
- ost::MutexLock m(_threadMutex);
-
- // something should stop the thread before...
- if ( _RTXThread != 0 ) {
- _debug("! ARTP Failure: Thread already exists..., stopping it\n");
- delete _RTXThread; _RTXThread = 0;
- //return -1;
- }
+ ost::MutexLock m(_threadMutex);
+
+ // something should stop the thread before...
+ if ( _RTXThread != 0 ) {
+ _debug("**********************************************************\n");
+ _debug("! ARTP Failure: Thread already exists..., stopping it\n");
+ _debug("**********************************************************\n");
+ delete _RTXThread; _RTXThread = 0;
+ }
// Start RTP Send/Receive threads
_symmetric = Manager::instance().getConfigInt(SIGNALISATION,SYMMETRIC) ? true : false;
_RTXThread = new AudioRtpRTX (ca, _symmetric);
try {
if (_RTXThread->start() != 0) {
- _debug("! ARTP Failure: unable to start RTX Thread\n");
+ _debug("! ARTP Failure: unable to start RTX Thread\n");
return -1;
}
} catch(...) {
@@ -88,8 +89,8 @@ AudioRtp::closeRtpSession () {
_debugException("! ARTP Exception: when stopping audiortp\n");
throw;
}
- AudioLayer* audiolayer = Manager::instance().getAudioDriver();
- audiolayer->stopStream();
+ //AudioLayer* audiolayer = Manager::instance().getAudioDriver();
+ //audiolayer->stopStream();
}
////////////////////////////////////////////////////////////////////////////////
@@ -99,7 +100,7 @@ AudioRtpRTX::AudioRtpRTX (SIPCall *sipcall, bool sym) : time(new ost::Time()), _
_sym(sym), micData(NULL), micDataConverted(NULL), micDataEncoded(NULL), spkrDataDecoded(NULL), spkrDataConverted(NULL),
converter(NULL), _layerSampleRate(),_codecSampleRate(), _layerFrameSize(), _audiocodec(NULL)
{
- setCancel(cancelDeferred);
+ setCancel(cancelDefault);
// AudioRtpRTX should be close if we change sample rate
// TODO: Change bind address according to user settings.
// TODO: this should be the local ip not the external (router) IP
@@ -261,7 +262,7 @@ AudioRtpRTX::sendSessionFromMic(int timestamp)
int nb_sample_up = nbSample;
int nbSamplesMax = _layerFrameSize * _audiocodec->getClockRate() / 1000;
- //_debug("resample data\n");
+ //_debug("resample data = %i\n", nb_sample_up);
nbSample = reSampleData(_audiocodec->getClockRate(), nb_sample_up, DOWN_SAMPLING);
if ( nbSample < nbSamplesMax - 10 ) { // if only 10 is missing, it's ok
@@ -339,8 +340,10 @@ AudioRtpRTX::receiveSessionForSpkr (int& countTime)
#else
#endif
- audiolayer->playSamples( spkrDataConverted, nbSample * sizeof(SFLDataFormat), true);
+ //audiolayer->playSamples( spkrDataConverted, nbSample * sizeof(SFLDataFormat), true);
+ audiolayer->putMain (spkrDataConverted, nbSample * sizeof(SFLDataFormat));
+
// Notify (with a beep) an incoming call when there is already a call
countTime += time->getSecond();
if (Manager::instance().incomingCallWaiting() > 0) {
diff --git a/src/audio/pulselayer.cpp b/src/audio/pulselayer.cpp
index 98a6cd6aa54fcf9e3a25a0ae50eb1e263a9f53c5..45fb6edbae5810fec4c1b66935583532857327cf 100644
--- a/src/audio/pulselayer.cpp
+++ b/src/audio/pulselayer.cpp
@@ -32,14 +32,10 @@ static void audioCallback ( pa_stream* s, size_t bytes, void* userdata )
PulseLayer::PulseLayer(ManagerImpl* manager)
: AudioLayer( manager , PULSEAUDIO )
- , _mainSndRingBuffer( SIZEBUF )
- , _urgentRingBuffer( SIZEBUF)
- , _micRingBuffer( SIZEBUF )
, context(NULL)
, m(NULL)
, playback()
, record()
- , cache()
{
PulseLayer::streamState = 0;
_debug("Pulse audio constructor: Create context\n");
@@ -48,24 +44,31 @@ static void audioCallback ( pa_stream* s, size_t bytes, void* userdata )
// Destructor
PulseLayer::~PulseLayer (void)
{
- _debug(" Destroy pulselayer\n");
- delete playback;
- delete record;
- pa_context_disconnect(context);
- pa_context_unref( context );
+ //closeLayer();
+
+ /* Delete the pointer streams */
+ delete playback;
+ delete record;
+
+ pa_context_disconnect( context );
+ pa_context_unref( context );
+
+ sleep(2);
}
void
PulseLayer::closeLayer( void )
{
+ _debug(" Destroy pulselayer\n");
+
playback->disconnect();
record->disconnect();
while(PulseLayer::streamState != 2)
;
PulseLayer::streamState = 0;
- pa_context_disconnect( context );
- pa_context_unref( context );
+
+ //TODO Remove this ugly hack
sleep(2);
}
@@ -89,7 +92,7 @@ PulseLayer::connectPulseAudioServer( void )
}
pa_threaded_mainloop_unlock( m );
- serverinfo();
+ //serverinfo();
//muteAudioApps(99);
_debug("Context creation done\n");
}
@@ -139,7 +142,6 @@ PulseLayer::createStreams( pa_context* c )
record = new AudioStream(c, CAPTURE_STREAM, CAPTURE_STREAM_NAME , _manager->getMicVolume());
pa_stream_set_read_callback( record->pulseStream() , audioCallback, this);
//pa_stream_set_underflow_callback( record->pulseStream() , underflow , this);
- //cache = new AudioStream(c, UPLOAD_STREAM, "Cache samples", _manager->getSpkrVolume());
pa_threaded_mainloop_signal(m , 0);
}
@@ -179,37 +181,6 @@ PulseLayer::closePlaybackStream( void )
{
}
- int
-PulseLayer::playSamples(void* buffer, int toCopy, bool isTalking UNUSED)
-{
- int a = _mainSndRingBuffer.AvailForPut();
- if ( a >= toCopy ) {
- return _mainSndRingBuffer.Put(buffer, toCopy, 100);
- } else {
- _debug("Chopping sound, Ouch! RingBuffer full ?\n");
- return _mainSndRingBuffer.Put(buffer, a, 100);
- }
- return 0;
-}
-
- void
-PulseLayer::flushMain()
-{
- _mainSndRingBuffer.flush();
-}
-
- int
-PulseLayer::putUrgent(void* buffer, int toCopy)
-{
- int a = _urgentRingBuffer.AvailForPut();
- if ( a >= toCopy ) {
- return _urgentRingBuffer.Put(buffer, toCopy, 100 );
- } else {
- return _urgentRingBuffer.Put(buffer, a, 100 );
- }
- return 0;
-}
-
int
PulseLayer::canGetMic()
{
@@ -229,12 +200,6 @@ PulseLayer::getMic(void *buffer, int toCopy)
return 0;
}
- void
-PulseLayer::flushMic()
-{
- _micRingBuffer.flush();
-}
-
void
PulseLayer::startStream (void)
{
@@ -252,6 +217,8 @@ PulseLayer::stopStream (void)
pa_stream_flush( playback->pulseStream(), NULL, NULL );
pa_stream_flush( record->pulseStream(), NULL, NULL );
flushMic();
+ flushMain();
+ flushUrgent();
}
@@ -304,13 +271,13 @@ void PulseLayer::writeToSpeaker( void )
if (urgentAvail > 0) {
// Urgent data (dtmf, incoming call signal) come first.
- //_debug("Play urgent!: %i\n" , urgentAvail);
+ //_debug("Play urgent!: %i\e" , urgentAvail);
toGet = (urgentAvail < (int)(framesPerBuffer * sizeof(SFLDataFormat))) ? urgentAvail : framesPerBuffer * sizeof(SFLDataFormat);
out = (SFLDataFormat*)pa_xmalloc(toGet * sizeof(SFLDataFormat) );
_urgentRingBuffer.Get(out, toGet, 100);
pa_stream_write( playback->pulseStream() , out , toGet , pa_xfree, 0 , PA_SEEK_RELATIVE);
// Consume the regular one as well (same amount of bytes)
- _mainSndRingBuffer.Discard(toGet);
+ _voiceRingBuffer.Discard(toGet);
}
else
{
@@ -330,11 +297,11 @@ void PulseLayer::writeToSpeaker( void )
}
else {
out = (SFLDataFormat*)pa_xmalloc(framesPerBuffer * sizeof(SFLDataFormat));
- normalAvail = _mainSndRingBuffer.AvailForGet();
+ normalAvail = _voiceRingBuffer.AvailForGet();
toGet = (normalAvail < (int)(framesPerBuffer * sizeof(SFLDataFormat))) ? normalAvail : framesPerBuffer * sizeof(SFLDataFormat);
if (toGet) {
- _mainSndRingBuffer.Get(out, toGet, 100);
- _mainSndRingBuffer.Discard(toGet);
+ _voiceRingBuffer.Get(out, toGet, 100);
+ _voiceRingBuffer.Discard(toGet);
}
else {
bzero(out, framesPerBuffer * sizeof(SFLDataFormat));
@@ -364,15 +331,6 @@ void PulseLayer::readFromMic( void )
}
}
- int
-PulseLayer::putInCache( char code UNUSED, void *buffer UNUSED, int toCopy UNUSED )
-{
- _debug("Put the DTMF in cache\n");
- //pa_stream_write( cache->pulseStream() , buffer , toCopy , pa_xfree, 0 , PA_SEEK_RELATIVE);
- //pa_stream_finish_upload( cache->pulseStream() );
- return 1;
-}
-
static void retrieve_server_info(pa_context *c UNUSED, const pa_server_info *i, void *userdata UNUSED)
{
_debug("Server Info: Process owner : %s\n" , i->user_name);
diff --git a/src/audio/pulselayer.h b/src/audio/pulselayer.h
index dc22bfaedd968324b232946770ff06b7d08b5428..bdb34085b62e9abea85f27f4517b4f76b701e780 100644
--- a/src/audio/pulselayer.h
+++ b/src/audio/pulselayer.h
@@ -23,6 +23,8 @@
#include "audiolayer.h"
#include "audiostream.h"
+#include <pulse/pulseaudio.h>
+
#define PLAYBACK_STREAM_NAME "SFLphone out"
#define CAPTURE_STREAM_NAME "SFLphone in"
@@ -36,8 +38,6 @@ class PulseLayer : public AudioLayer {
void closeLayer( void );
- void trigger_thread(void){}
-
/**
* Check if no devices are opened, otherwise close them.
* Then open the specified devices by calling the private functions open_device
@@ -57,28 +57,6 @@ class PulseLayer : public AudioLayer {
void stopStream(void);
- /**
- * UNUSED in pulseaudio layer
- */
- bool isCaptureActive( void ) { return true; }
-
- /**
- * UNUSED in pulseaudio layer
- */
- bool isStreamActive (void);
-
- /**
- * Flush the main ringbuffer, reserved for the voice
- */
- void flushMain();
-
- int putUrgent(void* buffer, int toCopy);
-
- /**
- * UNUSED in pulseaudio layer
- */
- int putInCache( char code, void* buffer , int toCopy );
-
/**
* Query the capture device for number of bytes available in the hardware ring buffer
* @return int The number of bytes available
@@ -93,45 +71,21 @@ class PulseLayer : public AudioLayer {
*/
int getMic(void *, int);
- /**
- * Flush the mic ringbuffer
- */
- void flushMic();
-
- /**
- * UNUSED in pulseaudio layer
- */
- int playSamples(void* buffer, int toCopy, bool isTalking) ;
-
- //static void audioCallback ( pa_stream* s, size_t bytes, void* userdata );
static void overflow ( pa_stream* s, void* userdata );
static void underflow ( pa_stream* s, void* userdata );
static void stream_state_callback( pa_stream* s, void* user_data );
static void context_state_callback( pa_context* c, void* user_data );
- /**
- * UNUSED in pulseaudio layer
- */
- std::vector<std::string> getSoundCardsInfo( int stream UNUSED ) {
- std::vector<std::string> tmp;
- return tmp;
- }
+ bool isCaptureActive (void){return true;}
/**
* UNUSED in pulseaudio layer
*/
- bool soundCardIndexExist( int card UNUSED, int stream UNUSED ) { return true; }
-
- /**
- * UNUSED in pulseaudio layer
- */
- int soundCardGetIndex( std::string description UNUSED ) { return 0;}
+ //std::vector<std::string> getSoundCardsInfo( int stream UNUSED ) {
+ //std::vector<std::string> tmp;
+ //return tmp;
+ //}
- /**
- * UNUSED in pulseaudio layer
- */
- std::string getAudioPlugin( void ) { return "default"; }
-
/**
* Reduce volume of every audio applications connected to the same sink
*/
@@ -218,21 +172,6 @@ class PulseLayer : public AudioLayer {
*/
void serverinfo( void );
- /**
- * Ringbuffer for incoming voice data (playback)
- */
- RingBuffer _mainSndRingBuffer;
-
- /**
- * Ringbuffer for dtmf data
- */
- RingBuffer _urgentRingBuffer;
-
- /**
- * Ringbuffer for outgoing voice data (mic)
- */
- RingBuffer _micRingBuffer;
-
/** PulseAudio context and asynchronous loop */
pa_context* context;
pa_threaded_mainloop* m;
@@ -247,11 +186,6 @@ class PulseLayer : public AudioLayer {
*/
AudioStream* record;
- /**
- * A stream object to handle the pulseaudio upload stream
- */
- AudioStream* cache;
-
int spkrVolume;
int micVolume;
diff --git a/src/dbus/Makefile.am b/src/dbus/Makefile.am
index ee8355dbe90728906ece7ac3acf18622f5e9cecb..7e97ffc2dcd3771ed31fc020c23e1eb76b3bad0c 100644
--- a/src/dbus/Makefile.am
+++ b/src/dbus/Makefile.am
@@ -7,7 +7,6 @@ noinst_LTLIBRARIES = libdbus.la
libdbus_la_SOURCES = \
callmanager.cpp \
configurationmanager.cpp \
- contactmanager.cpp \
instance.cpp \
dbusmanagerimpl.cpp
@@ -20,8 +19,6 @@ noinst_HEADERS = \
callmanager.h \
configurationmanager.h \
configurationmanager-glue.h \
- contactmanager.h \
- contactmanager-glue.h \
instance.h \
instance-glue.h \
dbusmanager.h \
diff --git a/src/dbus/configurationmanager-glue.h b/src/dbus/configurationmanager-glue.h
index 292756cdb7e2c1c6b476b25432d1d56ac29322df..673a0f5f4efe0240bc7d5a90673ae6ac30ba8381 100644
--- a/src/dbus/configurationmanager-glue.h
+++ b/src/dbus/configurationmanager-glue.h
@@ -73,6 +73,10 @@ public:
register_method(ConfigurationManager_adaptor, getPulseAppVolumeControl, _getPulseAppVolumeControl_stub);
register_method(ConfigurationManager_adaptor, setSipPort, _setSipPort_stub);
register_method(ConfigurationManager_adaptor, getSipPort, _getSipPort_stub);
+ register_method(ConfigurationManager_adaptor, setStunServer, _setStunServer_stub);
+ register_method(ConfigurationManager_adaptor, getStunServer, _getStunServer_stub);
+ register_method(ConfigurationManager_adaptor, enableStun, _enableStun_stub);
+ register_method(ConfigurationManager_adaptor, isStunEnabled, _isStunEnabled_stub);
}
::DBus::IntrospectedInterface *const introspect() const
@@ -338,6 +342,25 @@ public:
{ "port", "i", false },
{ 0, 0, 0 }
};
+ static ::DBus::IntrospectedArgument setStunServer_args[] =
+ {
+ { "server", "s", true },
+ { 0, 0, 0 }
+ };
+ static ::DBus::IntrospectedArgument getStunServer_args[] =
+ {
+ { "server", "s", false },
+ { 0, 0, 0 }
+ };
+ static ::DBus::IntrospectedArgument enableStun_args[] =
+ {
+ { 0, 0, 0 }
+ };
+ static ::DBus::IntrospectedArgument isStunEnabled_args[] =
+ {
+ { "state", "i", false },
+ { 0, 0, 0 }
+ };
static ::DBus::IntrospectedArgument parametersChanged_args[] =
{
{ "list", "a{ss}", false },
@@ -407,6 +430,10 @@ public:
{ "getPulseAppVolumeControl", getPulseAppVolumeControl_args },
{ "setSipPort", setSipPort_args },
{ "getSipPort", getSipPort_args },
+ { "setStunServer", setStunServer_args },
+ { "getStunServer", getStunServer_args },
+ { "enableStun", enableStun_args },
+ { "isStunEnabled", isStunEnabled_args },
{ 0, 0 }
};
static ::DBus::IntrospectedMethod ConfigurationManager_adaptor_signals[] =
@@ -494,6 +521,10 @@ public:
virtual int32_t getPulseAppVolumeControl() = 0;
virtual void setSipPort(const int32_t& port) = 0;
virtual int32_t getSipPort() = 0;
+ virtual void setStunServer(const std::string& server) = 0;
+ virtual std::string getStunServer() = 0;
+ virtual void enableStun() = 0;
+ virtual int32_t isStunEnabled() = 0;
public:
@@ -1027,6 +1058,43 @@ private:
wi << argout1;
return reply;
}
+ ::DBus::Message _setStunServer_stub(const ::DBus::CallMessage &call)
+ {
+ ::DBus::MessageIter ri = call.reader();
+
+ std::string argin1; ri >> argin1;
+ setStunServer(argin1);
+ ::DBus::ReturnMessage reply(call);
+ return reply;
+ }
+ ::DBus::Message _getStunServer_stub(const ::DBus::CallMessage &call)
+ {
+ ::DBus::MessageIter ri = call.reader();
+
+ std::string argout1 = getStunServer();
+ ::DBus::ReturnMessage reply(call);
+ ::DBus::MessageIter wi = reply.writer();
+ wi << argout1;
+ return reply;
+ }
+ ::DBus::Message _enableStun_stub(const ::DBus::CallMessage &call)
+ {
+ ::DBus::MessageIter ri = call.reader();
+
+ enableStun();
+ ::DBus::ReturnMessage reply(call);
+ return reply;
+ }
+ ::DBus::Message _isStunEnabled_stub(const ::DBus::CallMessage &call)
+ {
+ ::DBus::MessageIter ri = call.reader();
+
+ int32_t argout1 = isStunEnabled();
+ ::DBus::ReturnMessage reply(call);
+ ::DBus::MessageIter wi = reply.writer();
+ wi << argout1;
+ return reply;
+ }
};
} } }
diff --git a/src/dbus/configurationmanager-introspec.xml b/src/dbus/configurationmanager-introspec.xml
index df72d96e79c6ecf77b4c47ca27c2a4a90b94106c..0990b1443c15a7f53cb2bfee5388885a67e48697 100644
--- a/src/dbus/configurationmanager-introspec.xml
+++ b/src/dbus/configurationmanager-introspec.xml
@@ -218,6 +218,21 @@
<arg type="i" name="port" direction="out"/>
</method>
+ <method name="setStunServer">
+ <arg type="s" name="server" direction="in"/>
+ </method>
+
+ <method name="getStunServer">
+ <arg type="s" name="server" direction="out"/>
+ </method>
+
+ <method name="enableStun">
+ </method>
+
+ <method name="isStunEnabled">
+ <arg type="i" name="state" direction="out"/>
+ </method>
+
<!-- ///////////////////////////// -->
<signal name="parametersChanged">
<arg type="a{ss}" name="list" direction="out"/>
diff --git a/src/dbus/configurationmanager.cpp b/src/dbus/configurationmanager.cpp
index 547afded01e9912d0e65923bf0ce193ba8830cce..8c1e5af8f5712bae4224d5942a77a6da9d08247e 100644
--- a/src/dbus/configurationmanager.cpp
+++ b/src/dbus/configurationmanager.cpp
@@ -395,3 +395,22 @@ ConfigurationManager::setSipPort( const int32_t& portNum )
Manager::instance().setSipPort(portNum);
}
+std::string ConfigurationManager::getStunServer( void )
+{
+ return Manager::instance().getStunServer();
+}
+
+void ConfigurationManager::setStunServer( const std::string& server )
+{
+ Manager::instance().setStunServer( server );
+}
+
+void ConfigurationManager::enableStun (void)
+{
+ Manager::instance().enableStun();
+}
+
+int32_t ConfigurationManager::isStunEnabled (void)
+{
+ return Manager::instance().isStunEnabled();
+}
diff --git a/src/dbus/configurationmanager.h b/src/dbus/configurationmanager.h
index 86c6ee7dec89ae7352a8ea423ea401cfa4edab8f..20f01317d075b974fe8a85e9927976b854e155a2 100644
--- a/src/dbus/configurationmanager.h
+++ b/src/dbus/configurationmanager.h
@@ -96,7 +96,11 @@ public:
int32_t getPulseAppVolumeControl( void );
void setPulseAppVolumeControl( void );
int32_t getSipPort( void );
- void setSipPort( const int32_t& portNum );
+ void setSipPort( const int32_t& portNum);
+ std::string getStunServer( void );
+ void setStunServer( const std::string& server );
+ void enableStun (void);
+ int32_t isStunEnabled (void);
};
diff --git a/src/eventthread.cpp b/src/eventthread.cpp
index 66e6d441090be9d70a785ecc7fa0ff44a5256b44..b0378c57ec651746988f822f6206254ac6bec2bd 100644
--- a/src/eventthread.cpp
+++ b/src/eventthread.cpp
@@ -1,8 +1,7 @@
/*
- * Copyright (C) 2004, 2005, 2006 Savoir-Faire Linux inc.
- * Author: Yan Morin <yan.morin@savoirfairelinux.com>
- * Author : Laurielle Lea <laurielle.lea@savoirfairelinux.com>
- *
+ * Copyright (C) 2009 Savoir-Faire Linux inc.
+ * Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
+ *
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 3 of the License, or
@@ -20,44 +19,41 @@
#include "eventthread.h"
#include "voiplink.h"
+#include "audio/alsalayer.h"
-EventThread::EventThread (VoIPLink* link) : Thread (), _linkthread(link), stopIt(false)
+/********************************** Voiplink thread *************************************/
+EventThread::EventThread( VoIPLink *link )
+ : Thread(), _linkthread(link)
{
- setCancel(cancelDeferred);
+ setCancel( cancelDeferred );
}
-EventThread::~EventThread (void)
-{
- terminate();
-}
/**
* Reimplementation of run()
*/
-void
-EventThread::run (void)
+void EventThread::run (void)
{
- //stopIt = false;
while(!testCancel()) {
_linkthread->getEvent();
}
-}
+}
-void
-EventThread::stop( void )
-{
- stopIt = true;
-}
+/********************************************************************************************/
-void
-EventThread::startLoop( void )
+AudioThread::AudioThread (AlsaLayer *alsa)
+ : Thread(), _alsa(alsa)
{
- stopIt = false;
- //start();
+ setCancel (cancelDeferred);
}
-bool
-EventThread::isStopped( void )
+/**
+ * Reimplementation of run()
+ */
+void AudioThread::run (void)
{
- return stopIt;
+ while(!testCancel()) {
+ _alsa->audioCallback();
+ }
}
+
diff --git a/src/eventthread.h b/src/eventthread.h
index 8772c72043fec8d719b7c50aecf706dbdb241396..0d631ef066254593e976e1837b1f1e54c2df7594 100644
--- a/src/eventthread.h
+++ b/src/eventthread.h
@@ -1,7 +1,6 @@
/*
- * Copyright (C) 2004-2005 Savoir-Faire Linux inc.
- * Author: Yan Morin <yan.morin@savoirfairelinux.com>
- * Author : Laurielle Lea <laurielle.lea@savoirfairelinux.com>
+ * Copyright (C) 2009 Savoir-Faire Linux inc.
+ * Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
@@ -24,6 +23,7 @@
#include <cc++/thread.h>
class VoIPLink;
+class AlsaLayer;
/**
* @file eventthread.h
@@ -31,24 +31,43 @@ class VoIPLink;
*/
class EventThread : public ost::Thread {
-public:
- /**
- * Build a thread that call getEvents
- */
- EventThread (VoIPLink*);
- ~EventThread (void);
- virtual void run ();
- virtual void stop();
- virtual void startLoop();
- bool isStopped();
-
-private:
+
+ public:
+ /**
+ * Thread constructor
+ */
+ EventThread (VoIPLink* link);
+
+ ~EventThread (void){
+ terminate();
+ }
+
+ virtual void run () ;
+
+ private:
EventThread(const EventThread& rh); // copy constructor
EventThread& operator=(const EventThread& rh); // assignment operator
/** VoIPLink is the object being called by getEvents() method */
- VoIPLink* _linkthread;
- bool stopIt;
+ VoIPLink* _linkthread;
+};
+
+class AudioThread : public ost::Thread {
+
+ public:
+ AudioThread (AlsaLayer *alsa);
+
+ ~AudioThread (void){
+ terminate();
+ }
+
+ virtual void run (void);
+
+ private:
+ AudioThread (const AudioThread& at);
+ AudioThread& operator=(const AudioThread& at);
+
+ AlsaLayer* _alsa;
};
#endif // __EVENT_THREAD_H__
diff --git a/src/global.h b/src/global.h
index e752d41b2f5ee4be78392cd0dd67308e19bf2b93..45332563dc0bbb28a0f22a777d997deafa5c32f5 100644
--- a/src/global.h
+++ b/src/global.h
@@ -26,6 +26,11 @@
#include <stdio.h>
#include <libintl.h>
#include <locale.h>
+#include <string>
+#include <stdlib.h>
+#include <sstream>
+#include <map>
+#include <vector>
#define SFLPHONED_VERSION "0.9.2-4" /** Version number */
@@ -98,6 +103,7 @@ typedef short int16;
#define WINDOW_POPUP 1 /** Popup mode */
#define NOTIFY_ALL 1 /** Desktop notification level 0: never notify */
#define NOTIFY_MAILS 1 /** Desktop mail notification level 0: never notify */
+#define STUN_ENABLED 1
// Error codes for error handling
#define NO_ERROR 0x0000 /** No error - Everything alright */
diff --git a/src/iaxaccount.cpp b/src/iaxaccount.cpp
index 364f0ab31b028944112bc177ce85e2b45e6e3b81..504a1cc7106dfa7d1c441a64b7a186fc584152bd 100644
--- a/src/iaxaccount.cpp
+++ b/src/iaxaccount.cpp
@@ -22,8 +22,8 @@
#include "iaxaccount.h"
#include "iaxvoiplink.h"
- IAXAccount::IAXAccount(const AccountID& accountID)
-: Account(accountID)
+IAXAccount::IAXAccount(const AccountID& accountID)
+ : Account(accountID, "iax2")
{
_link = new IAXVoIPLink(accountID);
}
@@ -35,22 +35,16 @@ IAXAccount::~IAXAccount()
_link = NULL;
}
- int
-IAXAccount::registerVoIPLink()
+int IAXAccount::registerVoIPLink()
{
- IAXVoIPLink *thislink;
-
_link->init();
- thislink = dynamic_cast<IAXVoIPLink*> (_link);
- if (thislink) {
- // Stuff needed for IAX registration
- thislink->setHost(Manager::instance().getConfigString(_accountID, HOSTNAME));
- thislink->setUser(Manager::instance().getConfigString(_accountID, USERNAME));
- thislink->setPass(Manager::instance().getConfigString(_accountID, PASSWORD));
- }
+ // Stuff needed for IAX registration
+ setHostname(Manager::instance().getConfigString(_accountID, HOSTNAME));
+ setUsername(Manager::instance().getConfigString(_accountID, USERNAME));
+ setPassword(Manager::instance().getConfigString(_accountID, PASSWORD));
- _link->sendRegister();
+ _link->sendRegister( _accountID );
return SUCCESS;
}
@@ -58,7 +52,7 @@ IAXAccount::registerVoIPLink()
int
IAXAccount::unregisterVoIPLink()
{
- _link->sendUnregister();
+ _link->sendUnregister( _accountID );
_link->terminate();
return SUCCESS;
diff --git a/src/iaxvoiplink.cpp b/src/iaxvoiplink.cpp
index 6d04dffcaeae225d5d75207136262f3a36705b1c..f085ea1544981952c128c736071261fca0d7ceed 100644
--- a/src/iaxvoiplink.cpp
+++ b/src/iaxvoiplink.cpp
@@ -18,14 +18,12 @@
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#include "iaxvoiplink.h"
-#include "global.h" // for _debug
#include "iaxcall.h"
#include "eventthread.h"
-
+#include "iaxaccount.h"
#include "manager.h"
#include "audio/audiolayer.h"
-//#include <iax/iax-client.h>
#include <math.h>
#include <dlfcn.h>
@@ -40,182 +38,182 @@
#define MUSIC_ONHOLD true
#define CHK_VALID_CALL if (call == NULL) { _debug("IAX: Call doesn't exists\n"); \
- return false; }
+ return false; }
- IAXVoIPLink::IAXVoIPLink(const AccountID& accountID)
+ IAXVoIPLink::IAXVoIPLink(const AccountID& accountID)
: VoIPLink(accountID)
{
- _evThread = new EventThread(this);
- _regSession = NULL;
- _nextRefreshStamp = 0;
+ _evThread = new EventThread(this);
+ _regSession = NULL;
+ _nextRefreshStamp = 0;
- // to get random number for RANDOM_PORT
- srand (time(NULL));
+ // to get random number for RANDOM_PORT
+ srand (time(NULL));
- audiolayer = NULL;
+ audiolayer = NULL;
- converter = new SamplerateConverter();
+ converter = new SamplerateConverter();
- int nbSamplesMax = (int) ( converter->getFrequence() * converter->getFramesize() / 1000 );
+ int nbSamplesMax = (int) ( converter->getFrequence() * converter->getFramesize() / 1000 );
- micData = new SFLDataFormat[nbSamplesMax];
- micDataConverted = new SFLDataFormat[nbSamplesMax];
- micDataEncoded = new unsigned char[nbSamplesMax];
+ micData = new SFLDataFormat[nbSamplesMax];
+ micDataConverted = new SFLDataFormat[nbSamplesMax];
+ micDataEncoded = new unsigned char[nbSamplesMax];
- spkrDataConverted = new SFLDataFormat[nbSamplesMax];
- spkrDataDecoded = new SFLDataFormat[nbSamplesMax];
+ spkrDataConverted = new SFLDataFormat[nbSamplesMax];
+ spkrDataDecoded = new SFLDataFormat[nbSamplesMax];
}
IAXVoIPLink::~IAXVoIPLink()
{
- delete _evThread; _evThread = NULL;
- _regSession = NULL; // shall not delete it
- terminate();
+ delete _evThread; _evThread = NULL;
+ _regSession = NULL; // shall not delete it
+ terminate();
- audiolayer = NULL;
+ audiolayer = NULL;
- delete [] micData; micData = NULL;
- delete [] micDataConverted; micDataConverted = NULL;
- delete [] micDataEncoded; micDataEncoded = NULL;
+ delete [] micData; micData = NULL;
+ delete [] micDataConverted; micDataConverted = NULL;
+ delete [] micDataEncoded; micDataEncoded = NULL;
- delete [] spkrDataDecoded; spkrDataDecoded = NULL;
- delete [] spkrDataConverted; spkrDataConverted = NULL;
+ delete [] spkrDataDecoded; spkrDataDecoded = NULL;
+ delete [] spkrDataConverted; spkrDataConverted = NULL;
}
- bool
+ bool
IAXVoIPLink::init()
{
- // If it was done, don't do it again, until we call terminate()
- if (_initDone)
- return false;
-
- bool returnValue = false;
- //_localAddress = "127.0.0.1";
- // port 0 is default
- // iax_enable_debug(); have to enable debug when compiling iax...
- int port = IAX_DEFAULT_PORTNO;
- int last_port = 0;
- int nbTry = 3;
-
- while (port != IAX_FAILURE && nbTry) {
- last_port = port;
- port = iax_init(port);
- if ( port < 0 ) {
- _debug("IAX Warning: already initialize on port %d\n", last_port);
- port = RANDOM_IAX_PORT;
- } else if (port == IAX_FAILURE) {
- _debug("IAX Fail to start on port %d", last_port);
- port = RANDOM_IAX_PORT;
- } else {
- _debug("IAX Info: listening on port %d\n", last_port);
- _localPort = last_port;
- returnValue = true;
- _evThread->start();
-
- audiolayer = Manager::instance().getAudioDriver();
- break;
+ // If it was done, don't do it again, until we call terminate()
+ if (initDone())
+ return false;
+
+ bool returnValue = false;
+ //_localAddress = "127.0.0.1";
+ // port 0 is default
+ // iax_enable_debug(); have to enable debug when compiling iax...
+ int port = IAX_DEFAULT_PORTNO;
+ int last_port = 0;
+ int nbTry = 3;
+
+ while (port != IAX_FAILURE && nbTry) {
+ last_port = port;
+ port = iax_init(port);
+ if ( port < 0 ) {
+ _debug("IAX Warning: already initialize on port %d\n", last_port);
+ port = RANDOM_IAX_PORT;
+ } else if (port == IAX_FAILURE) {
+ _debug("IAX Fail to start on port %d", last_port);
+ port = RANDOM_IAX_PORT;
+ } else {
+ _debug("IAX Info: listening on port %d\n", last_port);
+ _localPort = last_port;
+ returnValue = true;
+ _evThread->start();
+
+ audiolayer = Manager::instance().getAudioDriver();
+ break;
+ }
+ nbTry--;
+
+ initDone(true);
}
- nbTry--;
-
- _initDone = true;
- }
- if (port == IAX_FAILURE || nbTry==0) {
- _debug("Fail to initialize iax\n");
+ if (port == IAX_FAILURE || nbTry==0) {
+ _debug("Fail to initialize iax\n");
- _initDone = false;
- }
+ initDone(false);
+ }
- return returnValue;
+ return returnValue;
}
- void
+ void
IAXVoIPLink::terminate()
{
- // If it was done, don't do it again, until we call init()
- if (!_initDone)
- return;
+ // If it was done, don't do it again, until we call init()
+ if (!initDone())
+ return;
- // iaxc_shutdown();
+ // iaxc_shutdown();
- // Hangup all calls
- terminateIAXCall();
+ // Hangup all calls
+ terminateIAXCall();
- _initDone = false;
+ initDone(false);
}
- void
+ void
IAXVoIPLink::terminateIAXCall()
{
- std::string reason = "Dumped Call";
- ost::MutexLock m(_callMapMutex);
- CallMap::iterator iter = _callMap.begin();
- IAXCall *call;
- while( iter != _callMap.end() ) {
- call = dynamic_cast<IAXCall*>(iter->second);
- if (call) {
- _mutexIAX.enterMutex();
- iax_hangup(call->getSession(), (char*)reason.c_str());
- _mutexIAX.leaveMutex();
- call->setSession(NULL);
- delete call; call = NULL;
+ std::string reason = "Dumped Call";
+ ost::MutexLock m(_callMapMutex);
+ CallMap::iterator iter = _callMap.begin();
+ IAXCall *call;
+ while( iter != _callMap.end() ) {
+ call = dynamic_cast<IAXCall*>(iter->second);
+ if (call) {
+ _mutexIAX.enterMutex();
+ iax_hangup(call->getSession(), (char*)reason.c_str());
+ _mutexIAX.leaveMutex();
+ call->setSession(NULL);
+ delete call; call = NULL;
+ }
+ iter++;
}
- iter++;
- }
- _callMap.clear();
+ _callMap.clear();
}
- void
+ void
IAXVoIPLink::getEvent()
{
- IAXCall* call = NULL;
-
- // lock iax_ stuff..
- _mutexIAX.enterMutex();
- iax_event* event = NULL;
+ IAXCall* call = NULL;
- while ( (event = iax_get_event(IAX_NONBLOCKING)) != NULL ) {
- // If we received an 'ACK', libiax2 tells apps to ignore them.
- if (event->etype == IAX_EVENT_NULL) {
- continue;
+ // lock iax_ stuff..
+ _mutexIAX.enterMutex();
+ iax_event* event = NULL;
+
+ while ( (event = iax_get_event(IAX_NONBLOCKING)) != NULL ) {
+ // If we received an 'ACK', libiax2 tells apps to ignore them.
+ if (event->etype == IAX_EVENT_NULL) {
+ continue;
+ }
+
+ //_debug ("Receive IAX Event: %d (0x%x)\n", event->etype, event->etype);
+
+ call = iaxFindCallBySession(event->session);
+
+ if (call) {
+ // We know that call, deal with it
+ iaxHandleCallEvent(event, call);
+ //_audiocodec = Manager::instance().getCodecDescriptorMap().getCodec( call -> getAudioCodec() );
+ }
+ else if (event->session && event->session == _regSession) {
+ // This is a registration session, deal with it
+ iaxHandleRegReply(event);
+ }
+ else {
+ // We've got an event before it's associated with any call
+ iaxHandlePrecallEvent(event);
+ }
+
+ iax_event_free(event);
}
+ _mutexIAX.leaveMutex();
- //_debug ("Receive IAX Event: %d (0x%x)\n", event->etype, event->etype);
-
- call = iaxFindCallBySession(event->session);
+ // Do the doodle-moodle to send audio from the microphone to the IAX channel.
+ sendAudioFromMic();
- if (call) {
- // We know that call, deal with it
- iaxHandleCallEvent(event, call);
- //_audiocodec = Manager::instance().getCodecDescriptorMap().getCodec( call -> getAudioCodec() );
- }
- else if (event->session && event->session == _regSession) {
- // This is a registration session, deal with it
- iaxHandleRegReply(event);
+ // Refresh registration.
+ if (_nextRefreshStamp && _nextRefreshStamp - 2 < time(NULL)) {
+ sendRegister("");
}
- else {
- // We've got an event before it's associated with any call
- iaxHandlePrecallEvent(event);
- }
-
- iax_event_free(event);
- }
- _mutexIAX.leaveMutex();
- // Do the doodle-moodle to send audio from the microphone to the IAX channel.
- sendAudioFromMic();
-
- // Refresh registration.
- if (_nextRefreshStamp && _nextRefreshStamp - 2 < time(NULL)) {
- sendRegister();
- }
-
- // thread wait 3 millisecond
- _evThread->sleep(3);
- free(event);
+ // thread wait 3 millisecond
+ _evThread->sleep(3);
+ free(event);
}
- void
+ void
IAXVoIPLink::sendAudioFromMic(void)
{
int maxBytesToGet, availBytesFromMic, bytesAvail, nbSample, compSize;
@@ -226,7 +224,7 @@ IAXVoIPLink::sendAudioFromMic(void)
updateAudiolayer();
IAXCall* currentCall = getIAXCall(Manager::instance().getCurrentCallId());
-
+
if (!currentCall) {
// Let's mind our own business.
return;
@@ -235,546 +233,542 @@ IAXVoIPLink::sendAudioFromMic(void)
if( currentCall -> getAudioCodec() < 0 )
return;
- // Just make sure the currentCall is in state to receive audio right now.
- //_debug("Here we get: connectionState: %d state: %d \n",
- //currentCall->getConnectionState(),
- //currentCall->getState());
+ // Just make sure the currentCall is in state to receive audio right now.
+ //_debug("Here we get: connectionState: %d state: %d \n",
+ //currentCall->getConnectionState(),
+ //currentCall->getState());
- if (currentCall->getConnectionState() != Call::Connected ||
- currentCall->getState() != Call::Active) {
- return;
- }
+ if (currentCall->getConnectionState() != Call::Connected ||
+ currentCall->getState() != Call::Active) {
+ return;
+ }
- ac = currentCall->getCodecMap().getCodec( currentCall -> getAudioCodec() );
- if (!ac) {
- // Audio codec still not determined.
- if (audiolayer) {
- // To keep latency low..
- //audiolayer->flushMic();
+ ac = currentCall->getCodecMap().getCodec( currentCall -> getAudioCodec() );
+ if (!ac) {
+ // Audio codec still not determined.
+ if (audiolayer) {
+ // To keep latency low..
+ //audiolayer->flushMic();
+ }
+ return;
}
- return;
- }
- // Send sound here
- if (audiolayer) {
+ // Send sound here
+ if (audiolayer) {
- // we have to get 20ms of data from the mic *20/1000 = /50
- // rate/50 shall be lower than IAX__20S_48KHZ_MAX
- maxBytesToGet = audiolayer->getSampleRate()* audiolayer->getFrameSize() / 1000 * sizeof(SFLDataFormat);
-
- // available bytes inside ringbuffer
- availBytesFromMic = audiolayer->canGetMic();
+ // we have to get 20ms of data from the mic *20/1000 = /50
+ // rate/50 shall be lower than IAX__20S_48KHZ_MAX
+ maxBytesToGet = audiolayer->getSampleRate()* audiolayer->getFrameSize() / 1000 * sizeof(SFLDataFormat);
- if (availBytesFromMic < maxBytesToGet) {
- // We need packets full!
- return;
- }
+ // available bytes inside ringbuffer
+ availBytesFromMic = audiolayer->canGetMic();
- // take the lowest
- bytesAvail = (availBytesFromMic < maxBytesToGet) ? availBytesFromMic : maxBytesToGet;
- //_debug("available = %d, maxBytesToGet = %d\n", availBytesFromMic, maxBytesToGet);
+ if (availBytesFromMic < maxBytesToGet) {
+ // We need packets full!
+ return;
+ }
- // Get bytes from micRingBuffer to data_from_mic
- nbSample = audiolayer->getMic( micData, bytesAvail ) / sizeof(SFLDataFormat);
+ // take the lowest
+ bytesAvail = (availBytesFromMic < maxBytesToGet) ? availBytesFromMic : maxBytesToGet;
+ //_debug("available = %d, maxBytesToGet = %d\n", availBytesFromMic, maxBytesToGet);
- // resample
- nbSample = converter->downsampleData( micData , micDataConverted , (int)ac ->getClockRate() , (int)audiolayer->getSampleRate() , nbSample );
+ // Get bytes from micRingBuffer to data_from_mic
+ nbSample = audiolayer->getMic( micData, bytesAvail ) / sizeof(SFLDataFormat);
- // for the mono: range = 0 to IAX_FRAME2SEND * sizeof(int16)
- compSize = ac->codecEncode( micDataEncoded, micDataConverted , nbSample*sizeof(int16));
+ // resample
+ nbSample = converter->downsampleData( micData , micDataConverted , (int)ac ->getClockRate() , (int)audiolayer->getSampleRate() , nbSample );
- // Send it out!
- _mutexIAX.enterMutex();
- // Make sure the session and the call still exists.
- if (currentCall->getSession()) {
- if ( iax_send_voice(currentCall->getSession(), currentCall->getFormat(), micDataEncoded, compSize, nbSample) == -1) {
- _debug("IAX: Error sending voice data.\n");
- }
+ // for the mono: range = 0 to IAX_FRAME2SEND * sizeof(int16)
+ compSize = ac->codecEncode( micDataEncoded, micDataConverted , nbSample*sizeof(int16));
+
+ // Send it out!
+ _mutexIAX.enterMutex();
+ // Make sure the session and the call still exists.
+ if (currentCall->getSession()) {
+ if (iax_send_voice(currentCall->getSession(), currentCall->getFormat(), micDataEncoded, compSize, nbSample) == -1) {
+ _debug("IAX: Error sending voice data.\n");
+ }
+ }
+ _mutexIAX.leaveMutex();
}
- _mutexIAX.leaveMutex();
- }
}
- IAXCall*
+ IAXCall*
IAXVoIPLink::getIAXCall(const CallID& id)
{
- Call* call = getCall(id);
- if (call) {
- return dynamic_cast<IAXCall*>(call);
- }
- return NULL;
+ Call* call = getCall(id);
+ if (call) {
+ return dynamic_cast<IAXCall*>(call);
+ }
+ return NULL;
}
- int
-IAXVoIPLink::sendRegister()
+ int
+IAXVoIPLink::sendRegister(AccountID id)
{
- bool result = false;
- if (_host.empty()) {
- return false;
- }
- if (_user.empty()) {
- return false;
- }
-
- // lock
- _mutexIAX.enterMutex();
-
- // Always use a brand new session
- if (_regSession) {
- iax_destroy(_regSession);
- }
-
- _regSession = iax_session_new();
-
- if (!_regSession) {
- _debug("Error when generating new session for register");
- } else {
- // refresh
- // last reg
- char host[_host.length()+1];
- strcpy(host, _host.c_str());
- char user[_user.length()+1];
- strcpy(user, _user.c_str());
- char pass[_pass.length()+1];
- strcpy(pass, _pass.c_str());
- // iax_register doesn't use const char*
-
- _debug("IAX Sending registration to %s with user %s\n", host, user);
- int val = iax_register(_regSession, host, user, pass, 120);
- _debug ("Return value: %d\n", val);
- // set the time-out to 15 seconds, after that, resend a registration request.
- // until we unregister.
- _nextRefreshStamp = time(NULL) + 10;
- result = true;
-
- setRegistrationState(Trying);
- }
-
- // unlock
- _mutexIAX.leaveMutex();
-
- return result;
-}
+ IAXAccount *account;
+ bool result;
+
+ result = false;
+ account = dynamic_cast<IAXAccount *> (getAccountPtr());
+ if (account->getHostname().empty()) {
+ return false;
+ }
+ if (account->getUsername().empty()) {
+ return false;
+ }
+
+ // lock
+ _mutexIAX.enterMutex();
+ // Always use a brand new session
+ if (_regSession) {
+ iax_destroy(_regSession);
+ }
+ _regSession = iax_session_new();
- int
-IAXVoIPLink::sendUnregister()
+ if (!_regSession) {
+ _debug("Error when generating new session for register");
+ } else {
+ _debug("IAX Sending registration to %s with user %s\n", account->getHostname().c_str() , account->getUsername().c_str() );
+ int val = iax_register(_regSession, account->getHostname().data(), account->getUsername().data(), account->getPassword().data(), 120);
+ _debug ("Return value: %d\n", val);
+ // set the time-out to 15 seconds, after that, resend a registration request.
+ // until we unregister.
+ _nextRefreshStamp = time(NULL) + 10;
+ result = true;
+
+ account->setRegistrationState(Trying);
+ }
+
+ // unlock
+ _mutexIAX.leaveMutex();
+
+ return result;
+}
+
+ int
+IAXVoIPLink::sendUnregister(AccountID id)
{
- _mutexIAX.enterMutex();
- if (_regSession) {
- /** @todo Should send a REGREL in sendUnregister()... */
+ IAXAccount *account;
- //iax_send_regrel(); doesn't exist yet :)
- iax_destroy(_regSession);
+ account = dynamic_cast<IAXAccount*>(getAccountPtr());
+ if(!account)
+ return 1;
- _regSession = NULL;
- }
- _mutexIAX.leaveMutex();
+ _mutexIAX.enterMutex();
+ if (_regSession) {
+ /** @todo Should send a REGREL in sendUnregister()... */
+ //iax_send_regrel(); doesn't exist yet :)
+ iax_destroy(_regSession);
+ _regSession = NULL;
+ }
+ _mutexIAX.leaveMutex();
- _nextRefreshStamp = 0;
+ _nextRefreshStamp = 0;
- _debug("IAX2 send unregister\n");
- setRegistrationState(Unregistered);
+ _debug("IAX2 send unregister\n");
+ account->setRegistrationState(Unregistered);
- return SUCCESS;
+ return SUCCESS;
}
- Call*
+ Call*
IAXVoIPLink::newOutgoingCall(const CallID& id, const std::string& toUrl)
{
- IAXCall* call = new IAXCall(id, Call::Outgoing);
- call->setCodecMap(Manager::instance().getCodecDescriptorMap());
-
+ IAXCall* call = new IAXCall(id, Call::Outgoing);
+ call->setCodecMap(Manager::instance().getCodecDescriptorMap());
- if (call) {
- call->setPeerNumber(toUrl);
-
- if ( iaxOutgoingInvite(call) ) {
- call->setConnectionState(Call::Progressing);
- call->setState(Call::Active);
- addCall(call);
- } else {
- delete call; call = NULL;
+ if (call) {
+ call->setPeerNumber(toUrl);
+
+ if ( iaxOutgoingInvite(call) ) {
+ call->setConnectionState(Call::Progressing);
+ call->setState(Call::Active);
+ addCall(call);
+ } else {
+ delete call; call = NULL;
+ }
}
- }
- return call;
+ return call;
}
- bool
+ bool
IAXVoIPLink::answer(const CallID& id)
{
- IAXCall* call = getIAXCall(id);
- call->setCodecMap(Manager::instance().getCodecDescriptorMap());
+ IAXCall* call = getIAXCall(id);
+ call->setCodecMap(Manager::instance().getCodecDescriptorMap());
- CHK_VALID_CALL;
+ CHK_VALID_CALL;
- _mutexIAX.enterMutex();
- iax_answer(call->getSession());
- _mutexIAX.leaveMutex();
+ _mutexIAX.enterMutex();
+ iax_answer(call->getSession());
+ _mutexIAX.leaveMutex();
- call->setState(Call::Active);
- call->setConnectionState(Call::Connected);
- // Start audio
- audiolayer->startStream();
- //audiolayer->flushMic();
+ call->setState(Call::Active);
+ call->setConnectionState(Call::Connected);
+ // Start audio
+ audiolayer->startStream();
+ //audiolayer->flushMic();
- return true;
+ return true;
}
- bool
+ bool
IAXVoIPLink::hangup(const CallID& id)
{
- IAXCall* call = getIAXCall(id);
- std::string reason = "Dumped Call";
- CHK_VALID_CALL;
-
- _mutexIAX.enterMutex();
- iax_hangup(call->getSession(), (char*) reason.c_str());
- _mutexIAX.leaveMutex();
- call->setSession(NULL);
- if (Manager::instance().isCurrentCall(id)) {
- // stop audio
- audiolayer->stopStream();
- }
- removeCall(id);
- return true;
+ IAXCall* call = getIAXCall(id);
+ std::string reason = "Dumped Call";
+ CHK_VALID_CALL;
+
+ _mutexIAX.enterMutex();
+ iax_hangup(call->getSession(), (char*) reason.c_str());
+ _mutexIAX.leaveMutex();
+ call->setSession(NULL);
+ if (Manager::instance().isCurrentCall(id)) {
+ // stop audio
+ audiolayer->stopStream();
+ }
+ removeCall(id);
+ return true;
}
- bool
+ bool
IAXVoIPLink::onhold(const CallID& id)
{
- IAXCall* call = getIAXCall(id);
+ IAXCall* call = getIAXCall(id);
- CHK_VALID_CALL;
+ CHK_VALID_CALL;
- //if (call->getState() == Call::Hold) { _debug("Call is already on hold\n"); return false; }
+ //if (call->getState() == Call::Hold) { _debug("Call is already on hold\n"); return false; }
- _mutexIAX.enterMutex();
- iax_quelch_moh(call->getSession() , MUSIC_ONHOLD);
- _mutexIAX.leaveMutex();
+ _mutexIAX.enterMutex();
+ iax_quelch_moh(call->getSession() , MUSIC_ONHOLD);
+ _mutexIAX.leaveMutex();
- call->setState(Call::Hold);
- return true;
+ call->setState(Call::Hold);
+ return true;
}
- bool
+ bool
IAXVoIPLink::offhold(const CallID& id)
{
- IAXCall* call = getIAXCall(id);
+ IAXCall* call = getIAXCall(id);
- CHK_VALID_CALL;
+ CHK_VALID_CALL;
- //if (call->getState() == Call::Active) { _debug("Call is already active\n"); return false; }
- _mutexIAX.enterMutex();
- iax_unquelch(call->getSession());
- _mutexIAX.leaveMutex();
- audiolayer->startStream();
- call->setState(Call::Active);
- return true;
+ //if (call->getState() == Call::Active) { _debug("Call is already active\n"); return false; }
+ _mutexIAX.enterMutex();
+ iax_unquelch(call->getSession());
+ _mutexIAX.leaveMutex();
+ audiolayer->startStream();
+ call->setState(Call::Active);
+ return true;
}
- bool
+ bool
IAXVoIPLink::transfer(const CallID& id, const std::string& to)
{
- IAXCall* call = getIAXCall(id);
+ IAXCall* call = getIAXCall(id);
- CHK_VALID_CALL;
+ CHK_VALID_CALL;
- char callto[to.length()+1];
- strcpy(callto, to.c_str());
+ char callto[to.length()+1];
+ strcpy(callto, to.c_str());
- _mutexIAX.enterMutex();
- iax_transfer(call->getSession(), callto);
- _mutexIAX.leaveMutex();
+ _mutexIAX.enterMutex();
+ iax_transfer(call->getSession(), callto);
+ _mutexIAX.leaveMutex();
- return true;
+ return true;
- // should we remove it?
- // removeCall(id);
+ // should we remove it?
+ // removeCall(id);
}
- bool
+ bool
IAXVoIPLink::refuse(const CallID& id)
{
- IAXCall* call = getIAXCall(id);
- std::string reason = "Call rejected manually.";
+ IAXCall* call = getIAXCall(id);
+ std::string reason = "Call rejected manually.";
- CHK_VALID_CALL;
+ CHK_VALID_CALL;
- _mutexIAX.enterMutex();
- iax_reject(call->getSession(), (char*) reason.c_str());
- _mutexIAX.leaveMutex();
- removeCall(id);
+ _mutexIAX.enterMutex();
+ iax_reject(call->getSession(), (char*) reason.c_str());
+ _mutexIAX.leaveMutex();
+ removeCall(id);
- return true;
+ return true;
}
- bool
+ bool
IAXVoIPLink::carryingDTMFdigits(const CallID& id, char code)
{
- IAXCall* call = getIAXCall(id);
+ IAXCall* call = getIAXCall(id);
- CHK_VALID_CALL;
+ CHK_VALID_CALL;
- _mutexIAX.enterMutex();
- iax_send_dtmf(call->getSession(), code);
- _mutexIAX.leaveMutex();
+ _mutexIAX.enterMutex();
+ iax_send_dtmf(call->getSession(), code);
+ _mutexIAX.leaveMutex();
- return true;
+ return true;
}
- bool
+ bool
IAXVoIPLink::iaxOutgoingInvite(IAXCall* call)
{
- struct iax_session *newsession;
- ost::MutexLock m(_mutexIAX);
-
- newsession = iax_session_new();
- if (!newsession) {
- _debug("IAX Error: Can't make new session for a new call\n");
- return false;
- }
- call->setSession(newsession);
- /* reset activity and ping "timers" */
- // iaxc_note_activity(callNo);
-
- std::string strNum = _user + ":" + _pass + "@" + _host + "/" + call->getPeerNumber();
-
- char user[_user.length()+1];
- strcpy(user, _user.c_str());
+ struct iax_session *newsession;
+ ost::MutexLock m(_mutexIAX);
+ std::string username, strNum;
+ char *lang=NULL;
+ int wait, audio_format_preferred, audio_format_capability;
+ IAXAccount *account;
+
+ newsession = iax_session_new();
+ if (!newsession) {
+ _debug("IAX Error: Can't make new session for a new call\n");
+ return false;
+ }
+ call->setSession(newsession);
- char num[strNum.length()+1];
- strcpy(num, strNum.c_str());
+ account = dynamic_cast<IAXAccount*> (getAccountPtr());
+ username = account->getUsername();
+ strNum = username + ":" + account->getPassword() + "@" + account->getHostname() + "/" + call->getPeerNumber();
- char* lang = NULL;
- int wait = 0;
- /** @todo Make preference dynamic, and configurable */
- int audio_format_preferred = call->getFirstMatchingFormat(call->getSupportedFormat());
- int audio_format_capability = call->getSupportedFormat();
+ wait = 0;
+ /** @todo Make preference dynamic, and configurable */
+ audio_format_preferred = call->getFirstMatchingFormat(call->getSupportedFormat());
+ audio_format_capability = call->getSupportedFormat();
- _debug("IAX New call: %s\n", num);
- iax_call(newsession, user, user, num, lang, wait, audio_format_preferred, audio_format_capability);
+ _debug("IAX New call: %s\n", strNum.c_str());
+ iax_call(newsession, username.c_str(), username.c_str(), strNum.c_str(), lang, wait, audio_format_preferred, audio_format_capability);
- return true;
+ return true;
}
- IAXCall*
+ IAXCall*
IAXVoIPLink::iaxFindCallBySession(struct iax_session* session)
{
- // access to callMap shoud use that
- // the code below is like findSIPCallWithCid()
- ost::MutexLock m(_callMapMutex);
- IAXCall* call = NULL;
- CallMap::iterator iter = _callMap.begin();
- while(iter != _callMap.end()) {
- call = dynamic_cast<IAXCall*>(iter->second);
- if (call && call->getSession() == session) {
- return call;
+ // access to callMap shoud use that
+ // the code below is like findSIPCallWithCid()
+ ost::MutexLock m(_callMapMutex);
+ IAXCall* call = NULL;
+ CallMap::iterator iter = _callMap.begin();
+ while(iter != _callMap.end()) {
+ call = dynamic_cast<IAXCall*>(iter->second);
+ if (call && call->getSession() == session) {
+ return call;
+ }
+ iter++;
}
- iter++;
- }
- return NULL; // not found
+ return NULL; // not found
}
- void
+ void
IAXVoIPLink::iaxHandleCallEvent(iax_event* event, IAXCall* call)
{
- // call should not be 0
- // note activity?
- //
- CallID id = call->getCallId();
-
- switch(event->etype) {
- case IAX_EVENT_HANGUP:
- if (Manager::instance().isCurrentCall(id)) {
- // stop audio
- audiolayer->stopStream();
- }
- Manager::instance().peerHungupCall(id);
- removeCall(id);
- break;
-
- case IAX_EVENT_REJECT:
- //Manager::instance().peerHungupCall(id);
- if (Manager::instance().isCurrentCall(id)) {
- // stop audio
- audiolayer->stopStream();
- }
- call->setConnectionState(Call::Connected);
- call->setState(Call::Error);
- Manager::instance().callFailure(id);
- removeCall(id);
- break;
-
- case IAX_EVENT_ACCEPT:
- // Call accepted over there by the computer, not the user yet.
- if (event->ies.format) {
- call->setFormat(event->ies.format);
- }
- break;
-
- case IAX_EVENT_ANSWER:
- if (call->getConnectionState() != Call::Connected){
- call->setConnectionState(Call::Connected);
- call->setState(Call::Active);
-
- if (event->ies.format) {
- // Should not get here, should have been set in EVENT_ACCEPT
- call->setFormat(event->ies.format);
- }
-
- Manager::instance().peerAnsweredCall(id);
- //audiolayer->flushMic();
- audiolayer->startStream();
- // start audio here?
- } else {
- // deja connecté ?
- }
- break;
-
- case IAX_EVENT_BUSY:
- call->setConnectionState(Call::Connected);
- call->setState(Call::Busy);
- Manager::instance().callBusy(id);
- removeCall(id);
- break;
-
- case IAX_EVENT_VOICE:
- //_debug("Should have a decent value!!!!!! = %i\n" , call -> getAudioCodec());
- if( !audiolayer -> isCaptureActive())
- audiolayer->startStream();
- iaxHandleVoiceEvent(event, call);
- break;
-
- case IAX_EVENT_TEXT:
- break;
-
- case IAX_EVENT_RINGA:
- call->setConnectionState(Call::Ringing);
- Manager::instance().peerRingingCall(call->getCallId());
- break;
-
- case IAX_IE_MSGCOUNT:
- break;
- case IAX_EVENT_PONG:
- break;
-
- case IAX_EVENT_URL:
- break;
-
- // case IAX_EVENT_CNG: ??
- // break;
-
- case IAX_EVENT_TIMEOUT:
- break;
-
- case IAX_EVENT_TRANSFER:
- break;
-
- default:
- _debug("Unknown event type (in call event): %d\n", event->etype);
-
- }
+ // call should not be 0
+ // note activity?
+ //
+ CallID id = call->getCallId();
+
+ switch(event->etype) {
+ case IAX_EVENT_HANGUP:
+ if (Manager::instance().isCurrentCall(id)) {
+ // stop audio
+ audiolayer->stopStream();
+ }
+ Manager::instance().peerHungupCall(id);
+ removeCall(id);
+ break;
+
+ case IAX_EVENT_REJECT:
+ //Manager::instance().peerHungupCall(id);
+ if (Manager::instance().isCurrentCall(id)) {
+ // stop audio
+ audiolayer->stopStream();
+ }
+ call->setConnectionState(Call::Connected);
+ call->setState(Call::Error);
+ Manager::instance().callFailure(id);
+ removeCall(id);
+ break;
+
+ case IAX_EVENT_ACCEPT:
+ // Call accepted over there by the computer, not the user yet.
+ if (event->ies.format) {
+ call->setFormat(event->ies.format);
+ }
+ break;
+
+ case IAX_EVENT_ANSWER:
+ if (call->getConnectionState() != Call::Connected){
+ call->setConnectionState(Call::Connected);
+ call->setState(Call::Active);
+
+ if (event->ies.format) {
+ // Should not get here, should have been set in EVENT_ACCEPT
+ call->setFormat(event->ies.format);
+ }
+
+ Manager::instance().peerAnsweredCall(id);
+ //audiolayer->flushMic();
+ audiolayer->startStream();
+ // start audio here?
+ } else {
+ // deja connecté ?
+ }
+ break;
+
+ case IAX_EVENT_BUSY:
+ call->setConnectionState(Call::Connected);
+ call->setState(Call::Busy);
+ Manager::instance().callBusy(id);
+ removeCall(id);
+ break;
+
+ case IAX_EVENT_VOICE:
+ if (!audiolayer->isCaptureActive ())
+ audiolayer->startStream ();
+ iaxHandleVoiceEvent(event, call);
+ break;
+
+ case IAX_EVENT_TEXT:
+ break;
+
+ case IAX_EVENT_RINGA:
+ call->setConnectionState(Call::Ringing);
+ Manager::instance().peerRingingCall(call->getCallId());
+ break;
+
+ case IAX_IE_MSGCOUNT:
+ break;
+ case IAX_EVENT_PONG:
+ break;
+
+ case IAX_EVENT_URL:
+ break;
+
+ // case IAX_EVENT_CNG: ??
+ // break;
+
+ case IAX_EVENT_TIMEOUT:
+ break;
+
+ case IAX_EVENT_TRANSFER:
+ break;
+
+ default:
+ _debug("Unknown event type (in call event): %d\n", event->etype);
+
+ }
}
/* Handle audio event, VOICE packet received */
- void
+ void
IAXVoIPLink::iaxHandleVoiceEvent(iax_event* event, IAXCall* call)
{
-
unsigned char *data;
unsigned int size, max, nbInt16;
int expandedSize, nbSample;
AudioCodec *ac;
- // If we receive datalen == 0, some things of the jitter buffer in libiax2/iax.c
- // were triggered
- if (!event->datalen) {
- // Skip this empty packet.
- //_debug("IAX: Skipping empty jitter-buffer interpolated packet\n");
- return;
- }
-
- if (audiolayer) {
- // On-the-fly codec changing (normally, when we receive a full packet)
- // as per http://tools.ietf.org/id/draft-guy-iax-03.txt
- // - subclass holds the voiceformat property.
- if (event->subclass && event->subclass != call->getFormat()) {
- call->setFormat(event->subclass);
+ // If we receive datalen == 0, some things of the jitter buffer in libiax2/iax.c
+ // were triggered
+ if (!event->datalen) {
+ // Skip this empty packet.
+ //_debug("IAX: Skipping empty jitter-buffer interpolated packet\n");
+ return;
}
- //_debug("Receive: len=%d, format=%d, _receiveDataDecoded=%p\n", event->datalen, call->getFormat(), _receiveDataDecoded);
- ac = call->getCodecMap().getCodec( call -> getAudioCodec() );
- data = (unsigned char*)event->data;
- size = event->datalen;
+ if (audiolayer) {
+ // On-the-fly codec changing (normally, when we receive a full packet)
+ // as per http://tools.ietf.org/id/draft-guy-iax-03.txt
+ // - subclass holds the voiceformat property.
+ if (event->subclass && event->subclass != call->getFormat()) {
+ call->setFormat(event->subclass);
+ }
+ //_debug("Receive: len=%d, format=%d, _receiveDataDecoded=%p\n", event->datalen, call->getFormat(), _receiveDataDecoded);
+ ac = call->getCodecMap().getCodec( call -> getAudioCodec() );
- // Decode data with relevant codec
- max = (int)( ac->getClockRate() * audiolayer->getFrameSize() / 1000 );
+ data = (unsigned char*)event->data;
+ size = event->datalen;
- if (size > max) {
- _debug("The size %d is bigger than expected %d. Packet cropped. Ouch!\n", size, max);
- size = max;
- }
+ // Decode data with relevant codec
+ max = (int)( ac->getClockRate() * audiolayer->getFrameSize() / 1000 );
- expandedSize = ac->codecDecode( spkrDataDecoded , data , size );
- nbInt16 = expandedSize/sizeof(int16);
+ if (size > max) {
+ _debug("The size %d is bigger than expected %d. Packet cropped. Ouch!\n", size, max);
+ size = max;
+ }
- if (nbInt16 > max) {
- _debug("We have decoded an IAX VOICE packet larger than expected: %i VS %i. Cropping.\n", nbInt16, max);
- nbInt16 = max;
- }
+ expandedSize = ac->codecDecode( spkrDataDecoded , data , size );
+ nbInt16 = expandedSize/sizeof(int16);
- nbSample = nbInt16;
- // resample
- nbInt16 = converter->upsampleData( spkrDataDecoded , spkrDataConverted , ac->getClockRate() , audiolayer->getSampleRate() , nbSample);
+ if (nbInt16 > max) {
+ _debug("We have decoded an IAX VOICE packet larger than expected: %i VS %i. Cropping.\n", nbInt16, max);
+ nbInt16 = max;
+ }
- audiolayer->playSamples( spkrDataConverted , nbInt16 * sizeof(SFLDataFormat), true);
+ nbSample = nbInt16;
+ // resample
+ nbInt16 = converter->upsampleData( spkrDataDecoded , spkrDataConverted , ac->getClockRate() , audiolayer->getSampleRate() , nbSample);
- } else {
- _debug("IAX: incoming audio, but no sound card open");
- }
+ //audiolayer->playSamples( spkrDataConverted , nbInt16 * sizeof(SFLDataFormat), true);
+ audiolayer->putMain (spkrDataConverted , nbInt16 * sizeof(SFLDataFormat));
+
+ } else {
+ _debug("IAX: incoming audio, but no sound card open");
+ }
}
/**
* Handle the registration process
*/
- void
+ void
IAXVoIPLink::iaxHandleRegReply(iax_event* event)
{
int new_voicemails;
std::string account_id;
-
+ IAXAccount *account;
+
+ account_id = getAccountID();
+ account = dynamic_cast<IAXAccount *>(Manager::instance().getAccount(account_id));
+
if (event->etype == IAX_EVENT_REGREJ) {
/* Authentication failed! */
_mutexIAX.enterMutex();
iax_destroy(_regSession);
_mutexIAX.leaveMutex();
_regSession = NULL;
- setRegistrationState(ErrorAuth);
+ // Update the account registration state
+ account->setRegistrationState(ErrorAuth);
}
-
+
else if (event->etype == IAX_EVENT_REGACK) {
/* Authentication succeeded */
_mutexIAX.enterMutex();
// Looking for the voicemail information
//if( event->ies != 0 )
- new_voicemails = processIAXMsgCount(event->ies.msgcount);
- _debug("iax voicemail number notification: %i\n", new_voicemails);
+ //new_voicemails = processIAXMsgCount(event->ies.msgcount);
+ //_debug("iax voicemail number notification: %i\n", new_voicemails);
// Notify the client if new voicemail waiting for the current account
- account_id = getAccountID();
- Manager::instance().startVoiceMessageNotification(account_id.c_str(), new_voicemails);
+ //account_id = getAccountID();
+ //Manager::instance().startVoiceMessageNotification(account_id.c_str(), new_voicemails);
iax_destroy(_regSession);
_mutexIAX.leaveMutex();
@@ -783,13 +777,13 @@ IAXVoIPLink::iaxHandleRegReply(iax_event* event)
// I mean, save the timestamp, so that we re-register again in the REFRESH time.
// Defaults to 60, as per draft-guy-iax-03.
_nextRefreshStamp = time(NULL) + (event->ies.refresh ? event->ies.refresh : 60);
- setRegistrationState(Registered);
+ account->setRegistrationState(Registered);
}
}
int IAXVoIPLink::processIAXMsgCount( int msgcount )
{
-
+
// IAX sends the message count under a specific format:
// 1
// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
@@ -803,120 +797,112 @@ int IAXVoIPLink::processIAXMsgCount( int msgcount )
// Thus:
// 0 <= msgcount <= 255 => msgcount new messages, 0 old messages
// msgcount >= 256 => msgcount/256 old messages , msgcount%256 new messages (RULES)
-
+
return msgcount%256;
}
- void
+ void
IAXVoIPLink::iaxHandlePrecallEvent(iax_event* event)
{
- IAXCall* call = NULL;
- CallID id;
- std::string reason = "Error ringing user.";
-
- switch(event->etype) {
- case IAX_EVENT_REGACK:
- case IAX_EVENT_REGREJ:
- _debug("IAX Registration Event in a pre-call setup\n");
- break;
-
- case IAX_EVENT_REGREQ:
- // Received when someone wants to register to us!?!
- // Asterisk receives and answers to that, not us, we're a phone.
- _debug("Registration by a peer, don't allow it\n");
- break;
+ IAXCall* call = NULL;
+ CallID id;
+ std::string reason = "Error ringing user.";
- case IAX_EVENT_CONNECT:
- // We've got an incoming call! Yikes!
- _debug("> IAX_EVENT_CONNECT (receive)\n");
+ switch(event->etype) {
+ case IAX_EVENT_REGACK:
+ case IAX_EVENT_REGREJ:
+ _debug("IAX Registration Event in a pre-call setup\n");
+ break;
- id = Manager::instance().getNewCallID();
+ case IAX_EVENT_REGREQ:
+ // Received when someone wants to register to us!?!
+ // Asterisk receives and answers to that, not us, we're a phone.
+ _debug("Registration by a peer, don't allow it\n");
+ break;
+ case IAX_EVENT_CONNECT:
+ // We've got an incoming call! Yikes!
+ _debug("> IAX_EVENT_CONNECT (receive)\n");
- call = new IAXCall(id, Call::Incoming);
+ id = Manager::instance().getNewCallID();
- if (!call) {
- _debug("! IAX Failure: unable to create an incoming call");
- return;
- }
+ call = new IAXCall(id, Call::Incoming);
- // Setup the new IAXCall
- // Associate the call to the session.
- call->setSession(event->session);
+ if (!call) {
+ _debug("! IAX Failure: unable to create an incoming call");
+ return;
+ }
- // setCallAudioLocal(call);
- call->setCodecMap(Manager::instance().getCodecDescriptorMap());
- call->setConnectionState(Call::Progressing);
+ // Setup the new IAXCall
+ // Associate the call to the session.
+ call->setSession(event->session);
+ // setCallAudioLocal(call);
+ call->setCodecMap(Manager::instance().getCodecDescriptorMap());
+ call->setConnectionState(Call::Progressing);
- if (event->ies.calling_number)
- call->setPeerNumber(std::string(event->ies.calling_number));
- if (event->ies.calling_name)
- call->setPeerName(std::string(event->ies.calling_name));
- if (Manager::instance().incomingCall(call, getAccountID())) {
- /** @todo Faudra considérer éventuellement le champ CODEC PREFS pour
- * l'établissement du codec de transmission */
+ if (event->ies.calling_number)
+ call->setPeerNumber(std::string(event->ies.calling_number));
+ if (event->ies.calling_name)
+ call->setPeerName(std::string(event->ies.calling_name));
- // Remote lists its capabilities
- int format = call->getFirstMatchingFormat(event->ies.capability);
- // Remote asks for preferred codec voiceformat
- int pref_format = call->getFirstMatchingFormat(event->ies.format);
+ if (Manager::instance().incomingCall(call, getAccountID())) {
+ /** @todo Faudra considérer éventuellement le champ CODEC PREFS pour
+ * l'établissement du codec de transmission */
- // Priority to remote's suggestion. In case it's a forwarding, no transcoding
- // will be needed from the server, thus less latency.
- if (pref_format)
- format = pref_format;
+ // Remote lists its capabilities
+ int format = call->getFirstMatchingFormat(event->ies.capability);
+ // Remote asks for preferred codec voiceformat
+ int pref_format = call->getFirstMatchingFormat(event->ies.format);
- iax_accept(event->session, format);
- iax_ring_announce(event->session);
+ // Priority to remote's suggestion. In case it's a forwarding, no transcoding
+ // will be needed from the server, thus less latency.
+ if (pref_format)
+ format = pref_format;
- addCall(call);
- } else {
- // reject call, unable to add it
- iax_reject(event->session, (char*)reason.c_str());
+ iax_accept(event->session, format);
+ iax_ring_announce(event->session);
- delete call; call = NULL;
- }
+ addCall(call);
+ } else {
+ // reject call, unable to add it
+ iax_reject(event->session, (char*)reason.c_str());
- break;
+ delete call; call = NULL;
+ }
- case IAX_EVENT_HANGUP:
- // Remote peer hung up
- call = iaxFindCallBySession(event->session);
- id = call->getCallId();
+ break;
- Manager::instance().peerHungupCall(id);
- removeCall(id);
- break;
+ case IAX_EVENT_HANGUP:
+ // Remote peer hung up
+ call = iaxFindCallBySession(event->session);
+ id = call->getCallId();
- case IAX_EVENT_TIMEOUT: // timeout for an unknown session
+ Manager::instance().peerHungupCall(id);
+ removeCall(id);
+ break;
- break;
+ case IAX_EVENT_TIMEOUT: // timeout for an unknown session
- case IAX_IE_MSGCOUNT:
- //_debug("messssssssssssssssssssssssssssssssssssssssssssssssages\n");
- break;
+ break;
- default:
- _debug("Unknown event type (in precall): %d\n", event->etype);
- }
+ case IAX_IE_MSGCOUNT:
+ //_debug("messssssssssssssssssssssssssssssssssssssssssssssssages\n");
+ break;
-}
+ default:
+ _debug("Unknown event type (in precall): %d\n", event->etype);
+ }
- int
-IAXVoIPLink::iaxCodecMapToFormat(IAXCall* call)
-{
- CodecOrder map = call->getCodecMap().getActiveCodecs();
- printf("taytciatcia = %i\n", map.size());
- return 0;
}
-
void IAXVoIPLink::updateAudiolayer( void )
{
+ _mutexIAX.enterMutex();
audiolayer = NULL;
audiolayer = Manager::instance().getAudioDriver();
+ _mutexIAX.leaveMutex();
}
diff --git a/src/iaxvoiplink.h b/src/iaxvoiplink.h
index fe8978c1b390724ea8a8d2e172a4e15ee359fc25..b82949f640760b474913204c596616018ece1c98 100644
--- a/src/iaxvoiplink.h
+++ b/src/iaxvoiplink.h
@@ -76,7 +76,7 @@ class IAXVoIPLink : public VoIPLink
* Send out registration
* @return bool The new registration state (are we registered ?)
*/
- int sendRegister (void);
+ int sendRegister (AccountID id);
/**
* Destroy registration session
@@ -85,7 +85,7 @@ class IAXVoIPLink : public VoIPLink
* @return bool true if we're registered upstream
* false otherwise
*/
- int sendUnregister (void);
+ int sendUnregister (AccountID id);
/**
* Create a new outgoing call
@@ -166,23 +166,11 @@ class IAXVoIPLink : public VoIPLink
bool isContactPresenceSupported() { return false; }
public: // iaxvoiplink only
- /**
- * @param host Set the host name
- */
- void setHost(const std::string& host) { _host = host; }
-
- /**
- * @param user Set the user name
- */
- void setUser(const std::string& user) { _user = user; }
-
- /**
- * @param pass Set the password
- */
- void setPass(const std::string& pass) { _pass = pass; }
-
+
void updateAudiolayer( void );
+ void setStunServer( const std::string &server ) {};
+
private:
/*
@@ -192,7 +180,7 @@ class IAXVoIPLink : public VoIPLink
* @param msgcount The value sent by IAX in the REGACK message
* @return int The number of new messages waiting for the current registered user
*/
- int processIAXMsgCount( int msgcount );
+ int processIAXMsgCount( int msgcount );
/**
@@ -251,28 +239,12 @@ class IAXVoIPLink : public VoIPLink
*/
bool iaxOutgoingInvite(IAXCall* call);
-
- /**
- * Convert CodecMap to IAX format using IAX constants
- * @return `format` ready to go into iax_* calls
- */
- int iaxCodecMapToFormat(IAXCall* call);
-
/** Threading object */
EventThread* _evThread;
/** registration session : 0 if not register */
struct iax_session* _regSession;
- /** IAX Host */
- std::string _host;
-
- /** IAX User */
- std::string _user;
-
- /** IAX Password */
- std::string _pass;
-
/** Timestamp of when we should refresh the registration up with
* the registrar. Values can be: EPOCH timestamp, 0 if we want no registration, 1
* to force a registration. */
diff --git a/src/managerimpl.cpp b/src/managerimpl.cpp
index c6177f35fc920738250824a767cd0f07635b32f2..e5292532c720e874939a541b2bc6c4ee436bc915 100644
--- a/src/managerimpl.cpp
+++ b/src/managerimpl.cpp
@@ -46,15 +46,8 @@
#include "accountcreator.h" // create new account
#include "sipvoiplink.h"
-#include "useragent.h"
-
#include "user_cfg.h"
-#ifdef USE_ZEROCONF
-#include "zeroconf/DNSService.h"
-#include "zeroconf/DNSServiceTXTRecord.h"
-#endif
-
#define fill_config_str(name, value) \
(_config.addConfigTreeItem(section, Conf::ConfigTreeItem(std::string(name), std::string(value), type_str)))
#define fill_config_int(name, value) \
@@ -88,26 +81,10 @@ ManagerImpl::ManagerImpl (void)
, _callAccountMap()
, _callAccountMapMutex()
, _accountMap()
- , _userAgent(NULL)
- , _userAgentInitlized(false)
- , _sipThreadStop()
-
{
- /* Init private variables
- setup: _path, _exist, _setupLoaded , _dbus
- sound: _audiodriver, _dtmfKey,
- _spkr_volume,_mic_volume = 0; // Initialize after by init() -> initVolume()
- Call: _nbIncomingWaitingCall, _hasTriedToRegister
- SIP Link: _userAgent, _userAgentInitlized
- */
-
-#ifdef USE_ZEROCONF
- _hasZeroconf = true;
- _DNSService = new DNSService();
-#endif
-
- // initialize random generator for call id
- srand (time(NULL));
+
+ // initialize random generator for call id
+ srand (time(NULL));
#ifdef TEST
testAccountMap();
@@ -123,93 +100,68 @@ ManagerImpl::ManagerImpl (void)
// never call if we use only the singleton...
ManagerImpl::~ManagerImpl (void)
{
- terminate();
-
-#ifdef USE_ZEROCONF
- delete _DNSService; _DNSService = 0;
-#endif
-
- _debug("%s stop correctly.\n", PROGNAME);
+ terminate();
+ _debug("%s stop correctly.\n", PROGNAME);
}
void
ManagerImpl::init()
{
- // Load accounts, init map
- loadAccountMap();
+ // Load accounts, init map
+ loadAccountMap();
- //Initialize sip manager
- if(_userAgentInitlized) {
- _userAgent->sipCreate();
- _userAgent->sipInit();
- }
-
- initVolume();
-
- if (_exist == 0) {
- _debug("Cannot create config file in your home directory\n");
- }
-
- initAudioDriver();
- selectAudioDriver();
+ initVolume();
- // Initialize the list of supported audio codecs
- initAudioCodec();
+ if (_exist == 0) {
+ _debug("Cannot create config file in your home directory\n");
+ }
- getAudioInputDeviceList();
+ initAudioDriver();
+ selectAudioDriver();
- AudioLayer *audiolayer = getAudioDriver();
- if (audiolayer!=0) {
- unsigned int sampleRate = audiolayer->getSampleRate();
+ // Initialize the list of supported audio codecs
+ initAudioCodec();
- _debugInit("Load Telephone Tone");
- std::string country = getConfigString(PREFERENCES, ZONE_TONE);
- _telephoneTone = new TelephoneTone(country, sampleRate);
+ AudioLayer *audiolayer = getAudioDriver();
+
+ if (audiolayer!=0) {
+ unsigned int sampleRate = audiolayer->getSampleRate();
- _debugInit("Loading DTMF key");
- _dtmfKey = new DTMF(sampleRate);
- }
+ _debugInit("Load Telephone Tone");
+ std::string country = getConfigString(PREFERENCES, ZONE_TONE);
+ _telephoneTone = new TelephoneTone(country, sampleRate);
- // initRegisterAccounts was here, but we doing it after the gui loaded...
- // the stun detection is long, so it's a better idea to do it after getEvents
- initZeroconf();
-
+ _debugInit("Loading DTMF key");
+ _dtmfKey = new DTMF(sampleRate);
+ }
}
void ManagerImpl::terminate()
{
- saveConfig();
+ saveConfig();
- unloadAccountMap();
+ unloadAccountMap();
- if(_userAgentInitlized) {
- delete _userAgent;
- _userAgent = NULL;
- _userAgentInitlized = false;
- }
-
- _debug("Unload DTMF Key\n");
- delete _dtmfKey;
+ _debug("Unload DTMF Key\n");
+ delete _dtmfKey;
- _debug("Unload Audio Driver\n");
- delete _audiodriver; _audiodriver = NULL;
+ _debug("Unload Audio Driver\n");
+ delete _audiodriver; _audiodriver = NULL;
- _debug("Unload Telephone Tone\n");
- delete _telephoneTone; _telephoneTone = NULL;
+ _debug("Unload Telephone Tone\n");
+ delete _telephoneTone; _telephoneTone = NULL;
- _debug("Unload Audio Codecs\n");
- _codecDescriptorMap.deleteHandlePointer();
+ _debug("Unload Audio Codecs\n");
+ _codecDescriptorMap.deleteHandlePointer();
}
bool
ManagerImpl::isCurrentCall(const CallID& callId) {
- ost::MutexLock m(_currentCallMutex);
return (_currentCallId2 == callId ? true : false);
}
bool
ManagerImpl::hasCurrentCall() {
- ost::MutexLock m(_currentCallMutex);
_debug("Current call ID = %s\n", _currentCallId2.c_str());
if ( _currentCallId2 != "") {
return true;
@@ -219,7 +171,6 @@ ManagerImpl::hasCurrentCall() {
const CallID&
ManagerImpl::getCurrentCallId() {
- ost::MutexLock m(_currentCallMutex);
return _currentCallId2;
}
@@ -237,27 +188,31 @@ ManagerImpl::switchCall(const CallID& id ) {
bool
ManagerImpl::outgoingCall(const std::string& accountid, const CallID& id, const std::string& to)
{
- if (!accountExists(accountid)) {
- _debug("! Manager Error: Outgoing Call: account doesn't exist\n");
- return false;
- }
- if (getAccountFromCall(id) != AccountNULL) {
- _debug("! Manager Error: Outgoing Call: call id already exists\n");
- return false;
- }
- if (hasCurrentCall()) {
- _debug("* Manager Info: there is currently a call, try to hold it\n");
- onHoldCall(getCurrentCallId());
- }
- _debug("- Manager Action: Adding Outgoing Call %s on account %s\n", id.data(), accountid.data());
- if ( getAccountLink(accountid)->newOutgoingCall(id, to) ) {
+ if (!accountExists(accountid)) {
+ _debug("! Manager Error: Outgoing Call: account doesn't exist\n");
+ return false;
+ }
+
+ if (getAccountFromCall(id) != AccountNULL) {
+ _debug("! Manager Error: Outgoing Call: call id already exists\n");
+ return false;
+ }
+
+ if (hasCurrentCall()) {
+ _debug("* Manager Info: there is currently a call, try to hold it\n");
+ onHoldCall(getCurrentCallId());
+ }
+
+ _debug("- Manager Action: Adding Outgoing Call %s on account %s\n", id.data(), accountid.data());
associateCallToAccount( id, accountid );
- switchCall(id);
- return true;
- } else {
- _debug("! Manager Error: An error occur, the call was not created\n");
- }
- return false;
+ if ( getAccountLink(accountid)->newOutgoingCall(id, to) ) {
+ switchCall(id);
+ return true;
+ } else {
+ callFailure(id);
+ _debug("! Manager Error: An error occur, the call was not created\n");
+ }
+ return false;
}
//THREAD=Main : for outgoing Call
@@ -294,26 +249,35 @@ ManagerImpl::answerCall(const CallID& id)
bool
ManagerImpl::hangupCall(const CallID& id)
{
- stopTone(true);
- if (_dbus) _dbus->getCallManager()->callStateChanged(id, "HUNGUP");
- AccountID accountid = getAccountFromCall( id );
- if (accountid == AccountNULL) {
- /** @todo We should tell the GUI that the call doesn't exist, so
- * it clears up. This can happen. */
- _debug("! Manager Hangup Call: Call doesn't exists\n");
- return false;
- }
+ PulseLayer *pulselayer;
+ AccountID accountid;
+ bool returnValue;
- bool returnValue = getAccountLink(accountid)->hangup(id);
- _debug("After voip link hungup!\n");
- removeCallAccount(id);
- switchCall("");
+ stopTone(true);
- if( getConfigInt( PREFERENCES , CONFIG_PA_VOLUME_CTRL ) )
- _audiodriver->restorePulseAppsVolume();
+ /* Broadcast a signal over DBus */
+ if (_dbus) _dbus->getCallManager()->callStateChanged(id, "HUNGUP");
+
+ accountid = getAccountFromCall( id );
+ if (accountid == AccountNULL) {
+ /** @todo We should tell the GUI that the call doesn't exist, so
+ * it clears up. This can happen. */
+ _debug("! Manager Hangup Call: Call doesn't exists\n");
+ return false;
+ }
- _debug("Before hungup return!\n");
- return returnValue;
+ returnValue = getAccountLink(accountid)->hangup(id);
+
+ _debug("After voip link hungup!\n");
+ removeCallAccount(id);
+ switchCall("");
+
+ if( _audiodriver->getLayerType() == PULSEAUDIO && getConfigInt( PREFERENCES , CONFIG_PA_VOLUME_CTRL ) ) {
+ pulselayer = dynamic_cast<PulseLayer *> (getAudioDriver());
+ if(pulselayer) pulselayer->restorePulseAppsVolume();
+ }
+
+ return returnValue;
}
//THREAD=Main
@@ -440,7 +404,6 @@ ManagerImpl::saveConfig (void)
int
ManagerImpl::initRegisterAccounts()
{
-
int status;
bool flag = true;
AccountMap::iterator iter;
@@ -448,17 +411,21 @@ ManagerImpl::initRegisterAccounts()
_debugInit("Initiate VoIP Links Registration");
iter = _accountMap.begin();
+ /* Loop on the account map previously loaded */
while( iter != _accountMap.end() ) {
- if ( iter->second ) {
- iter->second->loadConfig();
- if ( iter->second->isEnabled() ) {
- status = iter->second->registerVoIPLink();
- if (status != SUCCESS)
- flag = false;
+ if ( iter->second ) {
+ iter->second->loadConfig();
+ /* If the account is set as enabled, try to register */
+ if ( iter->second->isEnabled() ) {
+ status = iter->second->registerVoIPLink();
+ if (status != SUCCESS){
+ flag = false;
+ }
+ }
}
- }
- iter++;
+ iter++;
}
+
// calls the client notification here in case of errors at startup...
if( _audiodriver -> getErrorMessage() != -1 )
notifyErrClient( _audiodriver -> getErrorMessage() );
@@ -500,68 +467,63 @@ ManagerImpl::sendDtmf(const CallID& id, char code)
bool
ManagerImpl::playDtmf(char code, bool isTalking)
{
- // HERE are the variable:
- // - boolean variable to play or not (config)
- // - length in milliseconds to play
- // - sample of audiolayer
- stopTone(false);
- int hasToPlayTone = getConfigInt(SIGNALISATION, PLAY_DTMF);
- if (!hasToPlayTone) return false;
-
- // length in milliseconds
- int pulselen = getConfigInt(SIGNALISATION, PULSE_LENGTH);
- if (!pulselen) { return false; }
-
- // numbers of int = length in milliseconds / 1000 (number of seconds)
- // = number of seconds * SAMPLING_RATE by SECONDS
- AudioLayer* audiolayer = getAudioDriver();
- int layer = audiolayer->getLayerType();
-
- // fast return, no sound, so no dtmf
- if (audiolayer==0 || _dtmfKey == 0) { return false; }
- // number of data sampling in one pulselen depends on samplerate
- // size (n sampling) = time_ms * sampling/s
- // ---------------------
- // ms/s
- int size = (int)(pulselen * ((float)audiolayer->getSampleRate()/1000));
-
- // this buffer is for mono
- // TODO <-- this should be global and hide if same size
- SFLDataFormat* _buf = new SFLDataFormat[size];
- bool returnValue = false;
-
- // Handle dtmf
- _dtmfKey->startTone(code);
+ int hasToPlayTone, pulselen, layer, size;
+ bool ret = false;
+ AudioLayer *audiolayer;
+ SFLDataFormat *buf;
+
+ stopTone(false);
+
+ hasToPlayTone = getConfigInt(SIGNALISATION, PLAY_DTMF);
+ if (!hasToPlayTone)
+ return false;
- // copy the sound
- if ( _dtmfKey->generateDTMF(_buf, size) ) {
+ // length in milliseconds
+ pulselen = getConfigInt(SIGNALISATION, PULSE_LENGTH);
+ if (!pulselen)
+ return false;
- // Put buffer to urgentRingBuffer
- // put the size in bytes...
- // so size * 1 channel (mono) * sizeof (bytes for the data)
- if(CHECK_INTERFACE( layer , ALSA ))
- audiolayer->playSamples(_buf, size * sizeof(SFLDataFormat), isTalking);
- else
- _debug("DTMF disabled\n");
- audiolayer->putUrgent( _buf, size * sizeof(SFLDataFormat) );
+ // numbers of int = length in milliseconds / 1000 (number of seconds)
+ // = number of seconds * SAMPLING_RATE by SECONDS
+ audiolayer = getAudioDriver();
+ layer = audiolayer->getLayerType();
+
+ // fast return, no sound, so no dtmf
+ if (audiolayer==0 || _dtmfKey == 0)
+ return false;
+
+ // number of data sampling in one pulselen depends on samplerate
+ // size (n sampling) = time_ms * sampling/s
+ // ---------------------
+ // ms/s
+ size = (int)(pulselen * ((float)audiolayer->getSampleRate()/1000));
+
+ // this buffer is for mono
+ // TODO <-- this should be global and hide if same size
+ buf = new SFLDataFormat[size];
+
+ // Handle dtmf
+ _dtmfKey->startTone(code);
+
+ // copy the sound
+ if ( _dtmfKey->generateDTMF(buf, size) ) {
+ // Put buffer to urgentRingBuffer
+ // put the size in bytes...
+ // so size * 1 channel (mono) * sizeof (bytes for the data)
+ audiolayer->putUrgent (buf, size * sizeof(SFLDataFormat));
+ }
+ ret = true;
- }
- returnValue = true;
+ // TODO Cache the DTMF
- if( CHECK_INTERFACE( layer , PULSEAUDIO ))
- {
- // Cache the samples on the sound server
- // (PulseLayer*)audiolayer->putInCache( code, _buf , size * sizeof(SFLDataFormat) );
- }
-
- delete[] _buf; _buf = 0;
- return returnValue;
+ delete[] buf; buf = 0;
+
+ return ret;
}
// Multi-thread
bool
ManagerImpl::incomingCallWaiting() {
- ost::MutexLock m(_waitingCallMutex);
return (_nbIncomingWaitingCall > 0) ? true : false;
}
@@ -583,7 +545,6 @@ ManagerImpl::removeWaitingCall(const CallID& id) {
bool
ManagerImpl::isWaitingCall(const CallID& id) {
- ost::MutexLock m(_waitingCallMutex);
CallIDSet::iterator iter = _waitingCall.find(id);
if (iter != _waitingCall.end()) {
return false;
@@ -598,38 +559,44 @@ ManagerImpl::isWaitingCall(const CallID& id) {
bool
ManagerImpl::incomingCall(Call* call, const AccountID& accountId)
{
- _debug("Incoming call %s\n", call->getCallId().data());
+ PulseLayer *pulselayer;
+ std::string from, number;
- associateCallToAccount(call->getCallId(), accountId);
+ _debug("Incoming call %s\n", call->getCallId().data());
- if ( !hasCurrentCall() ) {
- call->setConnectionState(Call::Ringing);
- ringtone();
- switchCall(call->getCallId());
- } else {
- addWaitingCall(call->getCallId());
- }
+ associateCallToAccount(call->getCallId(), accountId);
- std::string from = call->getPeerName();
- std::string number = call->getPeerNumber();
+ if ( !hasCurrentCall() ) {
+ call->setConnectionState(Call::Ringing);
+ ringtone();
+ switchCall(call->getCallId());
+ } else {
+ addWaitingCall(call->getCallId());
+ }
- if (from != "" && number != "") {
- from.append(" <");
- from.append(number);
- from.append(">");
- } else if ( from.empty() ) {
- from.append("<");
- from.append(number);
- from.append(">");
- }
+ from = call->getPeerName();
+ number = call->getPeerNumber();
+
+ if (from != "" && number != "") {
+ from.append(" <");
+ from.append(number);
+ from.append(">");
+ } else if ( from.empty() ) {
+ from.append("<");
+ from.append(number);
+ from.append(">");
+ }
- _dbus->getCallManager()->incomingCall(accountId, call->getCallId(), from);
+ /* Broadcast a signal over DBus */
+ _dbus->getCallManager()->incomingCall(accountId, call->getCallId(), from);
- // Reduce volume of the other pulseaudio-connected audio applications
- if( getConfigInt( PREFERENCES , CONFIG_PA_VOLUME_CTRL ) )
- _audiodriver->reducePulseAppsVolume();
+ // Reduce volume of the other pulseaudio-connected audio applications
+ if( _audiodriver->getLayerType() == PULSEAUDIO && getConfigInt( PREFERENCES , CONFIG_PA_VOLUME_CTRL ) ) {
+ pulselayer = dynamic_cast<PulseLayer *> (getAudioDriver());
+ if(pulselayer) pulselayer->reducePulseAppsVolume();
+ }
- return true;
+ return true;
}
//THREAD=VoIP
@@ -664,22 +631,30 @@ ManagerImpl::peerRingingCall(const CallID& id)
void
ManagerImpl::peerHungupCall(const CallID& id)
{
- AccountID accountid = getAccountFromCall( id );
- if (accountid == AccountNULL) {
- _debug("peerHungupCall: Call doesn't exists\n");
- return;
- }
+ PulseLayer *pulselayer;
+ AccountID accountid;
+
+ accountid = getAccountFromCall( id );
+ if (accountid == AccountNULL) {
+ _debug("peerHungupCall: Call doesn't exists\n");
+ return;
+ }
- if (_dbus) _dbus->getCallManager()->callStateChanged(id, "HUNGUP");
- if (isCurrentCall(id)) {
- stopTone(true);
- switchCall("");
- }
- removeWaitingCall(id);
- removeCallAccount(id);
+ /* Broadcast a signal over DBus */
+ if (_dbus) _dbus->getCallManager()->callStateChanged(id, "HUNGUP");
+
+ if (isCurrentCall(id)) {
+ stopTone(true);
+ switchCall("");
+ }
+
+ removeWaitingCall(id);
+ removeCallAccount(id);
- if( getConfigInt( PREFERENCES , CONFIG_PA_VOLUME_CTRL ) )
- _audiodriver->restorePulseAppsVolume();
+ if( _audiodriver->getLayerType() == PULSEAUDIO && getConfigInt( PREFERENCES , CONFIG_PA_VOLUME_CTRL ) ) {
+ pulselayer = dynamic_cast<PulseLayer *> (getAudioDriver());
+ if(pulselayer) pulselayer->restorePulseAppsVolume();
+ }
}
//THREAD=VoIP
@@ -718,7 +693,7 @@ ManagerImpl::startVoiceMessageNotification(const AccountID& accountId, int nb_ms
if (_dbus) _dbus->getCallManager()->voiceMailNotify(accountId, nb_msg) ;
}
-void ManagerImpl::connectionStatusNotification( void )
+void ManagerImpl::connectionStatusNotification( )
{
if (_dbus)
_dbus->getConfigurationManager()->accountsChanged();
@@ -729,30 +704,32 @@ void ManagerImpl::connectionStatusNotification( void )
/**
* Multi Thread
*/
-bool
-ManagerImpl::playATone(Tone::TONEID toneId) {
- int hasToPlayTone = getConfigInt(SIGNALISATION, PLAY_TONES);
- if (!hasToPlayTone) return false;
+bool ManagerImpl::playATone(Tone::TONEID toneId)
+{
+ int hasToPlayTone;
+ AudioLoop *audioloop;
+ AudioLayer *audiolayer;
+ unsigned int nbSamples;
- if (_telephoneTone != 0) {
- _toneMutex.enterMutex();
- _telephoneTone->setCurrentTone(toneId);
- _toneMutex.leaveMutex();
+ hasToPlayTone = getConfigInt(SIGNALISATION, PLAY_TONES);
+ if (!hasToPlayTone)
+ return false;
+
+ audiolayer = getAudioDriver();
- AudioLoop* audioloop = getTelephoneTone();
- unsigned int nbSampling = audioloop->getSize();
- AudioLayer* audiolayer = getAudioDriver();
- SFLDataFormat buf[nbSampling];
- if ( audiolayer ) {
- int layer = audiolayer->getLayerType();
- if(CHECK_INTERFACE( layer , ALSA ) )
- audiolayer->putUrgent( buf, nbSampling );
- else{
- audiolayer->startStream();
- }
- }
- else
- return false;
+ if (_telephoneTone != 0) {
+ _toneMutex.enterMutex();
+ _telephoneTone->setCurrentTone(toneId);
+ _toneMutex.leaveMutex();
+
+ audioloop = getTelephoneTone();
+ nbSamples = audioloop->getSize();
+ SFLDataFormat buf[nbSamples];
+
+ if ( audiolayer ){
+ audiolayer->putUrgent( buf, nbSamples );
+ } else
+ return false;
}
return true;
}
@@ -760,30 +737,30 @@ ManagerImpl::playATone(Tone::TONEID toneId) {
/**
* Multi Thread
*/
-void
-ManagerImpl::stopTone(bool stopAudio=true) {
- int hasToPlayTone = getConfigInt(SIGNALISATION, PLAY_TONES);
- if (!hasToPlayTone) return;
+void ManagerImpl::stopTone (bool stopAudio=true)
+{
+ int hasToPlayTone;
+ AudioLayer *audiolayer;
- if (stopAudio) {
- AudioLayer* audiolayer = getAudioDriver();
- int layer = audiolayer->getLayerType();
- if(CHECK_INTERFACE( layer , ALSA ) ){}
- else{}
- //if (audiolayer) { audiolayer->stopStream(); }
+ hasToPlayTone = getConfigInt(SIGNALISATION, PLAY_TONES);
+ if (!hasToPlayTone)
+ return;
- }
+ if (stopAudio) {
+ audiolayer = getAudioDriver();
+ if (audiolayer) audiolayer->stopStream();
+ }
- _toneMutex.enterMutex();
- if (_telephoneTone != 0) {
- _telephoneTone->setCurrentTone(Tone::TONE_NULL);
- }
- _toneMutex.leaveMutex();
+ _toneMutex.enterMutex();
+ if (_telephoneTone != 0) {
+ _telephoneTone->setCurrentTone(Tone::TONE_NULL);
+ }
+ _toneMutex.leaveMutex();
- // for ringing tone..
- _toneMutex.enterMutex();
- _audiofile.stop();
- _toneMutex.leaveMutex();
+ // for ringing tone..
+ _toneMutex.enterMutex();
+ _audiofile.stop();
+ _toneMutex.leaveMutex();
}
/**
@@ -820,7 +797,6 @@ ManagerImpl::congestion () {
void
ManagerImpl::ringback () {
playATone(Tone::TONE_RINGTONE);
- getAudioDriver()->trigger_thread();
}
/**
@@ -829,38 +805,42 @@ ManagerImpl::ringback () {
void
ManagerImpl::ringtone()
{
+ std::string ringchoice;
+ AudioLayer *audiolayer;
+ AudioCodec *codecForTone;
+ int layer, samplerate;
+ bool loadFile;
+
if( isRingtoneEnabled() )
{
//TODO Comment this because it makes the daemon crashes since the main thread
//synchronizes the ringtone thread.
- std::string ringchoice = getConfigString(AUDIO, RING_CHOICE);
+ ringchoice = getConfigString(AUDIO, RING_CHOICE);
//if there is no / inside the path
if ( ringchoice.find(DIR_SEPARATOR_CH) == std::string::npos ) {
// check inside global share directory
ringchoice = std::string(PROGSHAREDIR) + DIR_SEPARATOR_STR + RINGDIR + DIR_SEPARATOR_STR + ringchoice;
}
- AudioLayer* audiolayer = getAudioDriver();
- int layer = audiolayer->getLayerType();
- if (audiolayer==0) { return; }
- int sampleRate = audiolayer->getSampleRate();
- AudioCodec* codecForTone = _codecDescriptorMap.getFirstCodecAvailable();
+ audiolayer = getAudioDriver();
+ layer = audiolayer->getLayerType();
+ if (audiolayer == 0)
+ return;
+
+ samplerate = audiolayer->getSampleRate();
+ codecForTone = _codecDescriptorMap.getFirstCodecAvailable();
_toneMutex.enterMutex();
- bool loadFile = _audiofile.loadFile(ringchoice, codecForTone , sampleRate);
+ loadFile = _audiofile.loadFile(ringchoice, codecForTone , samplerate);
_toneMutex.leaveMutex();
+
if (loadFile) {
_toneMutex.enterMutex();
_audiofile.start();
_toneMutex.leaveMutex();
if(CHECK_INTERFACE( layer, ALSA )){
- /*int size = _audiofile.getSize();
- SFLDataFormat output[ size ];
- _audiofile.getNext(output, size , 100);
- audiolayer->putUrgent( output , size );
- audiolayer->trigger_thread();*/
- ringback();
+ //ringback();
}
else{
audiolayer->startStream();
@@ -899,26 +879,23 @@ ManagerImpl::getTelephoneFile()
}
}
-void
-ManagerImpl::notificationIncomingCall(void) {
-
- AudioLayer* audiolayer = getAudioDriver();
- if (audiolayer != 0) {
- int layer = audiolayer->getLayerType();
- unsigned int samplerate = audiolayer->getSampleRate();
+void ManagerImpl::notificationIncomingCall(void)
+{
+ AudioLayer *audiolayer;
std::ostringstream frequency;
- frequency << "440/" << FRAME_PER_BUFFER;
-
- Tone tone(frequency.str(), samplerate);
- unsigned int nbSampling = tone.getSize();
- SFLDataFormat buf[nbSampling];
- tone.getNext(buf, tone.getSize());
- if(CHECK_INTERFACE( layer , ALSA ))
- audiolayer->playSamples(buf, sizeof(SFLDataFormat)*nbSampling, true);
- else
- audiolayer->putUrgent( buf, sizeof(SFLDataFormat)*nbSampling );
-
- }
+ unsigned int samplerate, nbSampling;
+
+ audiolayer = getAudioDriver();
+ if (audiolayer != 0) {
+ samplerate = audiolayer->getSampleRate();
+ frequency << "440/" << FRAME_PER_BUFFER;
+ Tone tone(frequency.str(), samplerate);
+ nbSampling = tone.getSize();
+ SFLDataFormat buf[nbSampling];
+ tone.getNext(buf, tone.getSize());
+ /* Put the data in the urgent ring buffer */
+ audiolayer->putUrgent (buf, sizeof(SFLDataFormat)*nbSampling);
+ }
}
/**
@@ -1009,15 +986,17 @@ ManagerImpl::initConfigFile ( bool load_user_value )
std::string type_int("int");
std::string section;
- section = SIGNALISATION;
// Default values, that will be overwritten by the call to
// 'populateFromFile' below.
+ section = SIGNALISATION;
fill_config_int(SYMMETRIC, YES_STR);
fill_config_int(PLAY_DTMF, YES_STR);
fill_config_int(PLAY_TONES, YES_STR);
fill_config_int(PULSE_LENGTH, DFT_PULSE_LENGTH_STR);
fill_config_int(SEND_DTMF_AS, SIP_INFO_STR);
+ fill_config_int(STUN_ENABLE, DFT_STUN_ENABLE);
+ fill_config_int(STUN_SERVER, DFT_STUN_SERVER);
section = AUDIO;
fill_config_int(ALSA_CARD_ID_IN, ALSA_DFT_CARD);
@@ -1266,7 +1245,10 @@ ManagerImpl::setOutputAudioPlugin(const std::string& audioPlugin)
ManagerImpl::getAudioOutputDeviceList(void)
{
_debug("Get audio output device list\n");
- return _audiodriver -> getSoundCardsInfo(SFL_PCM_PLAYBACK);
+ AlsaLayer *layer;
+
+ layer = dynamic_cast<AlsaLayer*> (getAudioDriver ());
+ if (layer) return layer -> getSoundCardsInfo(SFL_PCM_PLAYBACK);
}
/**
@@ -1275,19 +1257,22 @@ ManagerImpl::getAudioOutputDeviceList(void)
void
ManagerImpl::setAudioOutputDevice(const int index)
{
- //int layer = _audiodriver -> getLayerType();
- _debug("Set audio output device: %i\n", index);
- _audiodriver -> setErrorMessage( -1 );
- _audiodriver->openDevice(_audiodriver->getIndexIn(),
- index,
- _audiodriver->getSampleRate(),
- _audiodriver->getFrameSize(),
- SFL_PCM_PLAYBACK,
- _audiodriver->getAudioPlugin());
- if( _audiodriver -> getErrorMessage() != -1)
- notifyErrClient( _audiodriver -> getErrorMessage() );
- // set config
- setConfig( AUDIO , ALSA_CARD_ID_OUT , index );
+ AlsaLayer *alsalayer;
+ std::string alsaplugin;
+ _debug("Set audio output device: %i\n", index);
+
+ _audiodriver -> setErrorMessage( -1 );
+
+ alsalayer = dynamic_cast<AlsaLayer*> (getAudioDriver ());
+ alsaplugin = alsalayer->getAudioPlugin ();
+
+ _audiodriver->openDevice(_audiodriver->getIndexIn(), index, _audiodriver->getSampleRate(), _audiodriver->getFrameSize(), SFL_PCM_PLAYBACK, alsaplugin );
+
+ if( _audiodriver -> getErrorMessage() != -1)
+ notifyErrClient( _audiodriver -> getErrorMessage() );
+
+ // set config
+ setConfig( AUDIO , ALSA_CARD_ID_OUT , index );
}
/**
@@ -1297,7 +1282,10 @@ ManagerImpl::setAudioOutputDevice(const int index)
ManagerImpl::getAudioInputDeviceList(void)
{
_debug("Get audio input device list\n");
- return _audiodriver->getSoundCardsInfo(SFL_PCM_CAPTURE);
+ AlsaLayer *audiolayer;
+
+ audiolayer = dynamic_cast<AlsaLayer *> ( getAudioDriver());
+ if(audiolayer) return audiolayer->getSoundCardsInfo(SFL_PCM_CAPTURE);
}
/**
@@ -1306,19 +1294,23 @@ ManagerImpl::getAudioInputDeviceList(void)
void
ManagerImpl::setAudioInputDevice(const int index)
{
- //int layer = _audiodriver -> getLayerType();
- _debug("Set audio input device %i\n", index);
- _audiodriver -> setErrorMessage( -1 );
- _audiodriver->openDevice(index,
- _audiodriver->getIndexOut(),
- _audiodriver->getSampleRate(),
- _audiodriver->getFrameSize(),
- SFL_PCM_CAPTURE,
- _audiodriver->getAudioPlugin());
- if( _audiodriver -> getErrorMessage() != -1)
- notifyErrClient( _audiodriver -> getErrorMessage() );
- // set config
- setConfig( AUDIO , ALSA_CARD_ID_IN , index );
+ AlsaLayer *alsalayer;
+ std::string alsaplugin;
+
+ _debug("Set audio input device %i\n", index);
+
+ _audiodriver -> setErrorMessage( -1 );
+
+ alsalayer = dynamic_cast<AlsaLayer*> (getAudioDriver ());
+ alsaplugin = alsalayer->getAudioPlugin ();
+
+ _audiodriver->openDevice(index, _audiodriver->getIndexOut(), _audiodriver->getSampleRate(), _audiodriver->getFrameSize(), SFL_PCM_CAPTURE, alsaplugin );
+
+ if( _audiodriver -> getErrorMessage() != -1)
+ notifyErrClient( _audiodriver -> getErrorMessage() );
+
+ // set config
+ setConfig( AUDIO , ALSA_CARD_ID_IN , index );
}
/**
@@ -1399,7 +1391,27 @@ ManagerImpl::setDialpad( void )
( getConfigInt( PREFERENCES , CONFIG_DIALPAD ) == DISPLAY_DIALPAD )? setConfig(PREFERENCES , CONFIG_DIALPAD , NO_STR ) : setConfig( PREFERENCES , CONFIG_DIALPAD , YES_STR );
}
-int
+std::string ManagerImpl::getStunServer( void )
+{
+ return getConfigString(SIGNALISATION , STUN_SERVER);
+}
+
+void ManagerImpl::setStunServer( const std::string &server )
+{
+ setConfig(SIGNALISATION , STUN_SERVER, server );
+}
+
+int ManagerImpl::isStunEnabled (void)
+{
+ return getConfigInt(SIGNALISATION , STUN_ENABLE);
+}
+
+void ManagerImpl::enableStun (void)
+{
+ ( getConfigInt( SIGNALISATION , STUN_ENABLE ) == STUN_ENABLED )? setConfig(SIGNALISATION , STUN_ENABLE , NO_STR ) : setConfig( SIGNALISATION , STUN_ENABLE , YES_STR );
+}
+
+ int
ManagerImpl::getVolumeControls( void )
{
return getConfigInt( PREFERENCES , CONFIG_VOLUME );
@@ -1543,16 +1555,23 @@ ManagerImpl::notifyErrClient( const int32_t& errCode )
int
ManagerImpl::getAudioDeviceIndex(const std::string name)
{
- _debug("Get audio device index\n");
- int num = _audiodriver -> soundCardGetIndex( name );
- return num;
+ AlsaLayer *alsalayer;
+
+ _debug("Get audio device index\n");
+
+ alsalayer = dynamic_cast<AlsaLayer *> (getAudioDriver());
+ if(alsalayer) return alsalayer -> soundCardGetIndex( name );
}
std::string
ManagerImpl::getCurrentAudioOutputPlugin( void )
{
- _debug("Get alsa plugin\n");
- return _audiodriver -> getAudioPlugin();
+ AlsaLayer *alsalayer;
+
+ _debug("Get alsa plugin\n");
+
+ alsalayer = dynamic_cast<AlsaLayer *> (getAudioDriver());
+ if(alsalayer) return alsalayer -> getAudioPlugin ();
}
int ManagerImpl::app_is_running( std::string process )
@@ -1610,104 +1629,88 @@ ManagerImpl::initAudioDriver(void)
void
ManagerImpl::selectAudioDriver (void)
{
- int layer = _audiodriver->getLayerType();
- _debug("Audio layer type: %i\n" , layer);
-
- std::string alsaPlugin = getConfigString( AUDIO , ALSA_PLUGIN );
- int numCardIn = getConfigInt( AUDIO , ALSA_CARD_ID_IN );
- int numCardOut = getConfigInt( AUDIO , ALSA_CARD_ID_OUT );
- int sampleRate = getConfigInt( AUDIO , ALSA_SAMPLE_RATE );
- if (sampleRate <=0 || sampleRate > 48000) {
- sampleRate = 44100;
- }
- int frameSize = getConfigInt( AUDIO , ALSA_FRAME_SIZE );
-
- if( !_audiodriver -> soundCardIndexExist( numCardIn , SFL_PCM_CAPTURE ) )
- {
- _debug(" Card with index %i doesn't exist or cannot capture. Switch to 0.\n", numCardIn);
- numCardIn = ALSA_DFT_CARD_ID ;
- setConfig( AUDIO , ALSA_CARD_ID_IN , ALSA_DFT_CARD_ID );
- }
- if( !_audiodriver -> soundCardIndexExist( numCardOut , SFL_PCM_PLAYBACK ) )
- {
- _debug(" Card with index %i doesn't exist or cannot playback . Switch to 0.\n", numCardOut);
- numCardOut = ALSA_DFT_CARD_ID ;
- setConfig( AUDIO , ALSA_CARD_ID_OUT , ALSA_DFT_CARD_ID );
- }
+ int layer, numCardIn, numCardOut, sampleRate, frameSize;
+ std::string alsaPlugin;
+ AlsaLayer *alsalayer;
+ layer = _audiodriver->getLayerType();
+ _debug("Audio layer type: %i\n" , layer);
- if(CHECK_INTERFACE( layer , ALSA ))
- {
- delete _audiodriver;
- _audiodriver = new AlsaLayer( this );
- _debugInit(" ALSA audio driver \n");
- _audiodriver->setErrorMessage(-1);
- _audiodriver->openDevice( numCardIn , numCardOut, sampleRate, frameSize, SFL_PCM_BOTH, alsaPlugin );
- if( _audiodriver -> getErrorMessage() != -1 )
- notifyErrClient( _audiodriver -> getErrorMessage());
- }else{
- delete _audiodriver;
- _audiodriver = new PulseLayer( this );
- _debug(" Pulse audio driver \n");
- _audiodriver->setErrorMessage(-1);
- _audiodriver->openDevice( numCardIn , numCardOut, sampleRate, frameSize, SFL_PCM_BOTH, alsaPlugin );
- if( _audiodriver -> getErrorMessage() != -1 )
- notifyErrClient( _audiodriver -> getErrorMessage());
- }
+ /* Retrieve the global devices info from the user config */
+ alsaPlugin = getConfigString( AUDIO , ALSA_PLUGIN );
+ numCardIn = getConfigInt( AUDIO , ALSA_CARD_ID_IN );
+ numCardOut = getConfigInt( AUDIO , ALSA_CARD_ID_OUT );
+ sampleRate = getConfigInt( AUDIO , ALSA_SAMPLE_RATE );
+ if (sampleRate <=0 || sampleRate > 48000) {
+ sampleRate = 44100;
+ }
+ frameSize = getConfigInt( AUDIO , ALSA_FRAME_SIZE );
+ /* Only for the ALSA layer, we check the sound card information */
+ if (layer == ALSA)
+ {
+ alsalayer = dynamic_cast<AlsaLayer*> (getAudioDriver ());
+ if( !alsalayer -> soundCardIndexExist( numCardIn , SFL_PCM_CAPTURE ) )
+ {
+ _debug(" Card with index %i doesn't exist or cannot capture. Switch to 0.\n", numCardIn);
+ numCardIn = ALSA_DFT_CARD_ID ;
+ setConfig( AUDIO , ALSA_CARD_ID_IN , ALSA_DFT_CARD_ID );
+ }
+ if( !alsalayer -> soundCardIndexExist( numCardOut , SFL_PCM_PLAYBACK ) )
+ {
+ _debug(" Card with index %i doesn't exist or cannot playback . Switch to 0.\n", numCardOut);
+ numCardOut = ALSA_DFT_CARD_ID ;
+ setConfig( AUDIO , ALSA_CARD_ID_OUT , ALSA_DFT_CARD_ID );
+ }
+ }
+
+ _audiodriver->setErrorMessage(-1);
+ /* Open the audio devices */
+ _audiodriver->openDevice( numCardIn , numCardOut, sampleRate, frameSize, SFL_PCM_BOTH, alsaPlugin );
+ /* Notify the error if there is one */
+ if( _audiodriver -> getErrorMessage() != -1 )
+ notifyErrClient( _audiodriver -> getErrorMessage());
}
-void
-ManagerImpl::switchAudioManager( void )
+void ManagerImpl::switchAudioManager (void)
{
- _debug( "Switching audio manager \n");
+ int type, samplerate, framesize, numCardIn, numCardOut;
+ std::string alsaPlugin;
- int type = _audiodriver->getLayerType();
- int samplerate = getConfigInt( AUDIO , ALSA_SAMPLE_RATE );
- int framesize = getConfigInt( AUDIO , ALSA_FRAME_SIZE );
- std::string alsaPlugin = getConfigString( AUDIO , ALSA_PLUGIN );
- int numCardIn = getConfigInt( AUDIO , ALSA_CARD_ID_IN );
- int numCardOut = getConfigInt( AUDIO , ALSA_CARD_ID_OUT );
+ _debug( "Switching audio manager \n");
- _debug("Deleting current layer... \n" );
- _audiodriver->closeLayer();
- delete _audiodriver; _audiodriver = NULL;
-
- switch( type ){
- case ALSA:
- _debug("Creating Pulseaudio layer...\n");
- _audiodriver = new PulseLayer( this );
- break;
- case PULSEAUDIO:
- _debug("Creating ALSA layer...\n");
- _audiodriver = new AlsaLayer( this );
- break;
- default:
- _debug("Error: audio layer unknown\n");
- }
- _audiodriver->setErrorMessage(-1);
- _audiodriver->openDevice( numCardIn , numCardOut, samplerate, framesize, SFL_PCM_BOTH, alsaPlugin );
- if( _audiodriver -> getErrorMessage() != -1 )
- notifyErrClient( _audiodriver -> getErrorMessage());
-}
+ if(!_audiodriver)
+ return;
-/**
- * Initialize the Zeroconf scanning services loop
- * Informations will be store inside a map DNSService->_services
- * Initialization: Main Thread
- */
- void
-ManagerImpl::initZeroconf(void)
-{
-#ifdef USE_ZEROCONF
- _debugInit("Zeroconf Initialization");
- int useZeroconf = getConfigInt(PREFERENCES, CONFIG_ZEROCONF);
+ type = _audiodriver->getLayerType();
+ samplerate = getConfigInt( AUDIO , ALSA_SAMPLE_RATE );
+ framesize = getConfigInt( AUDIO , ALSA_FRAME_SIZE );
+ alsaPlugin = getConfigString( AUDIO , ALSA_PLUGIN );
+ numCardIn = getConfigInt( AUDIO , ALSA_CARD_ID_IN );
+ numCardOut = getConfigInt( AUDIO , ALSA_CARD_ID_OUT );
- if (useZeroconf) {
- _DNSService->startScanServices();
- }
-#endif
-}
+ _debug("Deleting current layer... \n" );
+ //_audiodriver->closeLayer();
+ delete _audiodriver; _audiodriver = NULL;
+
+ switch( type ){
+ case ALSA:
+ _debug("Creating Pulseaudio layer...\n");
+ _audiodriver = new PulseLayer( this );
+ break;
+ case PULSEAUDIO:
+ _debug("Creating ALSA layer...\n");
+ _audiodriver = new AlsaLayer( this );
+ break;
+ default:
+ _debug("Error: audio layer unknown\n");
+ }
+
+ _audiodriver->setErrorMessage(-1);
+ _audiodriver->openDevice( numCardIn , numCardOut, samplerate, framesize, SFL_PCM_BOTH, alsaPlugin );
+ if( _audiodriver -> getErrorMessage() != -1 )
+ notifyErrClient( _audiodriver -> getErrorMessage());
+}
/**
* Init the volume for speakers/micro from 0 to 100 value
@@ -1724,111 +1727,37 @@ ManagerImpl::initVolume()
void ManagerImpl::setSpkrVolume(unsigned short spkr_vol)
{
- AudioLayer *audiolayer = NULL;
+ PulseLayer *pulselayer = NULL;
+ /* Set the manager sound volume */
_spkr_volume = spkr_vol;
- audiolayer = getAudioDriver();
- if( audiolayer )
+
+ /* Only for PulseAudio */
+ pulselayer = dynamic_cast<PulseLayer*> (getAudioDriver());
+ if (pulselayer)
{
- audiolayer->setPlaybackVolume( spkr_vol );
+ if( pulselayer->getLayerType() == PULSEAUDIO )
+ {
+ if(pulselayer) pulselayer->setPlaybackVolume (spkr_vol);
+ }
}
-
}
void ManagerImpl::setMicVolume(unsigned short mic_vol)
{
- //AudioLayer *audiolayer = NULL;
-
_mic_volume = mic_vol;
- //audiolayer = getAudioDriver();
- //if( audiolayer )
- //{
- // audiolayer->setCaptureVolume( mic_vol );
- //}
}
-/**
- * configuration function requests
- * Main Thread
- */
- bool
-ManagerImpl::getZeroconf(const std::string& sequenceId)
+void ManagerImpl::setSipPort( int port )
{
- bool returnValue = false;
-#ifdef USE_ZEROCONF
- int useZeroconf = getConfigInt(PREFERENCES, CONFIG_ZEROCONF);
- if (useZeroconf && _dbus != NULL) {
- TokenList arg;
- TokenList argTXT;
- std::string newService = "new service";
- std::string newTXT = "new txt record";
- if (!_DNSService->isStart()) { _DNSService->startScanServices(); }
- DNSServiceMap services = _DNSService->getServices();
- DNSServiceMap::iterator iter = services.begin();
- arg.push_back(newService);
- while(iter!=services.end()) {
- arg.push_front(iter->first);
- //_gui->sendMessage("100",sequenceId,arg);
- arg.pop_front(); // remove the first, the name
-
- TXTRecordMap record = iter->second.getTXTRecords();
- TXTRecordMap::iterator iterTXT = record.begin();
- while(iterTXT!=record.end()) {
- argTXT.clear();
- argTXT.push_back(iter->first);
- argTXT.push_back(iterTXT->first);
- argTXT.push_back(iterTXT->second);
- argTXT.push_back(newTXT);
- // _gui->sendMessage("101",sequenceId,argTXT);
- iterTXT++;
- }
- iter++;
- }
- returnValue = true;
- }
-#else
- (void)sequenceId;
-#endif
- return returnValue;
}
-/**
- * Main Thread
- */
- bool
-ManagerImpl::attachZeroconfEvents(const std::string& sequenceId, Pattern::Observer& observer)
+int ManagerImpl::getSipPort( void )
{
- bool returnValue = false;
- // don't need the _gui like getZeroconf function
- // because Observer is here
-#ifdef USE_ZEROCONF
- int useZeroconf = getConfigInt(PREFERENCES, CONFIG_ZEROCONF);
- if (useZeroconf) {
- if (!_DNSService->isStart()) { _DNSService->startScanServices(); }
- _DNSService->attach(observer);
- returnValue = true;
- }
-#else
- (void)sequenceId;
- (void)observer;
-#endif
- return returnValue;
-}
- bool
-ManagerImpl::detachZeroconfEvents(Pattern::Observer& observer)
-{
- bool returnValue = false;
-#ifdef USE_ZEROCONF
- if (_DNSService) {
- _DNSService->detach(observer);
- returnValue = true;
- }
-#else
- (void)observer;
-#endif
- return returnValue;
+ return 5060;
}
+
// TODO: rewrite this
/**
* Main Thread
@@ -1961,7 +1890,7 @@ std::map< std::string, std::string > ManagerImpl::getAccountDetails(const Accoun
std::map<std::string, std::string> a;
std::string accountType;
- enum VoIPLink::RegistrationState state;
+ RegistrationState state;
state = _accountMap[accountID]->getRegistrationState();
accountType = getConfigString(accountID, CONFIG_ACCOUNT_TYPE);
@@ -1970,15 +1899,15 @@ std::map< std::string, std::string > ManagerImpl::getAccountDetails(const Accoun
a.insert( std::pair<std::string, std::string>( CONFIG_ACCOUNT_ENABLE, getConfigString(accountID, CONFIG_ACCOUNT_ENABLE) == "1" ? "TRUE": "FALSE"));
a.insert( std::pair<std::string, std::string>(
"Status",
- (state == VoIPLink::Registered ? "REGISTERED":
- (state == VoIPLink::Unregistered ? "UNREGISTERED":
- (state == VoIPLink::Trying ? "TRYING":
- (state == VoIPLink::ErrorAuth ? "ERROR_AUTH":
- (state == VoIPLink::ErrorNetwork ? "ERROR_NETWORK":
- (state == VoIPLink::ErrorHost ? "ERROR_HOST":
- (state == VoIPLink::ErrorExistStun ? "ERROR_EXIST_STUN":
- (state == VoIPLink::ErrorConfStun ? "ERROR_CONF_STUN":
- (state == VoIPLink::Error ? "ERROR": "ERROR")))))))))
+ (state == Registered ? "REGISTERED":
+ (state == Unregistered ? "UNREGISTERED":
+ (state == Trying ? "TRYING":
+ (state == ErrorAuth ? "ERROR_AUTH":
+ (state == ErrorNetwork ? "ERROR_NETWORK":
+ (state == ErrorHost ? "ERROR_HOST":
+ (state == ErrorExistStun ? "ERROR_EXIST_STUN":
+ (state == ErrorConfStun ? "ERROR_CONF_STUN":
+ (state == Error ? "ERROR": "ERROR")))))))))
)
);
@@ -2003,6 +1932,7 @@ void ManagerImpl::setAccountDetails( const std::string& accountID, const std::ma
std::string accountType;
Account *acc;
+ VoIPLink *link;
accountType = (*details.find(CONFIG_ACCOUNT_TYPE)).second;
@@ -2016,28 +1946,28 @@ void ManagerImpl::setAccountDetails( const std::string& accountID, const std::ma
// SIP SPECIFIC
if (accountType == "SIP") {
+
+ link = Manager::instance().getAccountLink( accountID );
+
+ if( link==0 )
+ {
+ _debug("Can not retrieve SIP link...\n");
+ return;
+ }
+
setConfig(accountID, SIP_STUN_SERVER,(*details.find(SIP_STUN_SERVER)).second);
setConfig(accountID, SIP_USE_STUN, (*details.find(SIP_USE_STUN)).second == "TRUE" ? "1" : "0");
-
- if(!_userAgentInitlized) {
- _userAgentInitlized = true;
- if((*details.find(SIP_USE_STUN)).second == "TRUE")
- _userAgent->setStunServer((*details.find(SIP_STUN_SERVER)).second.data());
- else
- _userAgent->setStunServer(NULL);
-
- _userAgent->sipCreate();
- _userAgent->sipInit();
- } else {
- if((*details.find(SIP_USE_STUN)).second == "TRUE")
- _userAgent->setStunServer((*details.find(SIP_STUN_SERVER)).second.data());
- else
- _userAgent->setStunServer(NULL);
-
- restartPjsip();
- }
- }
+ if((*details.find(SIP_USE_STUN)).second == "TRUE")
+ {
+ link->setStunServer((*details.find(SIP_STUN_SERVER)).second.data());
+ }
+ else
+ {
+ link->setStunServer("");
+ }
+ //restartPjsip();
+ }
saveConfig();
@@ -2053,6 +1983,8 @@ void ManagerImpl::setAccountDetails( const std::string& accountID, const std::ma
// Update account details to the client side
if (_dbus) _dbus->getConfigurationManager()->accountsChanged();
+ //delete link; link=0;
+
}
void
@@ -2079,38 +2011,30 @@ ManagerImpl::sendRegister( const std::string& accountID , const int32_t& expire
ManagerImpl::addAccount(const std::map< std::string, std::string >& details)
{
- /** @todo Deal with both the _accountMap and the Configuration */
- std::string accountType = (*details.find(CONFIG_ACCOUNT_TYPE)).second;
- Account* newAccount;
- std::stringstream accountID;
- accountID << "Account:" << time(NULL);
- AccountID newAccountID = accountID.str();
- /** @todo Verify the uniqueness, in case a program adds accounts, two in a row. */
-
- if (accountType == "SIP") {
- if(!_userAgentInitlized) {
- // Initialize the SIP Manager
- _userAgent = new UserAgent();
- _userAgent->setSipPort(Manager::instance().getConfigInt(PREFERENCES , CONFIG_SIP_PORT));
- }
+ /** @todo Deal with both the _accountMap and the Configuration */
+ std::string accountType = (*details.find(CONFIG_ACCOUNT_TYPE)).second;
+ Account* newAccount;
+ std::stringstream accountID;
+ accountID << "Account:" << time(NULL);
+ AccountID newAccountID = accountID.str();
+ /** @todo Verify the uniqueness, in case a program adds accounts, two in a row. */
- newAccount = AccountCreator::createAccount(AccountCreator::SIP_ACCOUNT, newAccountID);
- }
- else if (accountType == "IAX") {
- newAccount = AccountCreator::createAccount(AccountCreator::IAX_ACCOUNT, newAccountID);
- }
- else {
- _debug("Unknown %s param when calling addAccount(): %s\n", CONFIG_ACCOUNT_TYPE, accountType.c_str());
- return;
- }
- _accountMap[newAccountID] = newAccount;
- setAccountDetails(accountID.str(), details);
-
- saveConfig();
+ if (accountType == "SIP") {
+ newAccount = AccountCreator::createAccount(AccountCreator::SIP_ACCOUNT, newAccountID);
+ }
+ else if (accountType == "IAX") {
+ newAccount = AccountCreator::createAccount(AccountCreator::IAX_ACCOUNT, newAccountID);
+ }
+ else {
+ _debug("Unknown %s param when calling addAccount(): %s\n", CONFIG_ACCOUNT_TYPE, accountType.c_str());
+ return;
+ }
+ _accountMap[newAccountID] = newAccount;
+ setAccountDetails(accountID.str(), details);
- if (_dbus) _dbus->getConfigurationManager()->accountsChanged();
+ saveConfig();
- //restartPjsip();
+ if (_dbus) _dbus->getConfigurationManager()->accountsChanged();
}
void
@@ -2189,20 +2113,6 @@ ManagerImpl::getNewCallID()
return random_id.str();
}
- void
-ManagerImpl::restartPjsip()
-{
- if ( _userAgentInitlized ){
- unregisterCurSIPAccounts();
- _userAgent->sipDestory();
- //_userAgent->setSipPort(Manager::instance().getConfigInt(PREFERENCES , CONFIG_SIP_PORT));
-
- _userAgent->sipCreate();
- _userAgent->sipInit();
- registerCurSIPAccounts();
- }
-}
-
short
ManagerImpl::loadAccountMap()
{
@@ -2211,7 +2121,6 @@ ManagerImpl::loadAccountMap()
TokenList sections = _config.getSections();
std::string accountType;
Account* tmpAccount;
- std::string port;
TokenList::iterator iter = sections.begin();
while(iter != sections.end()) {
@@ -2223,28 +2132,7 @@ ManagerImpl::loadAccountMap()
accountType = getConfigString(*iter, CONFIG_ACCOUNT_TYPE);
if (accountType == "SIP") {
- if(!_userAgentInitlized) {
- // Initialize the SIP Manager
- _userAgent = new UserAgent();
- _userAgentInitlized = true;
- _userAgent->setSipPort(Manager::instance().getConfigInt(PREFERENCES , CONFIG_SIP_PORT));
- }
-
tmpAccount = AccountCreator::createAccount(AccountCreator::SIP_ACCOUNT, *iter);
-
- // Determine whether to use stun for the current account or not
- int useStun = Manager::instance().getConfigInt(tmpAccount->getAccountID(),SIP_USE_STUN);
-
- if(useStun == 1) {
- _userAgent->setStunServer(Manager::instance().getConfigString(tmpAccount->getAccountID(), SIP_STUN_SERVER).data());
- }
-
- /*// Set registration port for all accounts, The last non-5060 port will be recorded in _userAgent.
- port = Manager::instance().getConfigString(tmpAccount->getAccountID(), SIP_PORT);
- std::istringstream is(port);
- is >> iPort;
- if (iPort != DEFAULT_SIP_PORT)
- _userAgent->setRegPort(iPort); */
}
else if (accountType == "IAX") {
tmpAccount = AccountCreator::createAccount(AccountCreator::IAX_ACCOUNT, *iter);
@@ -2259,8 +2147,6 @@ ManagerImpl::loadAccountMap()
nbAccount++;
}
- _debug("\n");
-
iter++;
}
@@ -2328,8 +2214,25 @@ ManagerImpl::getAccountIdFromNameAndServer(const std::string& userName, const st
return AccountNULL;
}
- VoIPLink*
-ManagerImpl::getAccountLink(const AccountID& accountID)
+AccountMap ManagerImpl::getSipAccountMap( void )
+{
+
+ AccountMap::iterator iter;
+ AccountMap sipaccounts;
+ AccountID id;
+ Account *account;
+
+ for(iter = _accountMap.begin(); iter != _accountMap.end(); ++iter) {
+ if( iter->second->getType() == "sip" ){
+ //id = iter->first;
+ //account = iter->second;
+ //sipaccounts.insert( std::pair<id, account> );
+ }
+ }
+ return sipaccounts;
+}
+
+VoIPLink* ManagerImpl::getAccountLink(const AccountID& accountID)
{
Account* acc = getAccount(accountID);
if ( acc ) {
@@ -2348,37 +2251,6 @@ pjsip_regc
return NULL;
}
-
-/**
- * Return the instance of sip manager
- */
-UserAgent *ManagerImpl::getUserAgent()
-{
- return _userAgent;
-}
-
-int
-ManagerImpl::getSipPort()
-{
- if( _userAgent )
- return _userAgent->getSipPort();
- else
- {
- // It means that no SIP accounts are configured, so return a default value
- return 0;
- }
-}
-
-void
-ManagerImpl::setSipPort(int portNum)
-{
- if(portNum != _userAgent->getSipPort()) {
- _userAgent->setSipPort(portNum);
- restartPjsip();
- setConfig( PREFERENCES , CONFIG_SIP_PORT , portNum );
- }
-}
-
void ManagerImpl::unregisterCurSIPAccounts()
{
AccountMap::iterator iter = _accountMap.begin();
diff --git a/src/managerimpl.h b/src/managerimpl.h
index a76987d758b48188f60e853759be06a51d5b75de..cc3411f3ade26470030f558b6cba32e18795bfdd 100644
--- a/src/managerimpl.h
+++ b/src/managerimpl.h
@@ -47,7 +47,6 @@ class CodecDescriptor;
class GuiFramework;
class TelephoneTone;
class VoIPLink;
-class UserAgent;
#ifdef USE_ZEROCONF
class DNSService;
@@ -272,9 +271,6 @@ class ManagerImpl {
*/
void sendRegister( const ::std::string& accountId , const int32_t& expire );
- bool getZeroconf(const std::string& sequenceId);
- bool attachZeroconfEvents(const std::string& sequenceId, Pattern::Observer& observer);
- bool detachZeroconfEvents(Pattern::Observer& observer);
bool getCallStatus(const std::string& sequenceId);
/**
@@ -795,14 +791,15 @@ class ManagerImpl {
*/
void restartPjsip();
- int getSipPort();
-
- void setSipPort(int port);
-
void unregisterCurSIPAccounts();
void registerCurSIPAccounts();
+ /**
+ * Returns a map with only the existing SIP accounts
+ */
+ AccountMap getSipAccountMap( void );
+
private:
/**
@@ -1019,6 +1016,15 @@ public:
AccountID getAccountIdFromNameAndServer(const std::string& userName, const std::string& server);
+ int getSipPort();
+
+ void setSipPort( int port );
+
+ std::string getStunServer (void);
+ void setStunServer (const std::string &server);
+
+ int isStunEnabled (void);
+ void enableStun (void);
private:
// Copy Constructor
@@ -1027,32 +1033,12 @@ private:
// Assignment Operator
ManagerImpl& operator=( const ManagerImpl& rh);
- /**
- * The UserAgent provides sip operation facilities for all sip accounts
- */
- UserAgent *_userAgent;
-
- /** Whether the _UserAgent has been initialized */
- bool _userAgentInitlized;
-
- bool _sipThreadStop;
-
#ifdef TEST
bool testCallAccountMap();
bool testAccountMap();
#endif
friend class ConfigurationTest;
-
-public:
- /**
- * Retuun the instance of sip manager
- */
- UserAgent *getUserAgent();
-
- void setSipThreadStatus(bool status) {_sipThreadStop = status;}
-
- bool getSipThreadStatus() {return _sipThreadStop;}
};
#endif // __MANAGER_H__
diff --git a/src/samplerateconverter.cpp b/src/samplerateconverter.cpp
index 3c6d0ed90db1902bc1e1ee0f29224f1e767453b4..52d047ca4cb5c5da507b1d91b4a7851b49cea9b9 100644
--- a/src/samplerateconverter.cpp
+++ b/src/samplerateconverter.cpp
@@ -78,6 +78,7 @@ void SamplerateConverter::init( void ) {
int SamplerateConverter::upsampleData( SFLDataFormat* dataIn , SFLDataFormat* dataOut, int samplerate1 , int samplerate2 , int nbSamples ){
double upsampleFactor = (double)samplerate2 / samplerate1 ;
+ //_debug("factor = %f\n" , upsampleFactor);
int nbSamplesMax = (int) ( samplerate2 * getFramesize() / 1000 );
if( upsampleFactor != 1 && dataIn != NULL )
{
@@ -106,7 +107,7 @@ int SamplerateConverter::downsampleData( SFLDataFormat* dataIn , SFLDataFormat*
double downsampleFactor = (double)samplerate1 / samplerate2;
//_debug("factor = %f\n" , downsampleFactor);
int nbSamplesMax = (int) ( samplerate1 * getFramesize() / 1000 );
- if ( downsampleFactor != 1)
+ if ( downsampleFactor != 1 )
{
SRC_DATA src_data;
src_data.data_in = _floatBufferUpMic;
diff --git a/src/sipaccount.cpp b/src/sipaccount.cpp
index acce1bb66ed5fff56fad4d6a3d20b77c6ba146a0..61c8bfd05b2a762891c61ebd8fee8f22fd3f2dda 100644
--- a/src/sipaccount.cpp
+++ b/src/sipaccount.cpp
@@ -2,8 +2,6 @@
* Copyright (C) 2006-2009 Savoir-Faire Linux inc.
*
* Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
- * Author: Alexandre Bourget <alexandre.bourget@savoirfairelinux.com>
- * Author: Yan Morin <yan.morin@savoirfairelinux.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
@@ -23,80 +21,71 @@
#include "sipaccount.h"
#include "manager.h"
#include "user_cfg.h"
-#include "useragent.h"
SIPAccount::SIPAccount(const AccountID& accountID)
- : Account(accountID)
- , _userName("")
- , _server("")
+ : Account(accountID, "sip")
, _cred(NULL)
, _contact("")
+ , _bRegister(false)
+ , _regc()
{
- _link = new SIPVoIPLink(accountID);
+ /* SIPVoIPlink is used as a singleton, because we want to have only one link for all the SIP accounts created */
+ /* So instead of creating a new instance, we just fetch the static instance, or create one if it is not yet */
+ /* The SIP library initialization is done in the SIPVoIPLink constructor */
+ /* The SIP voip link is now independant of the account ID as it can manage several SIP accounts */
+ _link = SIPVoIPLink::instance("");
+
+ /* Represents the number of SIP accounts connected the same link */
+ dynamic_cast<SIPVoIPLink*> (_link)->incrementClients();
+
}
SIPAccount::~SIPAccount()
{
- delete _link;
- _link = NULL;
- delete _cred;
- _cred = NULL;
+ /* One SIP account less connected to the sip voiplink */
+ dynamic_cast<SIPVoIPLink*> (_link)->decrementClients();
+ /* Delete accounts-related information */
+ _regc = NULL;
+ delete _cred; _cred = NULL;
}
-int
-SIPAccount::registerVoIPLink()
+int SIPAccount::registerVoIPLink()
{
+ int status;
- int status, useStun;
- SIPVoIPLink *thislink;
-
- _link->setHostname(Manager::instance().getConfigString(_accountID,HOSTNAME));
- useStun = Manager::instance().getConfigInt(_accountID,SIP_USE_STUN);
-
- thislink = dynamic_cast<SIPVoIPLink*> (_link);
- thislink->setStunServer(Manager::instance().getConfigString(_accountID,SIP_STUN_SERVER));
- thislink->setUseStun( useStun!=0 ? true : false);
-
- _link->init();
-
- // Stuff needed for SIP registration.
- thislink->setUsername(Manager::instance().getConfigString(_accountID, USERNAME));
- thislink->setPassword(Manager::instance().getConfigString(_accountID, PASSWORD));
- thislink->setHostname(Manager::instance().getConfigString(_accountID, HOSTNAME));
+ /* Retrieve the account information */
+ /* Stuff needed for SIP registration */
+ setHostname(Manager::instance().getConfigString(_accountID,HOSTNAME));
+ setUsername(Manager::instance().getConfigString(_accountID, USERNAME));
+ setPassword(Manager::instance().getConfigString(_accountID, PASSWORD));
- status = _link->sendRegister();
+ /* Start registration */
+ status = _link->sendRegister( _accountID );
ASSERT( status , SUCCESS );
return SUCCESS;
}
-int
-SIPAccount::unregisterVoIPLink()
+int SIPAccount::unregisterVoIPLink()
{
_debug("SIPAccount: unregister account %s\n" , getAccountID().c_str());
- _link->sendUnregister();
- _link->terminate();
-
- return SUCCESS;
+ return _link->sendUnregister( _accountID );
}
-void
-SIPAccount::loadConfig()
+void SIPAccount::loadConfig()
{
// Account generic
Account::loadConfig();
}
-bool
-SIPAccount::fullMatch(const std::string& userName, const std::string& server)
+bool SIPAccount::fullMatch(const std::string& username, const std::string& hostname)
{
- return (userName == _userName && server == _server);
+ return (username == getUsername() && hostname == getHostname());
}
-bool
-SIPAccount::userMatch(const std::string& userName)
+bool SIPAccount::userMatch(const std::string& username)
{
- return (userName == _userName);
+ return (username == getUsername());
}
diff --git a/src/sipaccount.h b/src/sipaccount.h
index 549d1a6d0d857326720e81b9863df9a4ca045083..a49af2d00864a29c674a2ac152dcd2663d75620c 100644
--- a/src/sipaccount.h
+++ b/src/sipaccount.h
@@ -26,76 +26,86 @@
#include "account.h"
#include "sipvoiplink.h"
-struct pjsip_cred_info;
-
class SIPVoIPLink;
/**
* @file sipaccount.h
* @brief A SIP Account specify SIP specific functions and object (SIPCall/SIPVoIPLink)
-*/
+ */
class SIPAccount : public Account
{
-public:
- /**
- * Constructor
- * @param accountID The account identifier
- */
- SIPAccount(const AccountID& accountID);
-
- /* Copy Constructor */
- SIPAccount(const SIPAccount& rh);
-
- /* Assignment Operator */
- SIPAccount& operator=( const SIPAccount& rh);
-
- /**
- * Virtual destructor
- */
- virtual ~SIPAccount();
-
- /**
- * Actually unuseful, since config loading is done in init()
- */
- void loadConfig();
-
- /**
- * Initialize the SIP voip link with the account parameters and send registration
- */
- int registerVoIPLink();
-
- /**
- * Send unregistration and clean all related stuff ( calls , thread )
- */
- int unregisterVoIPLink();
-
-
- void setUserName(const std::string &name) {_userName = name;}
-
- std::string getUserName() {return _userName;}
-
- void setServer(const std::string &server) {_server = server;}
-
- std::string getServer() {return _server;}
-
- void setCredInfo(pjsip_cred_info *cred) {_cred = cred;}
-
- pjsip_cred_info *getCredInfo() {return _cred;}
-
- void setContact(const std::string contact) {_contact = contact;}
-
- std::string getContact() {return _contact;}
-
- bool fullMatch(const std::string& userName, const std::string& server);
-
- bool userMatch(const std::string& userName);
-
-private:
- std::string _userName;
- std::string _server;
- pjsip_cred_info *_cred;
- std::string _contact;
+ public:
+ /**
+ * Constructor
+ * @param accountID The account identifier
+ */
+ SIPAccount(const AccountID& accountID);
+
+ /* Copy Constructor */
+ SIPAccount(const SIPAccount& rh);
+
+ /* Assignment Operator */
+ SIPAccount& operator=( const SIPAccount& rh);
+
+ /**
+ * Virtual destructor
+ */
+ virtual ~SIPAccount();
+
+ /**
+ * Actually unuseful, since config loading is done in init()
+ */
+ void loadConfig();
+
+ /**
+ * Initialize the SIP voip link with the account parameters and send registration
+ */
+ int registerVoIPLink();
+
+ /**
+ * Send unregistration and clean all related stuff ( calls , thread )
+ */
+ int unregisterVoIPLink();
+
+ inline void setCredInfo(pjsip_cred_info *cred) {_cred = cred;}
+ inline pjsip_cred_info *getCredInfo() {return _cred;}
+
+ inline void setContact(const std::string &contact) {_contact = contact;}
+ inline std::string getContact() {return _contact;}
+
+ bool fullMatch(const std::string& username, const std::string& hostname);
+ bool userMatch(const std::string& username);
+
+ pjsip_regc* getRegistrationInfo( void ) { return _regc; }
+ void setRegistrationInfo( pjsip_regc *regc ) { _regc = regc; }
+
+ /* Registration flag */
+ bool isRegister() {return _bRegister;}
+ void setRegister(bool result) {_bRegister = result;}
+
+
+ private:
+
+ /**
+ * Credential information
+ */
+ pjsip_cred_info *_cred;
+
+ /**
+ * The pjsip client registration information
+ */
+ pjsip_regc *_regc;
+
+ /**
+ * To check if the account is registered
+ */
+ bool _bRegister;
+
+ /*
+ * SIP address
+ */
+ std::string _contact;
};
#endif
diff --git a/src/sipcall.cpp b/src/sipcall.cpp
index 0e53bdd013d066744d80db93204ffbf14a46549f..d43ea938ac230ce691722de456a39a3dcaa7c32a 100644
--- a/src/sipcall.cpp
+++ b/src/sipcall.cpp
@@ -22,8 +22,6 @@
#include "sipcall.h"
#include "global.h" // for _debug
-#include <sstream> // for media buffer
-#include <string>
#define _SENDRECV 0
#define _SENDONLY 1
@@ -51,13 +49,13 @@ SIPCall::SIPCallInvite(pjsip_rx_data *rdata, pj_pool_t *pool)
{
pj_status_t status;
+ // We retrieve the remote sdp offer in the rdata struct to begin the negociation
pjmedia_sdp_session* remote_sdp = getRemoteSDPFromRequest(rdata);
if (remote_sdp == 0) {
return false;
}
// Have to do some stuff here with the SDP
- // We retrieve the remote sdp offer in the rdata struct to begin the negociation
_localSDP = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_session);
_localSDP->conn = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_conn);
@@ -68,7 +66,6 @@ SIPCall::SIPCallInvite(pjsip_rx_data *rdata, pj_pool_t *pool)
_localSDP->time.start = _localSDP->time.stop = 0;
sdpAddMediaDescription(pool);
- _debug("Before validate SDP!\n");
status = pjmedia_sdp_validate( _localSDP );
if (status != PJ_SUCCESS) {
_debug("Can not generate valid local sdp\n");
diff --git a/src/sipcall.h b/src/sipcall.h
index fe23553d5dcdd5edda45b80c6466ea3d6b124725..2ef3d2793451e85bd1fd48151428959cfd7db854 100644
--- a/src/sipcall.h
+++ b/src/sipcall.h
@@ -22,8 +22,8 @@
#define SIPCALL_H
#include "call.h"
+#include "sipvoiplink.h"
#include "audio/codecDescriptor.h"
-#include "useragent.h"
class AudioCodec;
diff --git a/src/sipvoiplink.cpp b/src/sipvoiplink.cpp
index d70e1221216de4257fe15373b637d39446152286..e3fc11b7920e38dd1e813a486f4fdae20ac110f4 100644
--- a/src/sipvoiplink.cpp
+++ b/src/sipvoiplink.cpp
@@ -2,8 +2,7 @@
* Copyright (C) 2004-2009 Savoir-Faire Linux inc.
*
* Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
- * Author: Yan Morin <yan.morin@savoirfairelinux.com>
- * Author : Laurielle Lea <laurielle.lea@savoirfairelinux.com>
+ * Author: Yun Liu <yun.liu@savoirfairelinux.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
@@ -19,261 +18,577 @@
* along with this program; if not, write to the Free Software
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
-/*
- * YM: 2006-11-15: changes unsigned int to std::string::size_type, thanks to Pierre Pomes (AMD64 compilation)
- */
+
#include "sipvoiplink.h"
#include "eventthread.h"
#include "sipcall.h"
-#include <sstream> // for istringstream
#include "sipaccount.h"
-#include "useragent.h"
#include "audio/audiortp.h"
-
-#include "manager.h"
-#include "user_cfg.h" // SIGNALISATION / PULSE #define
-
-#define RANDOM_LOCAL_PORT ((rand() % 27250) + 5250)*2
-
-// 1XX responses
-#define DIALOG_ESTABLISHED 101
-// see: osip_const.h
-
-// need for hold/unhold
-#define INVITE_METHOD "INVITE"
-
-SIPVoIPLink::SIPVoIPLink(const AccountID& accountID)
- : VoIPLink(accountID)
- , _initDone(false)
- , _nbTryListenAddr(2) // number of times to try to start SIP listener
- , _useStun(false)
- , _stunServer("")
- , _localExternAddress("")
- , _localExternPort(0)
- , _audiortp(new AudioRtp())
- , _regc()
- , _bRegister(false)
+
+/**************** EXTERN VARIABLES AND FUNCTIONS (callbacks) **************************/
+
+/*
+ * The global pool factory
+ */
+pj_caching_pool _cp;
+
+/*
+ * The pool to allocate memory
+ */
+pj_pool_t *_pool;
+
+/*
+ * The SIP endpoint
+ */
+pjsip_endpoint *_endpt;
+
+/*
+ * The SIP module
+ */
+pjsip_module _mod_ua;
+
+/*
+ * Thread related
+ */
+pj_thread_t *thread;
+pj_thread_desc desc;
+
+
+/**
+ * Get the number of voicemail waiting in a SIP message
+ */
+void set_voicemail_info( AccountID account, pjsip_msg_body *body );
+
+/**
+ * Set audio (SDP) configuration for a call
+ * localport, localip, localexternalport
+ * @param call a SIPCall valid pointer
+ * @return bool True
+ */
+bool setCallAudioLocal(SIPCall* call, std::string localIP, bool stun, std::string server);
+
+// Documentated from the PJSIP Developer's Guide, available on the pjsip website/
+
+/*
+ * Session callback
+ * Called when the invite session state has changed.
+ *
+ * @param inv A pointer on a pjsip_inv_session structure
+ * @param e A pointer on a pjsip_event structure
+ */
+void call_on_state_changed( pjsip_inv_session *inv, pjsip_event *e);
+
+/*
+ * Session callback
+ * Called after SDP offer/answer session has completed.
+ *
+ * @param inv A pointer on a pjsip_inv_session structure
+ * @param status A pj_status_t structure
+ */
+void call_on_media_update( pjsip_inv_session *inv UNUSED, pj_status_t status UNUSED);
+
+/*
+ * Called when the invote usage module has created a new dialog and invite
+ * because of forked outgoing request.
+ *
+ * @param inv A pointer on a pjsip_inv_session structure
+ * @param e A pointer on a pjsip_event structure
+ */
+void call_on_forked(pjsip_inv_session *inv, pjsip_event *e);
+
+/*
+ * Session callback
+ * Called whenever any transactions within the session has changed their state.
+ * Useful to monitor the progress of an outgoing request.
+ *
+ * @param inv A pointer on a pjsip_inv_session structure
+ * @param tsx A pointer on a pjsip_transaction structure
+ * @param e A pointer on a pjsip_event structure
+ */
+void call_on_tsx_changed(pjsip_inv_session *inv, pjsip_transaction *tsx, pjsip_event *e);
+
+/*
+ * Registration callback
+ */
+void regc_cb(struct pjsip_regc_cbparam *param);
+
+/*
+ * Called to handle incoming requests outside dialogs
+ * @param rdata
+ * @return pj_bool_t
+ */
+pj_bool_t mod_on_rx_request(pjsip_rx_data *rdata);
+
+/*
+ * Called to handle incoming response
+ * @param rdata
+ * @return pj_bool_t
+ */
+pj_bool_t mod_on_rx_response(pjsip_rx_data *rdata UNUSED) ;
+
+/*
+ * Transfer callbacks
+ */
+void xfer_func_cb( pjsip_evsub *sub, pjsip_event *event);
+void xfer_svr_cb(pjsip_evsub *sub, pjsip_event *event);
+void onCallTransfered(pjsip_inv_session *inv, pjsip_rx_data *rdata);
+
+/*************************************************************************************************/
+
+SIPVoIPLink* SIPVoIPLink::_instance = NULL;
+
+
+ SIPVoIPLink::SIPVoIPLink(const AccountID& accountID)
+ : VoIPLink(accountID)
+ , _nbTryListenAddr(2) // number of times to try to start SIP listener
+ , _stunServer("")
+ , _localExternAddress("")
+ , _localExternPort(0)
+ , _audiortp(new AudioRtp())
+ ,_regPort(DEFAULT_SIP_PORT)
+ , _useStun(false)
+ , _clients(0)
{
- // to get random number for RANDOM_PORT
- srand (time(NULL));
+ // to get random number for RANDOM_PORT
+ srand (time(NULL));
+
+ /* Instanciate the C++ thread */
+ _evThread = new EventThread(this);
+
+ /* Start pjsip initialization step */
+ init();
}
SIPVoIPLink::~SIPVoIPLink()
{
- terminate();
+ terminate();
+}
+
+SIPVoIPLink* SIPVoIPLink::instance( const AccountID& id)
+{
+
+ if(!_instance ){
+ _instance = new SIPVoIPLink( id );
+ }
+
+ return _instance;
+}
+
+void SIPVoIPLink::decrementClients (void)
+{
+ _clients--;
+ if(_clients == 0){
+ terminate();
+ SIPVoIPLink::_instance=NULL;
+ }
}
-bool
-SIPVoIPLink::init()
+bool SIPVoIPLink::init()
{
- _regc = NULL;
- _initDone = true;
- return true;
+ if(initDone())
+ return false;
+ /* Initialize the pjsip library */
+ pjsip_init();
+ initDone(true);
+
+ return true;
}
-void
+ void
SIPVoIPLink::terminate()
{
- _initDone = false;
+ delete _evThread; _evThread = NULL;
+
+ /* Clean shutdown of pjsip library */
+ if( initDone() )
+ {
+ pjsip_shutdown();
+ }
+ initDone(false);
}
-void
+ void
SIPVoIPLink::terminateSIPCall()
{
-
- ost::MutexLock m(_callMapMutex);
- CallMap::iterator iter = _callMap.begin();
- SIPCall *call;
- while( iter != _callMap.end() ) {
- call = dynamic_cast<SIPCall*>(iter->second);
- if (call) {
- //TODO terminate the sip call
- delete call; call = 0;
+
+ ost::MutexLock m(_callMapMutex);
+ CallMap::iterator iter = _callMap.begin();
+ SIPCall *call;
+ while( iter != _callMap.end() ) {
+ call = dynamic_cast<SIPCall*>(iter->second);
+ if (call) {
+ // terminate the sip call
+ delete call; call = 0;
+ }
+ iter++;
}
- iter++;
- }
- _callMap.clear();
+ _callMap.clear();
}
-void
+ void
SIPVoIPLink::getEvent()
{
- // Nothing anymore. PJSIP is based on asynchronous events
+ // We have to register the external thread so it could access the pjsip framework
+ if(!pj_thread_is_registered())
+ pj_thread_register( NULL, desc, &thread );
+
+ // PJSIP polling
+ pj_time_val timeout = {0, 10};
+ pjsip_endpt_handle_events( _endpt, &timeout);
}
-int
-SIPVoIPLink::sendRegister()
+int SIPVoIPLink::sendRegister( AccountID id )
{
- AccountID id;
- pj_status_t status;
-
- id = getAccountID();
+ pj_status_t status;
+ int expire_value;
+ char contactTmp[256];
+ pj_str_t svr, aor, contact;
+ pjsip_tx_data *tdata;
+ std::string tmp, hostname, username, password;
+ SIPAccount *account;
+ pjsip_regc *regc;
+
+ account = dynamic_cast<SIPAccount *> (Manager::instance().getAccount(id));
+ hostname = account->getHostname();
+ username = account->getUsername();
+ password = account->getPassword();
+
+ _mutexSIP.enterMutex();
+
+ /* Get the client registration information for this particular account */
+ regc = account->getRegistrationInfo();
+ /* If the registration already exists, delete it */
+ if(regc) {
+ status = pjsip_regc_destroy(regc);
+ regc = NULL;
+ PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
+ }
- if(_regc) {
- status = pjsip_regc_destroy(_regc);
- _regc = NULL;
- PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
- }
+ account->setRegister(true);
- _bRegister = true;
-
- int expire_value = Manager::instance().getRegistrationExpireValue();
- _debug("SIP Registration Expire Value = %i\n" , expire_value);
+ /* Set the expire value of the message from the config file */
+ expire_value = Manager::instance().getRegistrationExpireValue();
- setRegistrationState(Trying);
+ /* Update the state of the voip link */
+ account->setRegistrationState(Trying);
- return Manager::instance().getUserAgent()->addAccount(id, &_regc, getHostname(), getUsername(), getPassword(), expire_value);
-}
+ if (!validStunServer) {
+ account->setRegistrationState(ErrorExistStun);
+ account->setRegister(false);
+ _mutexSIP.leaveMutex();
+ return false;
+ }
-std::string
-SIPVoIPLink::SIPFromHeader(const std::string& userpart, const std::string& hostpart)
-{
- return ("\"" + getFullName() + "\"" + " <sip:" + userpart + "@" + hostpart + ">");
-}
+ /* Create the registration according to the account ID */
+ status = pjsip_regc_create(_endpt, (void*)account, ®c_cb, ®c);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: Unable to create regc.\n");
+ _mutexSIP.leaveMutex();
+ return false;
+ }
-bool
-SIPVoIPLink::sendSIPAuthentification()
-{
- if (getUsername().empty()) {
- /** @todo Ajouter ici un call à setRegistrationState(Error, "Fill balh") ? */
- return false;
- }
- if (getPassword().empty()) {
- /** @todo Même chose ici ? */
- return false;
- }
+ tmp = "sip:" + hostname;
+ pj_strdup2(_pool, &svr, tmp.data());
+
+ tmp = "<sip:" + username + "@" + hostname + ">";
+ pj_strdup2(_pool, &aor, tmp.data());
+
+ sprintf(contactTmp, "<sip:%s@%s:%d>", username.data(), _localExternAddress.data(), _localExternPort);
+ pj_strdup2(_pool, &contact, contactTmp);
+ account->setContact(contactTmp);
- return true;
+ status = pjsip_regc_init(regc, &svr, &aor, &aor, 1, &contact, 600); //timeout);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: Unable to initialize regc. %d\n", status); //, regc->str_srv_url.ptr);
+ _mutexSIP.leaveMutex();
+ return false;
+ }
+
+ pjsip_cred_info *cred = account->getCredInfo();
+
+ if(!cred)
+ cred = new pjsip_cred_info();
+
+ pj_bzero(cred, sizeof (pjsip_cred_info));
+ pj_strdup2(_pool, &cred->username, username.data());
+ cred->data_type = PJSIP_CRED_DATA_PLAIN_PASSWD;
+ pj_strdup2(_pool, &cred->data, password.data());
+ pj_strdup2(_pool, &cred->realm, "*");
+ pj_strdup2(_pool, &cred->scheme, "digest");
+ pjsip_regc_set_credentials(regc, 1, cred);
+
+ account->setCredInfo(cred);
+
+ status = pjsip_regc_register(regc, PJ_TRUE, &tdata);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: Unable to register regc.\n");
+ _mutexSIP.leaveMutex();
+ return false;
+ }
+
+ status = pjsip_regc_send(regc, tdata);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: Unable to send regc request.\n");
+ _mutexSIP.leaveMutex();
+ return false;
+ }
+
+ _mutexSIP.leaveMutex();
+
+ account->setRegistrationInfo(regc);
+
+ return true;
}
-int
-SIPVoIPLink::sendUnregister()
+ int
+SIPVoIPLink::sendUnregister( AccountID id )
{
- _debug("SEND UNREGISTER for account %s\n" , getAccountID().c_str());
-
- if(!_bRegister){
- setRegistrationState(VoIPLink::Unregistered);
- return true;
- }
-
- _bRegister = false;
-
- Manager::instance().getUserAgent()->removeAccount(_regc);
-
- return true;
+ pj_status_t status = 0;
+ pjsip_tx_data *tdata = NULL;
+ SIPAccount *account;
+ pjsip_regc *regc;
+
+ account = dynamic_cast<SIPAccount *> (Manager::instance().getAccount(id));
+ regc = account->getRegistrationInfo();
+
+ if(!account->isRegister()){
+ account->setRegistrationState(Unregistered);
+ return true;
+ }
+
+ if(regc) {
+ status = pjsip_regc_unregister(regc, &tdata);
+ if(status != PJ_SUCCESS) {
+ _debug("UserAgent: Unable to unregister regc.\n");
+ return false;
+ }
+
+ status = pjsip_regc_send( regc, tdata );
+ if(status != PJ_SUCCESS) {
+ _debug("UserAgent: Unable to send regc request.\n");
+ return false;
+ }
+ } else {
+ _debug("UserAgent: regc is null!\n");
+ return false;
+ }
+
+ account->setRegistrationInfo(regc);
+ account->setRegister(false);
+
+ return true;
}
-Call*
+ Call*
SIPVoIPLink::newOutgoingCall(const CallID& id, const std::string& toUrl)
{
- SIPCall* call = new SIPCall(id, Call::Outgoing);
- if (call) {
- //call->setPeerNumber(toUrl);
- call->setPeerNumber(getSipTo(toUrl));
- _debug("Try to make a call to: %s with call ID: %s\n", toUrl.data(), id.data());
- // we have to add the codec before using it in SIPOutgoingInvite...
- call->setCodecMap(Manager::instance().getCodecDescriptorMap());
- if ( SIPOutgoingInvite(call) ) {
- call->setConnectionState(Call::Progressing);
- call->setState(Call::Active);
- addCall(call);
- } else {
- delete call; call = 0;
+ Account* account;
+
+ SIPCall* call = new SIPCall(id, Call::Outgoing);
+
+ if (call) {
+ account = dynamic_cast<SIPAccount *>(Manager::instance().getAccount(Manager::instance().getAccountFromCall(id)));
+ if(!account)
+ {
+ _debug("Error retrieving the account to the make the call with\n");
+ call->setConnectionState(Call::Disconnected);
+ call->setState(Call::Error);
+ delete call; call=0;
+ return call;
+ }
+ //call->setPeerNumber(toUrl);
+ call->setPeerNumber(getSipTo(toUrl, account->getHostname()));
+ _debug("Try to make a call to: %s with call ID: %s\n", toUrl.data(), id.data());
+ // we have to add the codec before using it in SIPOutgoingInvite...
+ call->setCodecMap(Manager::instance().getCodecDescriptorMap());
+ if ( SIPOutgoingInvite(call) ) {
+ call->setConnectionState(Call::Progressing);
+ call->setState(Call::Active);
+ addCall(call);
+ } else {
+ delete call; call = 0;
+ }
}
- }
- return call;
+ return call;
}
-bool
+ bool
SIPVoIPLink::answer(const CallID& id)
{
- _debug("- SIP Action: start answering\n");
- SIPCall* call = getSIPCall(id);
- if (call==0) {
- _debug("! SIP Failure: SIPCall doesn't exists\n");
- return false;
- }
-
- int i = Manager::instance().getUserAgent()->answer(call);
-
- if (i != 0) {
- _debug("< SIP Building Error: send 400 Bad Request\n");
- } else {
- // use exosip, bug locked
- i = 0;
- _debug("* SIP Info: Starting AudioRTP when answering\n");
- if (_audiortp->createNewSession(call) >= 0) {
- call->setAudioStart(true);
- call->setConnectionState(Call::Connected);
- call->setState(Call::Active);
- return true;
- } else {
- _debug("! SIP Failure: Unable to start sound when answering %s/%d\n", __FILE__, __LINE__);
+ int i;
+ SIPCall *call;
+ pj_status_t status;
+ pjsip_tx_data *tdata;
+
+ _debug("- SIP Action: start answering\n");
+
+ call = getSIPCall(id);
+
+ if (call==0) {
+ _debug("! SIP Failure: SIPCall doesn't exists\n");
+ return false;
}
- }
- removeCall(call->getCallId());
- return false;
+
+ // User answered the incoming call, tell peer this news
+ if (call->startNegociation(_pool)) {
+ // Create and send a 200(OK) response
+ _debug("UserAgent: Negociation success!\n");
+ status = pjsip_inv_answer(call->getInvSession(), PJSIP_SC_OK, NULL, NULL, &tdata);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+ status = pjsip_inv_send_msg(call->getInvSession(), tdata);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+
+ _debug("* SIP Info: Starting AudioRTP when answering\n");
+ if (_audiortp->createNewSession(call) >= 0) {
+ call->setAudioStart(true);
+ call->setConnectionState(Call::Connected);
+ call->setState(Call::Active);
+ return true;
+ } else {
+ _debug("! SIP Failure: Unable to start sound when answering %s/%d\n", __FILE__, __LINE__);
+ }
+ }
+ removeCall(call->getCallId());
+ return false;
}
-bool
+ bool
SIPVoIPLink::hangup(const CallID& id)
{
- SIPCall* call = getSIPCall(id);
- if (call==0) { _debug("! SIP Error: Call doesn't exist\n"); return false; }
-
- Manager::instance().getUserAgent()->hangup(call);
-
- // Release RTP thread
- if (Manager::instance().isCurrentCall(id)) {
- _debug("* SIP Info: Stopping AudioRTP for hangup\n");
- _audiortp->closeRtpSession();
- }
- removeCall(id);
- return true;
+ pj_status_t status;
+ pjsip_tx_data *tdata = NULL;
+ SIPCall* call;
+
+ call = getSIPCall(id);
+
+ if (call==0) { _debug("! SIP Error: Call doesn't exist\n"); return false; }
+
+ // User hangup current call. Notify peer
+ status = pjsip_inv_end_session(call->getInvSession(), 404, NULL, &tdata);
+ if(status != PJ_SUCCESS)
+ return false;
+
+ if(tdata == NULL)
+ return true;
+
+ status = pjsip_inv_send_msg(call->getInvSession(), tdata);
+ if(status != PJ_SUCCESS)
+ return false;
+
+ call->getInvSession()->mod_data[getModId()] = NULL;
+ return true;
+
+ // Release RTP thread
+ if (Manager::instance().isCurrentCall(id)) {
+ _debug("* SIP Info: Stopping AudioRTP for hangup\n");
+ _audiortp->closeRtpSession();
+ }
+
+ removeCall(id);
+
+ return true;
}
-bool
+ bool
SIPVoIPLink::cancel(const CallID& id)
{
- SIPCall* call = getSIPCall(id);
- if (call==0) { _debug("! SIP Error: Call doesn't exist\n"); return false; }
+ SIPCall* call = getSIPCall(id);
+ if (call==0) { _debug("! SIP Error: Call doesn't exist\n"); return false; }
- _debug("- SIP Action: Cancel call %s [cid: %3d]\n", id.data(), call->getCid());
+ _debug("- SIP Action: Cancel call %s [cid: %3d]\n", id.data(), call->getCid());
- removeCall(id);
+ removeCall(id);
- return true;
+ return true;
}
-bool
+ bool
SIPVoIPLink::onhold(const CallID& id)
{
- SIPCall* call = getSIPCall(id);
- if (call==0) { _debug("! SIP Error: call doesn't exist\n"); return false; }
+ pj_status_t status;
+ pjsip_tx_data *tdata;
+ pjmedia_sdp_attr *attr;
+ pjmedia_sdp_session* local_sdp;
+ SIPCall* call;
+
+ call = getSIPCall(id);
+
+ if (call==0) { _debug("! SIP Error: call doesn't exist\n"); return false; }
+
+
+ // Stop sound
+ call->setAudioStart(false);
+ call->setState(Call::Hold);
+ _debug("* SIP Info: Stopping AudioRTP for onhold action\n");
+ //_mutexSIP.enterMutex();
+ _audiortp->closeRtpSession();
+ //_mutexSIP.leaveMutex();
+
+ local_sdp = call->getLocalSDPSession();
+
+ if( local_sdp == NULL ){
+ _debug("! SIP Failure: unable to find local_sdp\n");
+ return false;
+ }
- // Stop sound
- call->setAudioStart(false);
- call->setState(Call::Hold);
- _debug("* SIP Info: Stopping AudioRTP for onhold action\n");
- _audiortp->closeRtpSession();
+ /* Create re-INVITE with new offer */
+ // Remove all the attributes with the specified name
+ pjmedia_sdp_media_remove_all_attr(local_sdp->media[0], "sendrecv");
+ attr = pjmedia_sdp_attr_create(_pool, "sendonly", NULL);
+ pjmedia_sdp_media_add_attr(local_sdp->media[0], attr);
- return Manager::instance().getUserAgent()->onhold(call);
+ status = pjsip_inv_reinvite( call->getInvSession(), NULL, local_sdp, &tdata);
+ if( status != PJ_SUCCESS )
+ {
+ _debug("On hold: creation of the Re-invite request failed\n");
+ return false;
+ }
+ /* Send the request */
+ status = pjsip_inv_send_msg( call->getInvSession(), tdata);
+
+ return (status == PJ_SUCCESS);
}
-bool
+ bool
SIPVoIPLink::offhold(const CallID& id)
{
SIPCall *call;
+ pj_status_t status;
+ pjsip_tx_data *tdata;
+ pjmedia_sdp_attr *attr;
+ pjmedia_sdp_session* local_sdp;
call = getSIPCall(id);
+
if (call==0) {
_debug("! SIP Error: Call doesn't exist\n");
return false;
}
- if(!Manager::instance().getUserAgent()->offhold(call))
+ local_sdp = call->getLocalSDPSession();
+ if( local_sdp == NULL ){
+ _debug("! SIP Failure: unable to find local_sdp\n");
+ return false;
+ }
+
+ /* Create re-INVITE with new offer */
+ // Remove all the attributes with the specified name
+ pjmedia_sdp_media_remove_all_attr(local_sdp->media[0], "sendonly");
+ attr = pjmedia_sdp_attr_create(_pool, "sendrecv", NULL);
+ pjmedia_sdp_media_add_attr(local_sdp->media[0], attr);
+
+ status = pjsip_inv_reinvite( call->getInvSession(), NULL, local_sdp , &tdata);
+ if( status != PJ_SUCCESS )
+ {
+ _debug("Off hold: creation of the Re-invite request failed\n");
+ return false;
+ }
+
+ /* Send the request */
+ status = pjsip_inv_send_msg( call->getInvSession(), tdata);
+ if( status != PJ_SUCCESS )
return false;
// Enable audio
@@ -284,17 +599,28 @@ SIPVoIPLink::offhold(const CallID& id)
_debug("! SIP Failure: Unable to start sound (%s:%d)\n", __FILE__, __LINE__);
return false;
}
-
+
return true;
}
-bool
+ bool
SIPVoIPLink::transfer(const CallID& id, const std::string& to)
{
SIPCall *call;
std::string tmp_to;
+ pjsip_evsub *sub;
+ pjsip_tx_data *tdata;
+ struct pjsip_evsub_user xfer_cb;
+ pj_status_t status;
+ pj_str_t dest;
+ AccountID account_id;
+ Account* account;
+
call = getSIPCall(id);
+ account_id = Manager::instance().getAccountFromCall(id);
+ account = dynamic_cast<SIPAccount *>(Manager::instance().getAccount(account_id));
+
if (call==0) {
_debug("! SIP Failure: Call doesn't exist\n");
return false;
@@ -302,16 +628,48 @@ SIPVoIPLink::transfer(const CallID& id, const std::string& to)
tmp_to = SIPToHeader(to);
if (tmp_to.find("@") == std::string::npos) {
- tmp_to = tmp_to + "@" + getHostname();
+ tmp_to = tmp_to + "@" + account->getHostname();
}
_debug("In transfer, tmp_to is %s\n", tmp_to.data());
- return Manager::instance().getUserAgent()->transfer(call, tmp_to);
+ pj_strdup2(_pool, &dest, to.data());
+
+ /* Create xfer client subscription. */
+ pj_bzero(&xfer_cb, sizeof(xfer_cb));
+ xfer_cb.on_evsub_state = &xfer_func_cb;
+
+ status = pjsip_xfer_create_uac(call->getInvSession()->dlg, &xfer_cb, &sub);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: Unable to create xfer -- %d\n", status);
+ return false;
+ }
+
+ /* Associate this voiplink of call with the client subscription
+ * We can not just associate call with the client subscription
+ * because after this function, we can not find the cooresponding
+ * voiplink from the call any more. But the voiplink is useful!
+ */
+ AccountID accId = Manager::instance().getAccountFromCall(call->getCallId());
+ pjsip_evsub_set_mod_data(sub, getModId(), this);
+
+ /*
+ * Create REFER request.
+ */
+ status = pjsip_xfer_initiate(sub, &dest, &tdata);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: Unable to create REFER request -- %d\n", status);
+ return false;
+ }
+
+ /* Send. */
+ status = pjsip_xfer_send_request(sub, tdata);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: Unable to send REFER request -- %d\n", status);
+ return false;
+ }
- //_audiortp->closeRtpSession();
- // shall we delete the call?
- //removeCall(id);
+ return true;
}
bool SIPVoIPLink::transferStep2()
@@ -320,10 +678,13 @@ bool SIPVoIPLink::transferStep2()
return true;
}
-bool
+ bool
SIPVoIPLink::refuse (const CallID& id)
{
SIPCall *call;
+ pj_status_t status;
+ pjsip_tx_data *tdata;
+
call = getSIPCall(id);
@@ -338,10 +699,20 @@ SIPVoIPLink::refuse (const CallID& id)
return false;
}
- return Manager::instance().getUserAgent()->refuse(call);
+ // User refuse current call. Notify peer
+ status = pjsip_inv_end_session(call->getInvSession(), PJSIP_SC_DECLINE, NULL, &tdata); //603
+ if(status != PJ_SUCCESS)
+ return false;
+
+ status = pjsip_inv_send_msg(call->getInvSession(), tdata);
+ if(status != PJ_SUCCESS)
+ return false;
+
+ call->getInvSession()->mod_data[getModId()] = NULL;
+ return true;
}
-bool
+ bool
SIPVoIPLink::carryingDTMFdigits(const CallID& id, char code)
{
@@ -349,8 +720,14 @@ SIPVoIPLink::carryingDTMFdigits(const CallID& id, char code)
int duration;
const int body_len = 1000;
char *dtmf_body;
+ pj_status_t status;
+ pjsip_tx_data *tdata;
+ pj_str_t methodName, content;
+ pjsip_method method;
+ pjsip_media_type ctype;
call = getSIPCall(id);
+
if (call==0) {
_debug("Call doesn't exist\n");
return false;
@@ -358,210 +735,1431 @@ SIPVoIPLink::carryingDTMFdigits(const CallID& id, char code)
duration = Manager::instance().getConfigInt(SIGNALISATION, PULSE_LENGTH);
dtmf_body = new char[body_len];
-
+
snprintf(dtmf_body, body_len - 1, "Signal=%c\r\nDuration=%d\r\n", code, duration);
-
- return Manager::instance().getUserAgent()->carryingDTMFdigits(call, dtmf_body);
+
+ pj_strdup2(_pool, &methodName, "INFO");
+ pjsip_method_init_np(&method, &methodName);
+
+ /* Create request message. */
+ status = pjsip_dlg_create_request( call->getInvSession()->dlg, &method, -1, &tdata);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: Unable to create INFO request -- %d\n", status);
+ return false;
+ }
+
+ /* Get MIME type */
+ pj_strdup2(_pool, &ctype.type, "application");
+ pj_strdup2(_pool, &ctype.subtype, "dtmf-relay");
+
+ /* Create "application/dtmf-relay" message body. */
+ pj_strdup2(_pool, &content, dtmf_body);
+ tdata->msg->body = pjsip_msg_body_create( tdata->pool, &ctype.type, &ctype.subtype, &content);
+ if (tdata->msg->body == NULL) {
+ _debug("UserAgent: Unable to create msg body!\n");
+ pjsip_tx_data_dec_ref(tdata);
+ return false;
+ }
+
+ /* Send the request. */
+ status = pjsip_dlg_send_request( call->getInvSession()->dlg, tdata, getModId(), NULL);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: Unable to send MESSAGE request -- %d\n", status);
+ return false;
+ }
+
+ return true;
}
-bool
+ bool
SIPVoIPLink::SIPOutgoingInvite(SIPCall* call)
{
- // If no SIP proxy setting for direct call with only IP address
- if (!SIPStartCall(call, "")) {
- _debug("! SIP Failure: call not started\n");
- return false;
- }
- return true;
+ // If no SIP proxy setting for direct call with only IP address
+ if (!SIPStartCall(call, "")) {
+ _debug("! SIP Failure: call not started\n");
+ return false;
+ }
+ return true;
}
-bool
+ bool
SIPVoIPLink::SIPStartCall(SIPCall* call, const std::string& subject UNUSED)
{
- std::string to;
+ std::string strTo, strFrom;
+ pj_status_t status;
+ pjsip_dialog *dialog;
+ pjsip_tx_data *tdata;
+ pj_str_t from, to, contact;
+ AccountID id;
+ SIPAccount *account;
if (!call)
return false;
- to = getSipTo(call->getPeerNumber());
- _debug(" To: %s\n", to.data());
+ id = Manager::instance().getAccountFromCall(call->getCallId());
+ // Get the basic information about the callee account
+ account = dynamic_cast<SIPAccount *>(Manager::instance().getAccount(id));
- return Manager::instance().getUserAgent()->makeOutgoingCall(to, call, getAccountID());
-}
+ strTo = getSipTo(call->getPeerNumber(), account->getHostname());
+ _debug(" To: %s\n", strTo.data());
+
+ // Generate the from URI
+ strFrom = "sip:" + account->getUsername() + "@" + account->getHostname();
+
+ // pjsip need the from and to information in pj_str_t format
+ pj_strdup2(_pool, &from, strFrom.data());
+ pj_strdup2(_pool, &to, strTo.data());
+ pj_strdup2(_pool, &contact, account->getContact().data());
+
+ // create the dialog (UAC)
+ status = pjsip_dlg_create_uac(pjsip_ua_instance(), &from,
+ &contact,
+ &to,
+ NULL,
+ &dialog);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, false);
+
+ setCallAudioLocal(call, getLocalIPAddress(), useStun(), getStunServer());
+ call->setIp(getLocalIP());
+
+ // Building the local SDP offer
+ call->createInitialOffer(_pool);
+
+ // Create the invite session for this call
+ pjsip_inv_session *inv;
+ status = pjsip_inv_create_uac(dialog, call->getLocalSDPSession(), 0, &inv);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, false);
-std::string
-SIPVoIPLink::getSipFrom() {
+ // Set auth information
+ pjsip_auth_clt_set_credentials(&dialog->auth_sess, 1, account->getCredInfo());
- // Form the From header field basis on configuration panel
- std::string hostname;
-
- hostname = getHostname();
+ // Associate current call in the invite session
+ inv->mod_data[getModId()] = call;
- if ( hostname.empty() ) {
- hostname = _localIPAddress;
- }
- return SIPFromHeader(getUsername(), hostname);
+ status = pjsip_inv_invite(inv, &tdata);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, false);
+
+ // Associate current invite session in the call
+ call->setInvSession(inv);
+
+ status = pjsip_inv_send_msg(inv, tdata);
+ if(status != PJ_SUCCESS) {
+ return false;
+ }
+
+ return true;
}
-std::string
-SIPVoIPLink::getSipTo(const std::string& to_url) {
- // Form the From header field basis on configuration panel
- //bool isRegistered = (_eXosipRegID == EXOSIP_ERROR_STD) ? false : true;
+std::string SIPVoIPLink::getSipTo(const std::string& to_url, std::string hostname) {
+ // Form the From header field basis on configuration panel
+ //bool isRegistered = (_eXosipRegID == EXOSIP_ERROR_STD) ? false : true;
- // add a @host if we are registered and there is no one inside the url
+ // add a @host if we are registered and there is no one inside the url
if (to_url.find("@") == std::string::npos) {// && isRegistered) {
- std::string host = getHostname();
- if(!host.empty()) {
- return SIPToHeader(to_url + "@" + host);
+ if(!hostname.empty()) {
+ return SIPToHeader(to_url + "@" + hostname);
+ }
+ }
+ return SIPToHeader(to_url);
}
- }
- return SIPToHeader(to_url);
-}
-std::string
-SIPVoIPLink::SIPToHeader(const std::string& to)
-{
- if (to.find("sip:") == std::string::npos) {
- return ("sip:" + to );
- } else {
- return to;
- }
-}
+ std::string SIPVoIPLink::SIPToHeader(const std::string& to)
+ {
+ if (to.find("sip:") == std::string::npos) {
+ return ("sip:" + to );
+ } else {
+ return to;
+ }
+ }
-bool
-SIPVoIPLink::SIPCheckUrl(const std::string& url UNUSED)
-{
- return true;
-}
+ bool
+ SIPVoIPLink::SIPCheckUrl(const std::string& url UNUSED)
+ {
+ return true;
+ }
+
+ bool setCallAudioLocal(SIPCall* call, std::string localIP, bool stun, std::string server)
+ {
+ // Setting Audio
+ unsigned int callLocalAudioPort = RANDOM_LOCAL_PORT;
+ unsigned int callLocalExternAudioPort = callLocalAudioPort;
+ if (stun) {
+ // If use Stun server
+ if (Manager::instance().behindNat(server, callLocalAudioPort)) {
+ callLocalExternAudioPort = Manager::instance().getFirewallPort();
+ }
+ }
+ _debug(" Setting local audio port to: %d\n", callLocalAudioPort);
+ _debug(" Setting local audio port (external) to: %d\n", callLocalExternAudioPort);
+
+ // Set local audio port for SIPCall(id)
+ call->setLocalIp(localIP);
+ call->setLocalAudioPort(callLocalAudioPort);
+ call->setLocalExternAudioPort(callLocalExternAudioPort);
+
+ return true;
+ }
-bool
-SIPVoIPLink::setCallAudioLocal(SIPCall* call)
-{
- // Setting Audio
- unsigned int callLocalAudioPort = RANDOM_LOCAL_PORT;
- unsigned int callLocalExternAudioPort = callLocalAudioPort;
- if (_useStun) {
- // If use Stun server
- if (Manager::instance().behindNat(_stunServer, callLocalAudioPort)) {
- callLocalExternAudioPort = Manager::instance().getFirewallPort();
- }
- }
- _debug(" Setting local audio port to: %d\n", callLocalAudioPort);
- _debug(" Setting local audio port (external) to: %d\n", callLocalExternAudioPort);
-
- // Set local audio port for SIPCall(id)
- call->setLocalIp(_localIPAddress);
- call->setLocalAudioPort(callLocalAudioPort);
- call->setLocalExternAudioPort(callLocalExternAudioPort);
-
- return true;
-}
+ void
+ SIPVoIPLink::SIPCallServerFailure(SIPCall *call)
+ {
+ //if (!event->response) { return; }
+ //switch(event->response->status_code) {
+ //case SIP_SERVICE_UNAVAILABLE: // 500
+ //case SIP_BUSY_EVRYWHERE: // 600
+ //case SIP_DECLINE: // 603
+ //SIPCall* call = findSIPCallWithCid(event->cid);
+ if (call != 0) {
+ _debug("Server error!\n");
+ CallID id = call->getCallId();
+ Manager::instance().callFailure(id);
+ removeCall(id);
+ }
+ //break;
+ //}
+ }
+
+ void
+ SIPVoIPLink::SIPCallClosed(SIPCall *call)
+ {
+ // it was without did before
+ //SIPCall* call = findSIPCallWithCid(event->cid);
+ if (!call) { return; }
+
+ CallID id = call->getCallId();
+ //call->setDid(event->did);
+ if (Manager::instance().isCurrentCall(id)) {
+ call->setAudioStart(false);
+ _debug("* SIP Info: Stopping AudioRTP when closing\n");
+ _audiortp->closeRtpSession();
+ }
+ _debug("After close RTP\n");
+ Manager::instance().peerHungupCall(id);
+ removeCall(id);
+ _debug("After remove call ID\n");
+ }
+
+ void
+ SIPVoIPLink::SIPCallReleased(SIPCall *call)
+ {
+ // do cleanup if exists
+ // only cid because did is always 0 in these case..
+ //SIPCall* call = findSIPCallWithCid(event->cid);
+ if (!call) { return; }
+
+ // if we are here.. something when wrong before...
+ _debug("SIP call release\n");
+ CallID id = call->getCallId();
+ Manager::instance().callFailure(id);
+ removeCall(id);
+ }
+
+ void
+ SIPVoIPLink::SIPCallAnswered(SIPCall *call, pjsip_rx_data *rdata)
+ {
+ //SIPCall* call = dynamic_cast<SIPCall *>(theCall);//findSIPCallWithCid(event->cid);
+ if (!call) {
+ _debug("! SIP Failure: unknown call\n");
+ return;
+ }
+ //call->setDid(event->did);
+
+ if (call->getConnectionState() != Call::Connected) {
+ //call->SIPCallAnswered(event);
+ call->SIPCallAnsweredWithoutHold(rdata);
+
+ call->setConnectionState(Call::Connected);
+ call->setState(Call::Active);
+
+ Manager::instance().peerAnsweredCall(call->getCallId());
+ if (Manager::instance().isCurrentCall(call->getCallId())) {
+ _debug("* SIP Info: Starting AudioRTP when answering\n");
+ if ( _audiortp->createNewSession(call) < 0) {
+ _debug("RTP Failure: unable to create new session\n");
+ } else {
+ call->setAudioStart(true);
+ }
+ }
+ } else {
+ _debug("* SIP Info: Answering call (on/off hold to send ACK)\n");
+ //call->SIPCallAnswered(event);
+ }
+ }
+
+
+ SIPCall*
+ SIPVoIPLink::getSIPCall(const CallID& id)
+ {
+ Call* call = getCall(id);
+ if (call) {
+ return dynamic_cast<SIPCall*>(call);
+ }
+ return NULL;
+ }
+
+ void SIPVoIPLink::setStunServer( const std::string &server )
+ {
+ if(server != "") {
+ useStun(true);
+ _stunServer = server;
+ } else {
+ useStun(false);
+ _stunServer = std::string("");
+ }
+ }
-void
-SIPVoIPLink::SIPCallServerFailure(SIPCall *call)
-{
- //if (!event->response) { return; }
- //switch(event->response->status_code) {
- //case SIP_SERVICE_UNAVAILABLE: // 500
- //case SIP_BUSY_EVRYWHERE: // 600
- //case SIP_DECLINE: // 603
- //SIPCall* call = findSIPCallWithCid(event->cid);
- if (call != 0) {
- _debug("Server error!\n");
- CallID id = call->getCallId();
- Manager::instance().callFailure(id);
- removeCall(id);
- }
- //break;
- //}
-}
+ ///////////////////////////////////////////////////////////////////////////////
+ // Private functions
+ ///////////////////////////////////////////////////////////////////////////////
+
+ bool SIPVoIPLink::pjsip_init()
+ {
+ pj_status_t status;
+ int errPjsip = 0;
+ int port;
+ pjsip_inv_callback inv_cb;
+ pj_str_t accepted;
+ std::string name_mod;
+ bool useStun;
+ validStunServer = true;
+
+ name_mod = "sflphone";
+
+ // Init PJLIB: must be called before any call to the pjsip library
+ status = pj_init();
+ // Use pjsip macros for sanity check
+ PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
+
+ // Init PJLIB-UTIL library
+ status = pjlib_util_init();
+ PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
+
+ // Set the pjsip log level
+ pj_log_set_level( PJ_LOG_LEVEL );
+
+ // Init PJNATH
+ status = pjnath_init();
+ PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
+
+ // Create a pool factory to allocate memory
+ pj_caching_pool_init(&_cp, &pj_pool_factory_default_policy, 0);
+
+ // Create memory pool for application.
+ _pool = pj_pool_create(&_cp.factory, "sflphone", 4000, 4000, NULL);
+
+ if (!_pool) {
+ _debug("UserAgent: Could not initialize memory pool\n");
+ return PJ_ENOMEM;
+ }
+
+ // Create the SIP endpoint
+ status = pjsip_endpt_create(&_cp.factory, pj_gethostname()->ptr, &_endpt);
+ PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
+
+ /* Start resolving STUN server */
+ // if we useStun and we failed to receive something on port 5060, we try a random port
+ // If use STUN server, firewall address setup
+ if (!loadSIPLocalIP()) {
+ _debug("UserAgent: Unable to determine network capabilities\n");
+ return false;
+ }
+
+ port = _regPort;
+
+ /* Retrieve the STUN configuration */
+ useStun = Manager::instance().getConfigInt( SIGNALISATION, SIP_USE_STUN );
+ this->setStunServer(Manager::instance().getConfigString( SIGNALISATION, SIP_STUN_SERVER ));
+ this->useStun( useStun!=0 ? true : false);
+
+ if (useStun && !Manager::instance().behindNat(getStunServer(), port)) {
+ port = RANDOM_SIP_PORT;
+ if (!Manager::instance().behindNat(getStunServer(), port)) {
+ _debug("UserAgent: Unable to check NAT setting\n");
+ validStunServer = false;
+ return false; // hoho we can't use the random sip port too...
+ }
+ }
+
+ _localPort = port;
+ if (useStun) {
+ // set by last behindNat() call (ish)...
+ stunServerResolve();
+ _localExternAddress = Manager::instance().getFirewallAddress();
+ _localExternPort = Manager::instance().getFirewallPort();
+ errPjsip = createUDPServer();
+ if (errPjsip != 0) {
+ _debug("UserAgent: Could not initialize SIP listener on port %d\n", port);
+ return errPjsip;
+ }
+ } else {
+ _localExternAddress = _localIPAddress;
+ _localExternPort = _localPort;
+ errPjsip = createUDPServer();
+ if (errPjsip != 0) {
+ _debug("UserAgent: Could not initialize SIP listener on port %d\n", _localExternPort);
+ _localExternPort = _localPort = RANDOM_SIP_PORT;
+ _debug("UserAgent: Try to initialize SIP listener on port %d\n", _localExternPort);
+ errPjsip = createUDPServer();
+ if (errPjsip != 0) {
+ _debug("UserAgent: Fail to initialize SIP listener on port %d\n", _localExternPort);
+ return errPjsip;
+ }
+ }
+ }
+
+ _debug("UserAgent: SIP Init -- listening on port %d\n", _localExternPort);
+
+ // Initialize transaction layer
+ status = pjsip_tsx_layer_init_module(_endpt);
+ PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
+
+ // Initialize UA layer module
+ status = pjsip_ua_init_module(_endpt, NULL);
+ PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
+
+ // Initialize Replaces support. See the Replaces specification in RFC 3891
+ status = pjsip_replaces_init_module(_endpt);
+ PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
+
+ // Initialize 100rel support
+ status = pjsip_100rel_init_module(_endpt);
+ PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
+
+ // Initialize and register sflphone module
+ _mod_ua.name = pj_str((char*)name_mod.c_str());
+ _mod_ua.id = -1;
+ _mod_ua.priority = PJSIP_MOD_PRIORITY_APPLICATION;
+ _mod_ua.on_rx_request = &mod_on_rx_request;
+ _mod_ua.on_rx_response = &mod_on_rx_response;
+
+ status = pjsip_endpt_register_module(_endpt, &_mod_ua);
+ PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
+
+ // Init the event subscription module.
+ // It extends PJSIP by supporting SUBSCRIBE and NOTIFY methods
+ status = pjsip_evsub_init_module(_endpt);
+ PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
+
+ // Init xfer/REFER module
+ status = pjsip_xfer_init_module(_endpt);
+ PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
+
+ // Init the callback for INVITE session:
+ pj_bzero(&inv_cb, sizeof (inv_cb));
+
+ inv_cb.on_state_changed = &call_on_state_changed;
+ inv_cb.on_new_session = &call_on_forked;
+ inv_cb.on_media_update = &call_on_media_update;
+ inv_cb.on_tsx_state_changed = &call_on_tsx_changed;
+
+ // Initialize session invite module
+ status = pjsip_inv_usage_init(_endpt, &inv_cb);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+
+ _debug("UserAgent: VOIP callbacks initialized\n");
+
+ // Add endpoint capabilities (INFO, OPTIONS, etc) for this UA
+ pj_str_t allowed[] = { {(char*)"INFO", 4}, {(char*)"REGISTER", 8} }; // //{"INVITE", 6}, {"ACK",3}, {"BYE",3}, {"CANCEL",6}, {"OPTIONS", 7},
+ accepted = pj_str((char*)"application/sdp");
+
+ // Register supported methods
+ pjsip_endpt_add_capability(_endpt, &_mod_ua, PJSIP_H_ALLOW, NULL, PJ_ARRAY_SIZE(allowed), allowed);
+
+ // Register "application/sdp" in ACCEPT header
+ pjsip_endpt_add_capability(_endpt, &_mod_ua, PJSIP_H_ACCEPT, NULL, 1, &accepted);
+
+ _debug("UserAgent: pjsip version %s for %s initialized\n", pj_get_version(), PJ_OS_NAME);
+
+ // Create the secondary thread to poll sip events
+ _evThread->start();
+
+ /* Done! */
+ return PJ_SUCCESS;
+ }
-void
-SIPVoIPLink::SIPCallClosed(SIPCall *call)
-{
- // it was without did before
- //SIPCall* call = findSIPCallWithCid(event->cid);
- if (!call) { return; }
+ pj_status_t SIPVoIPLink::stunServerResolve( void )
+ {
+ pj_str_t stun_adr;
+ pj_hostent he;
+ pj_stun_config stunCfg;
+ pj_status_t stun_status;
+ pj_sockaddr stun_srv;
+ size_t pos;
+ std::string serverName, serverPort;
+ int nPort;
+ std::string stun_server;
+
+ stun_server = getStunServer();
+
+ // Initialize STUN configuration
+ pj_stun_config_init(&stunCfg, &_cp.factory, 0, pjsip_endpt_get_ioqueue(_endpt), pjsip_endpt_get_timer_heap(_endpt));
+
+ stun_status = PJ_EPENDING;
+
+ // Init STUN socket
+ pos = stun_server.find(':');
+ if(pos == std::string::npos) {
+ pj_strdup2(_pool, &stun_adr, stun_server.data());
+ stun_status = pj_sockaddr_in_init(&stun_srv.ipv4, &stun_adr, (pj_uint16_t) 3478);
+ } else {
+ serverName = stun_server.substr(0, pos);
+ serverPort = stun_server.substr(pos + 1);
+ nPort = atoi(serverPort.data());
+ pj_strdup2(_pool, &stun_adr, serverName.data());
+ stun_status = pj_sockaddr_in_init(&stun_srv.ipv4, &stun_adr, (pj_uint16_t) nPort);
+ }
+
+ if (stun_status != PJ_SUCCESS) {
+ _debug("UserAgent: Unresolved stun server!\n");
+ stun_status = pj_gethostbyname(&stun_adr, &he);
+
+ if (stun_status == PJ_SUCCESS) {
+ pj_sockaddr_in_init(&stun_srv.ipv4, NULL, 0);
+ stun_srv.ipv4.sin_addr = *(pj_in_addr*) he.h_addr;
+ stun_srv.ipv4.sin_port = pj_htons((pj_uint16_t) 3478);
+ }
+ }
+
+ return stun_status;
+ }
- CallID id = call->getCallId();
- //call->setDid(event->did);
- if (Manager::instance().isCurrentCall(id)) {
- call->setAudioStart(false);
- _debug("* SIP Info: Stopping AudioRTP when closing\n");
- _audiortp->closeRtpSession();
- }
- _debug("After close RTP\n");
- Manager::instance().peerHungupCall(id);
- removeCall(id);
- _debug("After remove call ID\n");
-}
+ int SIPVoIPLink::createUDPServer( void )
+ {
+
+ pj_status_t status;
+ pj_sockaddr_in bound_addr;
+ pjsip_host_port a_name;
+ char tmpIP[32];
+ pj_sock_t sock;
+
+ // Init bound address to ANY
+ pj_memset(&bound_addr, 0, sizeof (bound_addr));
+ bound_addr.sin_addr.s_addr = PJ_INADDR_ANY;
+
+ // Create UDP server socket
+ status = pj_sock_socket(PJ_AF_INET, PJ_SOCK_DGRAM, 0, &sock);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: (%d) UDP socket() error\n", status);
+ return status;
+ }
+
+ status = pj_sock_bind_in(sock, pj_ntohl(bound_addr.sin_addr.s_addr), (pj_uint16_t) _localPort);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: (%d) UDP bind() error\n", status);
+ pj_sock_close(sock);
+ return status;
+ }
+
+ _debug("UserAgent: Use IP: %s\n", _localExternAddress.data());
+
+ // Create UDP-Server (default port: 5060)
+ strcpy(tmpIP, _localExternAddress.data());
+ pj_strdup2(_pool, &a_name.host, tmpIP);
+ a_name.port = (pj_uint16_t) _localExternPort;
+
+ _debug("a_name: host: %s - port : %i\n", a_name.host.ptr, a_name.port);
+
+ status = pjsip_udp_transport_attach(_endpt, sock, &a_name, 1, NULL);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: (%d) Unable to start UDP transport!\n", status);
+ return -1;
+ } else {
+ _debug("UserAgent: UDP server listening on port %d\n", _localExternPort);
+ }
+
+ return PJ_SUCCESS;
+ }
-void
-SIPVoIPLink::SIPCallReleased(SIPCall *call)
-{
- // do cleanup if exists
- // only cid because did is always 0 in these case..
- //SIPCall* call = findSIPCallWithCid(event->cid);
- if (!call) { return; }
-
- // if we are here.. something when wrong before...
- _debug("SIP call release\n");
- CallID id = call->getCallId();
- Manager::instance().callFailure(id);
- removeCall(id);
-}
+ bool SIPVoIPLink::loadSIPLocalIP() {
+
+ bool returnValue = true;
+
+ if (_localIPAddress == "127.0.0.1") {
+ pj_sockaddr ip_addr;
+ if (pj_gethostip(pj_AF_INET(), &ip_addr) != PJ_SUCCESS) {
+ // Update the registration state if no network capabilities found
+ _debug("UserAgent: Get host ip failed!\n");
+ returnValue = false;
+ } else {
+ _localIPAddress = std::string(pj_inet_ntoa(ip_addr.ipv4.sin_addr));
+ _debug("UserAgent: Checking network, setting local IP address to: %s\n", _localIPAddress.data());
+ }
+ }
+ return returnValue;
+ }
-void
-SIPVoIPLink::SIPCallAnswered(SIPCall *call, pjsip_rx_data *rdata)
-{
- //SIPCall* call = dynamic_cast<SIPCall *>(theCall);//findSIPCallWithCid(event->cid);
- if (!call) {
- _debug("! SIP Failure: unknown call\n");
- return;
- }
- //call->setDid(event->did);
-
- if (call->getConnectionState() != Call::Connected) {
- //call->SIPCallAnswered(event);
- call->SIPCallAnsweredWithoutHold(rdata);
-
- call->setConnectionState(Call::Connected);
- call->setState(Call::Active);
+ bool SIPVoIPLink::pjsip_shutdown( void )
+ {
+ /* Destroy endpoint. */
+ if (_endpt) {
+ pjsip_endpt_destroy(_endpt);
+ _endpt = NULL;
+ }
+
+ /* Destroy pool and pool factory. */
+ if (_pool) {
+ pj_pool_release(_pool);
+ _pool = NULL;
+ pj_caching_pool_destroy(&_cp);
+ }
+
+ /* Shutdown PJLIB */
+ pj_shutdown();
+
+ /* Done. */
+ }
- Manager::instance().peerAnsweredCall(call->getCallId());
- if (Manager::instance().isCurrentCall(call->getCallId())) {
- _debug("* SIP Info: Starting AudioRTP when answering\n");
- if ( _audiortp->createNewSession(call) < 0) {
- _debug("RTP Failure: unable to create new session\n");
- } else {
- call->setAudioStart(true);
- }
- }
- } else {
- _debug("* SIP Info: Answering call (on/off hold to send ACK)\n");
- //call->SIPCallAnswered(event);
- }
-}
+ int SIPVoIPLink::getModId(){
+ return _mod_ua.id;
+ }
+ void set_voicemail_info( AccountID account, pjsip_msg_body *body ){
-SIPCall*
-SIPVoIPLink::getSIPCall(const CallID& id)
-{
- Call* call = getCall(id);
- if (call) {
- return dynamic_cast<SIPCall*>(call);
- }
- return NULL;
-}
+ int voicemail, pos_begin, pos_end;
+ std::string voice_str = "Voice-Message: ";
+ std::string delimiter = "/";
+ std::string msg_body, voicemail_str;
-///////////////////////////////////////////////////////////////////////////////
-// Private functions
-///////////////////////////////////////////////////////////////////////////////
+ _debug("UserAgent: checking the voice message!\n");
+ // The voicemail message is formated like that:
+ // Voice-Message: 1/0 . 1 is the number we want to retrieve in this case
-pj_str_t SIPVoIPLink::string2PJStr(const std::string &value)
-{
- char tmp[256];
-
- strcpy(tmp, value.data());
- return pj_str(tmp);
-}
+ // We get the notification body
+ msg_body = (char*)body->data;
+
+ // We need the position of the first character of the string voice_str
+ pos_begin = msg_body.find(voice_str);
+ // We need the position of the delimiter
+ pos_end = msg_body.find(delimiter);
+
+ // So our voicemail number between the both index
+ try {
+
+ voicemail_str = msg_body.substr(pos_begin + voice_str.length(), pos_end - ( pos_begin + voice_str.length()));
+ std::cout << "voicemail number : " << voicemail_str << std::endl;
+ voicemail = atoi( voicemail_str.c_str() );
+ }
+ catch( std::out_of_range& e ){
+ std::cerr << e.what() << std::endl;
+ }
+
+ // We need now to notify the manager
+ if( voicemail != 0 )
+ Manager::instance().startVoiceMessageNotification(account, voicemail);
+ }
+
+ /*******************************/
+ /* CALLBACKS IMPLEMENTATION */
+ /*******************************/
+
+ void call_on_state_changed( pjsip_inv_session *inv, pjsip_event *e){
+
+ PJ_UNUSED_ARG(inv);
+
+ SIPCall *call = reinterpret_cast<SIPCall*> (inv->mod_data[_mod_ua.id]);
+ if(!call)
+ return;
+
+ /* If this is an outgoing INVITE that was created because of
+ * REFER/transfer, send NOTIFY to transferer.
+ */
+ if (call->getXferSub() && e->type==PJSIP_EVENT_TSX_STATE) {
+ int st_code = -1;
+ pjsip_evsub_state ev_state = PJSIP_EVSUB_STATE_ACTIVE;
+
+ switch (call->getInvSession()->state) {
+ case PJSIP_INV_STATE_NULL:
+ case PJSIP_INV_STATE_CALLING:
+ /* Do nothing */
+ break;
+
+ case PJSIP_INV_STATE_EARLY:
+ case PJSIP_INV_STATE_CONNECTING:
+ st_code = e->body.tsx_state.tsx->status_code;
+ ev_state = PJSIP_EVSUB_STATE_ACTIVE;
+ break;
+
+ case PJSIP_INV_STATE_CONFIRMED:
+ /* When state is confirmed, send the final 200/OK and terminate
+ * subscription.
+ */
+ st_code = e->body.tsx_state.tsx->status_code;
+ ev_state = PJSIP_EVSUB_STATE_TERMINATED;
+ break;
+
+ case PJSIP_INV_STATE_DISCONNECTED:
+ st_code = e->body.tsx_state.tsx->status_code;
+ ev_state = PJSIP_EVSUB_STATE_TERMINATED;
+ break;
+
+ case PJSIP_INV_STATE_INCOMING:
+ /* Nothing to do. Just to keep gcc from complaining about
+ * unused enums.
+ */
+ break;
+ }
+
+ if (st_code != -1) {
+ pjsip_tx_data *tdata;
+ pj_status_t status;
+
+ status = pjsip_xfer_notify( call->getXferSub(),
+ ev_state, st_code,
+ NULL, &tdata);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: Unable to create NOTIFY -- %d\n", status);
+ } else {
+ status = pjsip_xfer_send_request(call->getXferSub(), tdata);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: Unable to send NOTIFY -- %d\n", status);
+ }
+ }
+ }
+ }
+ }
+
+ void call_on_media_update( pjsip_inv_session *inv UNUSED, pj_status_t status UNUSED) {
+ _debug("call_on_media_updated\n");
+ }
+
+ void call_on_forked(pjsip_inv_session *inv, pjsip_event *e){
+ _debug("call_on_forked\n");
+ }
+
+ void call_on_tsx_changed(pjsip_inv_session *inv, pjsip_transaction *tsx, pjsip_event *e){
+
+ pjsip_rx_data *rdata;
+ AccountID accId;
+ SIPCall *call;
+ SIPVoIPLink *link;
+ pjsip_msg *msg;
+
+ if(pj_strcmp2(&tsx->method.name, "INFO") == 0) {
+ // Receive a INFO message, ingore it!
+ return;
+ }
+
+ //Retrieve the body message
+ rdata = e->body.tsx_state.src.rdata;
+
+ if (tsx->role == PJSIP_ROLE_UAC) {
+ switch (tsx->state) {
+ case PJSIP_TSX_STATE_TERMINATED:
+ if (tsx->status_code == 200 &&
+ pjsip_method_cmp(&tsx->method, pjsip_get_refer_method()) != 0) {
+ // Peer answered the outgoing call
+ _debug("UserAgent: Peer answered the outgoing call!\n");
+ call = reinterpret_cast<SIPCall *> (inv->mod_data[_mod_ua.id]);
+ if (call == NULL)
+ return;
+
+ //_debug("UserAgent: The call id is %s\n", call->getCallId().data());
+
+ accId = Manager::instance().getAccountFromCall(call->getCallId());
+ link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink(accId));
+ if (link)
+ link->SIPCallAnswered(call, rdata);
+ } else if (tsx->status_code / 100 == 5) {
+ _debug("UserAgent: 5xx error message received\n");
+ }
+ break;
+ case PJSIP_TSX_STATE_PROCEEDING:
+ // Peer is ringing for the outgoing call
+ msg = rdata->msg_info.msg;
+
+ call = reinterpret_cast<SIPCall *> (inv->mod_data[_mod_ua.id]);
+ if (call == NULL)
+ return;
+
+ if (msg->line.status.code == 180) {
+ _debug("UserAgent: Peer is ringing!\n");
+
+ call->setConnectionState(Call::Ringing);
+ Manager::instance().peerRingingCall(call->getCallId());
+ }
+ break;
+ case PJSIP_TSX_STATE_COMPLETED:
+ if (tsx->status_code == 407 || tsx->status_code == 401) //FIXME
+ break;
+ if (tsx->status_code / 100 == 6 || tsx->status_code / 100 == 4) {
+ // We get error message of outgoing call from server
+ _debug("UserAgent: Server error message is received!\n");
+ call = reinterpret_cast<SIPCall *> (inv->mod_data[_mod_ua.id]);
+ if (call == NULL) {
+ _debug("UserAgent: Call has been removed!\n");
+ return;
+ }
+ accId = Manager::instance().getAccountFromCall(call->getCallId());
+ link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink(accId));
+ if (link) {
+ link->SIPCallServerFailure(call);
+ }
+ }
+ break;
+ default:
+ break;
+ } // end of switch
+
+ } else {
+ switch (tsx->state) {
+ case PJSIP_TSX_STATE_TRYING:
+ if (pjsip_method_cmp(&tsx->method, pjsip_get_refer_method()) == 0) {
+ // Peer ask me to transfer call to another number.
+ _debug("UserAgent: Incoming REFER request!\n");
+ //onCallTransfered(inv, e->body.tsx_state.src.rdata);
+ }
+ break;
+ case PJSIP_TSX_STATE_COMPLETED:
+ if (tsx->status_code == 200 && tsx->method.id == PJSIP_BYE_METHOD) {
+ // Peer hangup the call
+ _debug("UserAgent: Peer hangup(bye) message is received!\n");
+ call = reinterpret_cast<SIPCall *> (inv->mod_data[_mod_ua.id]);
+ if (call == NULL) {
+ _debug("UserAgent: Call has been removed!\n");
+ return;
+ }
+ accId = Manager::instance().getAccountFromCall(call->getCallId());
+ link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink(accId));
+ if (link) {
+ link->SIPCallClosed(call);
+ }
+ } else if (tsx->status_code == 200 && tsx->method.id == PJSIP_CANCEL_METHOD) {
+ // Peer refuse the call
+ _debug("UserAgent: Cancel message is received!\n");
+ call = reinterpret_cast<SIPCall *> (inv->mod_data[_mod_ua.id]);
+ if (call == NULL) {
+ _debug("UserAgent: Call has been removed!\n");
+ return;
+ }
+
+ accId = Manager::instance().getAccountFromCall(call->getCallId());
+ link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink(accId));
+ if (link) {
+ link->SIPCallClosed(call);
+ }
+ }
+ break;
+ default:
+ break;
+ } // end of switch
+ }
+ }
+
+ void regc_cb(struct pjsip_regc_cbparam *param){
+
+ //AccountID *id = static_cast<AccountID *> (param->token);
+ SIPAccount *account;
+
+ //_debug("UserAgent: Account ID is %s, Register result: %d, Status: %d\n", id->data(), param->status, param->code);
+ account = static_cast<SIPAccount *>(param->token);
+ if(!account)
+ return;
+
+ if (param->status == PJ_SUCCESS) {
+ if (param->code < 0 || param->code >= 300) {
+ /* Sometimes, the status is OK, but we still failed.
+ * So checking the code for real result
+ */
+ _debug("UserAgent: The error is: %d\n", param->code);
+ switch(param->code) {
+ case 408:
+ case 606:
+ account->setRegistrationState(ErrorConfStun);
+ break;
+ case 503:
+ account->setRegistrationState(ErrorHost);
+ break;
+ case 401:
+ case 403:
+ case 404:
+ account->setRegistrationState(ErrorAuth);
+ break;
+ default:
+ account->setRegistrationState(Error);
+ break;
+ }
+ account->setRegister(false);
+ } else {
+ // Registration/Unregistration is success
+
+ if(account->isRegister())
+ account->setRegistrationState(Registered);
+ else {
+ account->setRegistrationState(Unregistered);
+ account->setRegister(false);
+ }
+ }
+ } else {
+ account->setRegistrationState(ErrorAuth);
+ account->setRegister(false);
+ }
+
+ }
+
+ pj_bool_t
+ mod_on_rx_request(pjsip_rx_data *rdata)
+ {
+
+ pj_status_t status;
+ pj_str_t reason;
+ unsigned options = 0;
+ pjsip_dialog* dialog;
+ pjsip_tx_data *tdata;
+ AccountID account_id;
+ pjsip_uri *uri;
+ pjsip_sip_uri *sip_uri;
+ std::string userName, server, caller, callerServer, peerNumber;
+ SIPVoIPLink *link;
+ CallID id;
+ SIPCall* call;
+
+ // voicemail part
+ std::string method_name;
+ std::string request;
+
+ // Handle the incoming call invite in this function
+ _debug("UserAgent: Callback on_rx_request is involved!\n");
+
+ /* First, let's got the username and server name from the invite.
+ * We will use them to detect which account is the callee.
+ */
+ uri = rdata->msg_info.to->uri;
+ sip_uri = (pjsip_sip_uri *) pjsip_uri_get_uri(uri);
+
+ userName = std::string(sip_uri->user.ptr, sip_uri->user.slen);
+ server = std::string(sip_uri->host.ptr, sip_uri->host.slen) ;
+
+ // Get the account id of callee from username and server
+ account_id = Manager::instance().getAccountIdFromNameAndServer(userName, server);
+
+ /* If we don't find any account to receive the call */
+ if(account_id == AccountNULL) {
+ _debug("UserAgent: Username %s doesn't match any account!\n",userName.c_str());
+ return false;
+ }
+
+ /* Get the voip link associated to the incoming call */
+ /* The account must before have been associated to the call in ManagerImpl */
+ link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink(account_id));
+
+ /* If we can't find any voIP link to handle the incoming call */
+ if( link == 0 )
+ {
+ _debug("ERROR: can not retrieve the voiplink from the account ID...\n");
+ return false;
+ }
+
+ _debug("UserAgent: The receiver is : %s@%s\n", userName.data(), server.data());
+ _debug("UserAgent: The callee account id is %s\n", account_id.c_str());
+
+ /* Now, it is the time to find the information of the caller */
+ uri = rdata->msg_info.from->uri;
+ sip_uri = (pjsip_sip_uri *) pjsip_uri_get_uri(uri);
+
+ /* Retrieve only the fisrt characters */
+ caller = std::string(sip_uri->user.ptr, sip_uri->user.slen);
+ callerServer = std::string(sip_uri->host.ptr, sip_uri->host.slen);
+ peerNumber = caller + "@" + callerServer;
+
+ // Get the server voicemail notification
+ // Catch the NOTIFY message
+ if( rdata->msg_info.msg->line.req.method.id == PJSIP_OTHER_METHOD )
+ {
+ method_name = "NOTIFY";
+ // Retrieve all the message. Should contains only the method name but ...
+ request = rdata->msg_info.msg->line.req.method.name.ptr;
+ // Check if the message is a notification
+ if( request.find( method_name ) != (size_t)-1 ) {
+ /* Notify the right account */
+ set_voicemail_info( account_id, rdata->msg_info.msg->body );
+ }
+ pjsip_endpt_respond_stateless(_endpt, rdata, PJSIP_SC_OK, NULL, NULL, NULL);
+ return true;
+ }
+
+ // Respond statelessly any non-INVITE requests with 500
+ if (rdata->msg_info.msg->line.req.method.id != PJSIP_INVITE_METHOD) {
+ if (rdata->msg_info.msg->line.req.method.id != PJSIP_ACK_METHOD) {
+ pj_strdup2(_pool, &reason, "user agent unable to handle this request ");
+ pjsip_endpt_respond_stateless( _endpt, rdata, PJSIP_SC_METHOD_NOT_ALLOWED, &reason, NULL,
+ NULL);
+ return true;
+ }
+ }
+
+ // Verify that we can handle the request
+ status = pjsip_inv_verify_request(rdata, &options, NULL, NULL, _endpt, NULL);
+ if (status != PJ_SUCCESS) {
+ pj_strdup2(_pool, &reason, "user agent unable to handle this INVITE ");
+ pjsip_endpt_respond_stateless( _endpt, rdata, PJSIP_SC_METHOD_NOT_ALLOWED, &reason, NULL,
+ NULL);
+ return true;
+ }
+
+ // Generate a new call ID for the incoming call!
+ id = Manager::instance().getNewCallID();
+ call = new SIPCall(id, Call::Incoming);
+
+ /* If an error occured at the call creation */
+ if (!call) {
+ _debug("UserAgent: unable to create an incoming call");
+ return false;
+ }
+
+ // Set the codec map, IP, peer number and so on... for the SIPCall object
+ setCallAudioLocal(call, link->getLocalIPAddress(), link->useStun(), link->getStunServer());
+ call->setCodecMap(Manager::instance().getCodecDescriptorMap());
+ call->setConnectionState(Call::Progressing);
+ call->setIp(link->getLocalIPAddress());
+ call->setPeerNumber(peerNumber);
+
+ /* Call the SIPCallInvite function to generate the local sdp,
+ * remote sdp and negociator.
+ * This function is also used to set the parameters of audio RTP, including:
+ * local IP and port number
+ * remote IP and port number
+ * possilbe audio codec will be used in this call
+ */
+ if (call->SIPCallInvite(rdata, _pool)) {
+
+ // Notify UI there is an incoming call
+ if (Manager::instance().incomingCall(call, account_id)) {
+ // Add this call to the callAccountMap in ManagerImpl
+ Manager::instance().getAccountLink(account_id)->addCall(call);
+ } else {
+ // Fail to notify UI
+ delete call;
+ call = NULL;
+ _debug("UserAgent: Fail to notify UI!\n");
+ return false;
+ }
+ } else {
+ // Fail to collect call information
+ delete call;
+ call = NULL;
+ _debug("UserAgent: Call SIPCallInvite failed!\n");
+ return false;
+ }
+
+ /* Create the local dialog (UAS) */
+ status = pjsip_dlg_create_uas(pjsip_ua_instance(), rdata, NULL, &dialog);
+ if (status != PJ_SUCCESS) {
+ pjsip_endpt_respond_stateless( _endpt, rdata, PJSIP_SC_INTERNAL_SERVER_ERROR, &reason, NULL,
+ NULL);
+ return true;
+ }
+
+ // Specify media capability during invite session creation
+ pjsip_inv_session *inv;
+ status = pjsip_inv_create_uas(dialog, rdata, call->getLocalSDPSession(), 0, &inv);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+
+ // Associate the call in the invite session
+ inv->mod_data[_mod_ua.id] = call;
+
+ // Send a 180/Ringing response
+ status = pjsip_inv_initial_answer(inv, rdata, PJSIP_SC_RINGING, NULL, NULL, &tdata);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+ status = pjsip_inv_send_msg(inv, tdata);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+
+ // Associate invite session to the current call
+ call->setInvSession(inv);
+
+ // Update the connection state
+ call->setConnectionState(Call::Ringing);
+
+ /* Done */
+ return true;
+
+ }
+
+ pj_bool_t mod_on_rx_response(pjsip_rx_data *rdata UNUSED) {
+ _debug("mod_on_rx_response\n");
+ return PJ_SUCCESS;
+ }
+
+ void onCallTransfered(pjsip_inv_session *inv, pjsip_rx_data *rdata)
+ {
+ pj_status_t status;
+ pjsip_tx_data *tdata;
+ SIPCall *existing_call;
+ const pj_str_t str_refer_to = { (char*)"Refer-To", 8};
+ const pj_str_t str_refer_sub = { (char*)"Refer-Sub", 9 };
+ const pj_str_t str_ref_by = { (char*)"Referred-By", 11 };
+ pjsip_generic_string_hdr *refer_to;
+ pjsip_generic_string_hdr *refer_sub;
+ pjsip_hdr *ref_by_hdr;
+ pj_bool_t no_refer_sub = PJ_FALSE;
+ char *uri;
+ std::string tmp;
+ pjsip_status_code code;
+ pjsip_evsub *sub;
+
+ existing_call = (SIPCall *) inv->mod_data[_mod_ua.id];
+
+ /* Find the Refer-To header */
+ refer_to = (pjsip_generic_string_hdr*)
+ pjsip_msg_find_hdr_by_name(rdata->msg_info.msg, &str_refer_to, NULL);
+
+ if (refer_to == NULL) {
+ /* Invalid Request.
+ * No Refer-To header!
+ */
+ _debug("UserAgent: Received REFER without Refer-To header!\n");
+ pjsip_dlg_respond( inv->dlg, rdata, 400, NULL, NULL, NULL);
+ return;
+ }
+
+ /* Find optional Refer-Sub header */
+ refer_sub = (pjsip_generic_string_hdr*)
+ pjsip_msg_find_hdr_by_name(rdata->msg_info.msg, &str_refer_sub, NULL);
+
+ if (refer_sub) {
+ if (!pj_strnicmp2(&refer_sub->hvalue, "true", 4)==0)
+ no_refer_sub = PJ_TRUE;
+ }
+
+ /* Find optional Referred-By header (to be copied onto outgoing INVITE
+ * request.
+ */
+ ref_by_hdr = (pjsip_hdr*)
+ pjsip_msg_find_hdr_by_name(rdata->msg_info.msg, &str_ref_by,
+ NULL);
+
+ /* Notify callback */
+ code = PJSIP_SC_ACCEPTED;
+
+ _debug("UserAgent: Call to %.*s is being transfered to %.*s\n",
+ (int)inv->dlg->remote.info_str.slen,
+ inv->dlg->remote.info_str.ptr,
+ (int)refer_to->hvalue.slen,
+ refer_to->hvalue.ptr);
+
+ if (no_refer_sub) {
+ /*
+ * Always answer with 2xx.
+ */
+ pjsip_tx_data *tdata;
+ const pj_str_t str_false = { (char*)"false", 5};
+ pjsip_hdr *hdr;
+
+ status = pjsip_dlg_create_response(inv->dlg, rdata, code, NULL,
+ &tdata);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: Unable to create 2xx response to REFER -- %d\n", status);
+ return;
+ }
+
+ /* Add Refer-Sub header */
+ hdr = (pjsip_hdr*)
+ pjsip_generic_string_hdr_create(tdata->pool, &str_refer_sub,
+ &str_false);
+ pjsip_msg_add_hdr(tdata->msg, hdr);
+
+
+ /* Send answer */
+ status = pjsip_dlg_send_response(inv->dlg, pjsip_rdata_get_tsx(rdata),
+ tdata);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: Unable to create 2xx response to REFER -- %d\n", status);
+ return;
+ }
+
+ /* Don't have subscription */
+ sub = NULL;
+
+ } else {
+ struct pjsip_evsub_user xfer_cb;
+ pjsip_hdr hdr_list;
+
+ /* Init callback */
+ pj_bzero(&xfer_cb, sizeof(xfer_cb));
+ xfer_cb.on_evsub_state = &xfer_svr_cb;
+
+ /* Init addiTHIS_FILE, THIS_FILE, tional header list to be sent with REFER response */
+ pj_list_init(&hdr_list);
+
+ /* Create transferee event subscription */
+ status = pjsip_xfer_create_uas( inv->dlg, &xfer_cb, rdata, &sub);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: Unable to create xfer uas -- %d\n", status);
+ pjsip_dlg_respond( inv->dlg, rdata, 500, NULL, NULL, NULL);
+ return;
+ }
+
+ /* If there's Refer-Sub header and the value is "true", send back
+ * Refer-Sub in the response with value "true" too.
+ */
+ if (refer_sub) {
+ const pj_str_t str_true = { (char*)"true", 4 };
+ pjsip_hdr *hdr;
+
+ hdr = (pjsip_hdr*)
+ pjsip_generic_string_hdr_create(inv->dlg->pool,
+ &str_refer_sub,
+ &str_true);
+ pj_list_push_back(&hdr_list, hdr);
+
+ }
+
+ /* Accept the REFER request, send 2xx. */
+ pjsip_xfer_accept(sub, rdata, code, &hdr_list);
+
+ /* Create initial NOTIFY request */
+ status = pjsip_xfer_notify( sub, PJSIP_EVSUB_STATE_ACTIVE,
+ 100, NULL, &tdata);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: Unable to create NOTIFY to REFER -- %d", status);
+ return;
+ }
+
+ /* Send initial NOTIFY request */
+ status = pjsip_xfer_send_request( sub, tdata);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: Unable to send NOTIFY to REFER -- %d\n", status);
+ return;
+ }
+ }
+
+ /* We're cheating here.
+ * We need to get a null terminated string from a pj_str_t.
+ * So grab the pointer from the hvalue and NULL terminate it, knowing
+ * that the NULL position will be occupied by a newline.
+ */
+ uri = refer_to->hvalue.ptr;
+ uri[refer_to->hvalue.slen] = '\0';
+
+ /* Now make the outgoing call. */
+ tmp = std::string(uri);
+
+ if(existing_call == NULL) {
+ _debug("UserAgent: Call doesn't exist!\n");
+ return;
+ }
+
+ AccountID accId = Manager::instance().getAccountFromCall(existing_call->getCallId());
+ CallID newCallId = Manager::instance().getNewCallID();
+
+ if(!Manager::instance().outgoingCall(accId, newCallId, tmp)) {
+
+ /* Notify xferer about the error (if we have subscription) */
+ if (sub) {
+ status = pjsip_xfer_notify(sub, PJSIP_EVSUB_STATE_TERMINATED,
+ 500, NULL, &tdata);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: Unable to create NOTIFY to REFER -- %d\n", status);
+ return;
+ }
+ status = pjsip_xfer_send_request(sub, tdata);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: Unable to send NOTIFY to REFER -- %d\n", status);
+ return;
+ }
+ }
+ return;
+ }
+
+ SIPCall* newCall;
+ SIPVoIPLink *link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink(accId));
+ if(link) {
+ newCall = dynamic_cast<SIPCall *>(link->getCall(newCallId));
+ if(!newCall) {
+ _debug("UserAgent: can not find the call from sipvoiplink!\n");
+ return;
+ }
+ }
+
+ if (sub) {
+ /* Put the server subscription in inv_data.
+ * Subsequent state changed in pjsua_inv_on_state_changed() will be
+ * reported back to the server subscription.
+ */
+ newCall->setXferSub(sub);
+
+ /* Put the invite_data in the subscription. */
+ pjsip_evsub_set_mod_data(sub, _mod_ua.id,
+ newCall);
+ }
+ }
+
+
+
+
+ void xfer_func_cb( pjsip_evsub *sub, pjsip_event *event){
+
+ PJ_UNUSED_ARG(event);
+
+ _debug("UserAgent: Transfer callback is involved!\n");
+ /*
+ * When subscription is accepted (got 200/OK to REFER), check if
+ * subscription suppressed.
+ */
+ if (pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_ACCEPTED) {
+
+ pjsip_rx_data *rdata;
+ pjsip_generic_string_hdr *refer_sub;
+ const pj_str_t REFER_SUB = {(char*)"Refer-Sub", 9 };
+
+ SIPVoIPLink *link = reinterpret_cast<SIPVoIPLink *> (pjsip_evsub_get_mod_data(sub,
+ _mod_ua.id));
+
+ /* Must be receipt of response message */
+ pj_assert(event->type == PJSIP_EVENT_TSX_STATE &&
+ event->body.tsx_state.type == PJSIP_EVENT_RX_MSG);
+ rdata = event->body.tsx_state.src.rdata;
+
+ /* Find Refer-Sub header */
+ refer_sub = (pjsip_generic_string_hdr*)
+ pjsip_msg_find_hdr_by_name(rdata->msg_info.msg,
+ &REFER_SUB, NULL);
+
+ /* Check if subscription is suppressed */
+ if (refer_sub && pj_stricmp2(&refer_sub->hvalue, "false")==0) {
+ /* Since no subscription is desired, assume that call has been
+ * transfered successfully.
+ */
+ if (link) {
+ // It's the time to stop the RTP
+ link->transferStep2();
+ }
+
+ /* Yes, subscription is suppressed.
+ * Terminate our subscription now.
+ */
+ _debug("UserAgent: Xfer subscription suppressed, terminating event subcription...\n");
+ pjsip_evsub_terminate(sub, PJ_TRUE);
+
+ } else {
+ /* Notify application about call transfer progress.
+ * Initially notify with 100/Accepted status.
+ */
+ _debug("UserAgent: Xfer subscription 100/Accepted received...\n");
+ }
+ }
+ /*
+ * On incoming NOTIFY, notify application about call transfer progress.
+ */
+ else if (pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_ACTIVE ||
+ pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_TERMINATED)
+ {
+ pjsip_msg *msg;
+ pjsip_msg_body *body;
+ pjsip_status_line status_line;
+ pj_bool_t is_last;
+ pj_bool_t cont;
+ pj_status_t status;
+
+ SIPVoIPLink *link = reinterpret_cast<SIPVoIPLink *> (pjsip_evsub_get_mod_data(sub,
+ _mod_ua.id));
+
+ /* When subscription is terminated, clear the xfer_sub member of
+ * the inv_data.
+ */
+ if (pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_TERMINATED) {
+ pjsip_evsub_set_mod_data(sub, _mod_ua.id, NULL);
+ _debug("UserAgent: Xfer client subscription terminated\n");
+
+ }
+
+ if (!link || !event) {
+ /* Application is not interested with call progress status */
+ _debug("UserAgent: Either link or event is empty!\n");
+ return;
+ }
+
+ // Get current call
+ SIPCall *call = dynamic_cast<SIPCall *>(link->getCall(Manager::instance().getCurrentCallId()));
+ if(!call) {
+ _debug("UserAgent: Call doesn't exit!\n");
+ return;
+ }
+
+ /* This better be a NOTIFY request */
+ if (event->type == PJSIP_EVENT_TSX_STATE &&
+ event->body.tsx_state.type == PJSIP_EVENT_RX_MSG)
+ {
+ pjsip_rx_data *rdata;
+
+ rdata = event->body.tsx_state.src.rdata;
+
+ /* Check if there's body */
+ msg = rdata->msg_info.msg;
+ body = msg->body;
+ if (!body) {
+ _debug("UserAgent: Warning! Received NOTIFY without message body\n");
+ return;
+ }
+
+ /* Check for appropriate content */
+ if (pj_stricmp2(&body->content_type.type, "message") != 0 ||
+ pj_stricmp2(&body->content_type.subtype, "sipfrag") != 0)
+ {
+ _debug("UserAgent: Warning! Received NOTIFY with non message/sipfrag content\n");
+ return;
+ }
+
+ /* Try to parse the content */
+ status = pjsip_parse_status_line((char*)body->data, body->len,
+ &status_line);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: Warning! Received NOTIFY with invalid message/sipfrag content\n");
+ return;
+ }
+
+ } else {
+ _debug("UserAgent: Set code to 500!\n");
+ status_line.code = 500;
+ status_line.reason = *pjsip_get_status_text(500);
+ }
+
+ /* Notify application */
+ is_last = (pjsip_evsub_get_state(sub)==PJSIP_EVSUB_STATE_TERMINATED);
+ cont = !is_last;
+
+ if(status_line.code/100 == 2) {
+ _debug("UserAgent: Try to stop rtp!\n");
+ pjsip_tx_data *tdata;
+
+ status = pjsip_inv_end_session(call->getInvSession(), PJSIP_SC_GONE, NULL, &tdata);
+ if(status != PJ_SUCCESS) {
+ _debug("UserAgent: Fail to create end session msg!\n");
+ } else {
+ status = pjsip_inv_send_msg(call->getInvSession(), tdata);
+ if(status != PJ_SUCCESS)
+ _debug("UserAgent: Fail to send end session msg!\n");
+ }
+
+ link->transferStep2();
+ cont = PJ_FALSE;
+ }
+
+ if (!cont) {
+ pjsip_evsub_set_mod_data(sub, _mod_ua.id, NULL);
+ }
+ }
+
+ }
+
+
+ void xfer_svr_cb(pjsip_evsub *sub, pjsip_event *event)
+ {
+ PJ_UNUSED_ARG(event);
+
+ /*
+ * When subscription is terminated, clear the xfer_sub member of
+ * the inv_data.
+ */
+ if (pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_TERMINATED) {
+ SIPCall *call;
+
+ call = (SIPCall*) pjsip_evsub_get_mod_data(sub, _mod_ua.id);
+ if (!call)
+ return;
+
+ pjsip_evsub_set_mod_data(sub, _mod_ua.id, NULL);
+ call->setXferSub(NULL);
+
+ _debug("UserAgent: Xfer server subscription terminated\n");
+ }
+ }
diff --git a/src/sipvoiplink.h b/src/sipvoiplink.h
index 865703e2151eb4345c40e178b5fa2187aa06230a..03b12e6c38ba2d0dc8eae5dd73f3a9a9916f5720 100644
--- a/src/sipvoiplink.h
+++ b/src/sipvoiplink.h
@@ -2,7 +2,7 @@
* Copyright (C) 2004-2009 Savoir-Faire Linux inc.
*
* Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
- * Author: Yan Morin <yan.morin@savoirfairelinux.com>
+ * Author: Yun Liu <yun.liu@savoirfairelinux.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
@@ -23,284 +23,316 @@
#define SIPVOIPLINK_H
#include "voiplink.h"
-#include "useragent.h"
+
+//////////////////////////////
+/* PJSIP imports */
+#include <pjsip.h>
+#include <pjlib.h>
+#include <pjsip_ua.h>
+#include <pjlib-util.h>
+#include <pjnath/stun_config.h>
+///////////////////////////////
+
class EventThread;
class SIPCall;
class AudioRtp;
+#define RANDOM_LOCAL_PORT ((rand() % 27250) + 5250)*2
+#define RANDOM_SIP_PORT rand() % 64000 + 1024
+
+// To set the verbosity. From 0 (min) to 6 (max)
+#define PJ_LOG_LEVEL 1
+
/**
* @file sipvoiplink.h
- * @brief Specific VoIPLink for SIP (SIP core for incoming and outgoing events)
+ * @brief Specific VoIPLink for SIP (SIP core for incoming and outgoing events).
+ * This class is based on the singleton design pattern.
+ * One SIPVoIPLink can handle multiple SIP accounts, but all the SIP accounts have all the same SIPVoIPLink
*/
class SIPVoIPLink : public VoIPLink
{
- public:
-
- /**
- * Constructor
- * @param accountID The account identifier
- */
- SIPVoIPLink(const AccountID& accountID);
-
- /**
- * Destructor
- */
- ~SIPVoIPLink();
-
- /* Copy Constructor */
- SIPVoIPLink(const SIPVoIPLink& rh);
-
- /* Assignment Operator */
- SIPVoIPLink& operator=( const SIPVoIPLink& rh);
-
- /**
- * Try to initiate the pjsip engine/thread and set config
- * @return bool True if OK
- */
- bool init(void);
-
- /**
- * Delete link-related stuuf like calls
- */
- void terminate(void);
-
- /**
- * Event listener. Each event send by the call manager is received and handled from here
- */
- void getEvent(void);
-
- /**
- * Build and send SIP registration request
- * @return bool True on success
- * false otherwise
- */
- int sendRegister(void);
-
- /**
- * Build and send SIP unregistration request
- * @return bool True on success
- * false otherwise
- */
- int sendUnregister(void);
-
- /**
- * Place a new call
- * @param id The call identifier
- * @param toUrl The Sip address of the recipient of the call
- * @return Call* The current call
- */
- Call* newOutgoingCall(const CallID& id, const std::string& toUrl);
-
- /**
- * Answer the call
- * @param id The call identifier
- * @return bool True on success
- */
- bool answer(const CallID& id);
-
- /**
- * Hang up the call
- * @param id The call identifier
- * @return bool True on success
- */
- bool hangup(const CallID& id);
-
- /**
- * Cancel the call
- * @param id The call identifier
- * @return bool True on success
- */
- bool cancel(const CallID& id);
-
- /**
- * Put the call on hold
- * @param id The call identifier
- * @return bool True on success
- */
- bool onhold(const CallID& id);
-
- /**
- * Put the call off hold
- * @param id The call identifier
- * @return bool True on success
- */
- bool offhold(const CallID& id);
-
- /**
- * Transfer the call
- * @param id The call identifier
- * @param to The recipient of the transfer
- * @return bool True on success
- */
- bool transfer(const CallID& id, const std::string& to);
-
- /** Handle the incoming refer msg, not finished yet */
- bool transferStep2();
-
- /**
- * Refuse the call
- * @param id The call identifier
- * @return bool True on success
- */
- bool refuse (const CallID& id);
-
- /**
- * Send DTMF
- * @param id The call identifier
- * @param code The char code
- * @return bool True on success
- */
- bool carryingDTMFdigits(const CallID& id, char code);
-
- /**
- * If set to true, we check for a firewall
- * @param use true if we use STUN
- */
- void setUseStun(bool use) { _useStun = use; }
-
- /**
- * The name of the STUN server
- * @param server Server FQDN/IP
- */
- void setStunServer(const std::string& server) { _stunServer = server; }
-
- bool isRegister() {return _bRegister;}
-
- void setRegister(bool result) {_bRegister = result;}
-
- public:
-
- /**
- * Terminate every call not hangup | brutal | Protected by mutex
- */
- void terminateSIPCall();
-
- /**
- * send SIP authentification
- * @return bool true if sending succeed
- */
- bool sendSIPAuthentification();
-
- /**
- * Get a SIP From header ("fullname" <sip:userpart@hostpart>)
- * @param userpart User part
- * @param hostpart Host name
- * @return std::string SIP URI for from Header
- */
- std::string SIPFromHeader(const std::string& userpart, const std::string& hostpart);
-
- /**
- * Build a sip address with the number that you want to call
- * Example: sip:124@domain.com
- * @param to The header of the recipient
- * @return std::string Result as a string
- */
- std::string SIPToHeader(const std::string& to);
-
- /**
- * Check if an url is sip-valid
- * @param url The url to check
- * @return bool True if osip tell that is valid
- */
- bool SIPCheckUrl(const std::string& url);
-
-
- /**
- * Send an outgoing call invite
- * @param call The current call
- * @return bool True if all is correct
- */
- bool SIPOutgoingInvite(SIPCall* call);
-
- /**
- * Start a SIP Call
- * @param call The current call
- * @param subject Undocumented
- * @return true if all is correct
- */
- bool SIPStartCall(SIPCall* call, const std::string& subject);
-
- /**
- * Get the Sip FROM url (add sip:, add @host, etc...)
- * @return std::string The From url
- */
- std::string getSipFrom();
-
- /**
- * Get the Sip TO url (add sip:, add @host, etc...)
- * @param to_url The To url
- * @return std::string The SIP to address
- */
- std::string getSipTo(const std::string& to_url);
-
- /**
- * Set audio (SDP) configuration for a call
- * localport, localip, localexternalport
- * @param call a SIPCall valid pointer
- * @return bool True
- */
- bool setCallAudioLocal(SIPCall* call);
-
- /**
- * Tell the user that the call was answered
- * @param
- */
- void SIPCallAnswered(SIPCall *call, pjsip_rx_data *rdata);
-
- /**
- * Handling 5XX/6XX error
- * @param
- */
- void SIPCallServerFailure(SIPCall *call);
-
- /**
- * Peer close the connection
- * @param
- */
- void SIPCallClosed(SIPCall *call);
-
- /**
- * The call pointer was released
- * If the call was not cleared before, report an error
- * @param
- */
- void SIPCallReleased(SIPCall *call);
-
- /**
- * SIPCall accessor
- * @param id The call identifier
- * @return SIPCall* A pointer on SIPCall object
- */
- SIPCall* getSIPCall(const CallID& id);
-
- /** Tell if the initialisation was done */
- bool _initDone;
-
- /** when we init the listener, how many times we try to bind a port? */
- int _nbTryListenAddr;
-
- /** Do we use stun? */
- bool _useStun;
-
- /** What is the stun server? */
- std::string _stunServer;
-
- /** Local Extern Address is the IP address seen by peers for SIP listener */
- std::string _localExternAddress;
-
- /** Local Extern Port is the port seen by peers for SIP listener */
- unsigned int _localExternPort;
-
- /** Starting sound */
- AudioRtp* _audiortp;
-
- pj_str_t string2PJStr(const std::string &value);
-
-private:
- pjsip_regc *_regc;
- bool _bRegister;
+ public:
+
+ /**
+ * Singleton method. Enable to retrieve the unique static instance
+ * @return SIPVoIPLink* A pointer on the object
+ */
+ static SIPVoIPLink* instance( const AccountID& id );
+
+ /**
+ * Destructor
+ */
+ ~SIPVoIPLink();
+
+ /* Copy Constructor */
+ SIPVoIPLink(const SIPVoIPLink& rh);
+
+ /* Assignment Operator */
+ SIPVoIPLink& operator=( const SIPVoIPLink& rh);
+
+ /**
+ * Try to initiate the pjsip engine/thread and set config
+ * @return bool True if OK
+ */
+ bool init(void);
+
+ /**
+ * Shut the library and clean up
+ */
+ void terminate( void );
+
+ /**
+ * Event listener. Each event send by the call manager is received and handled from here
+ */
+ void getEvent(void);
+
+ /**
+ * Build and send SIP registration request
+ * @return bool True on success
+ * false otherwise
+ */
+ int sendRegister(AccountID id);
+
+ /**
+ * Build and send SIP unregistration request
+ * @return bool True on success
+ * false otherwise
+ */
+ int sendUnregister(AccountID id);
+
+ /**
+ * Place a new call
+ * @param id The call identifier
+ * @param toUrl The Sip address of the recipient of the call
+ * @return Call* The current call
+ */
+ Call* newOutgoingCall(const CallID& id, const std::string& toUrl);
+
+ /**
+ * Answer the call
+ * @param id The call identifier
+ * @return bool True on success
+ */
+ bool answer(const CallID& id);
+
+ /**
+ * Hang up the call
+ * @param id The call identifier
+ * @return bool True on success
+ */
+ bool hangup(const CallID& id);
+
+ /**
+ * Cancel the call
+ * @param id The call identifier
+ * @return bool True on success
+ */
+ bool cancel(const CallID& id);
+
+ /**
+ * Put the call on hold
+ * @param id The call identifier
+ * @return bool True on success
+ */
+ bool onhold(const CallID& id);
+
+ /**
+ * Put the call off hold
+ * @param id The call identifier
+ * @return bool True on success
+ */
+ bool offhold(const CallID& id);
+
+ /**
+ * Transfer the call
+ * @param id The call identifier
+ * @param to The recipient of the transfer
+ * @return bool True on success
+ */
+ bool transfer(const CallID& id, const std::string& to);
+
+ /** Handle the incoming refer msg, not finished yet */
+ bool transferStep2();
+
+ /**
+ * Refuse the call
+ * @param id The call identifier
+ * @return bool True on success
+ */
+ bool refuse (const CallID& id);
+
+ /**
+ * Send DTMF
+ * @param id The call identifier
+ * @param code The char code
+ * @return bool True on success
+ */
+ bool carryingDTMFdigits(const CallID& id, char code);
+
+ /**
+ * If set to true, we check for a firewall
+ * @param use true if we use STUN
+ */
+ inline void useStun(bool use) { _useStun=use; }
+
+ inline bool useStun( void ) { return _useStun; }
+
+ /**
+ * The name of the STUN server
+ * @param server Server FQDN/IP
+ */
+ void setStunServer(const std::string& server);
+
+ std::string getStunServer (void) { return _stunServer; }
+
+ /**
+ * Terminate every call not hangup | brutal | Protected by mutex
+ */
+ void terminateSIPCall();
+
+ /**
+ * Build a sip address with the number that you want to call
+ * Example: sip:124@domain.com
+ * @param to The header of the recipient
+ * @return std::string Result as a string
+ */
+ std::string SIPToHeader(const std::string& to);
+
+ /**
+ * Check if an url is sip-valid
+ * @param url The url to check
+ * @return bool True if osip tell that is valid
+ */
+ bool SIPCheckUrl(const std::string& url);
+
+
+ /**
+ * Send an outgoing call invite
+ * @param call The current call
+ * @return bool True if all is correct
+ */
+ bool SIPOutgoingInvite(SIPCall* call);
+
+ /**
+ * Start a SIP Call
+ * @param call The current call
+ * @param subject Undocumented
+ * @return true if all is correct
+ */
+ bool SIPStartCall(SIPCall* call, const std::string& subject);
+
+ /**
+ * Get the Sip TO url (add sip:, add @host, etc...)
+ * @param to_url The To url
+ * @return std::string The SIP to address
+ */
+ std::string getSipTo(const std::string& to_url, std::string hostname);
+
+ /**
+ * Tell the user that the call was answered
+ * @param
+ */
+ void SIPCallAnswered(SIPCall *call, pjsip_rx_data *rdata);
+
+ /**
+ * Handling 5XX/6XX error
+ * @param
+ */
+ void SIPCallServerFailure(SIPCall *call);
+
+ /**
+ * Peer close the connection
+ * @param
+ */
+ void SIPCallClosed(SIPCall *call);
+
+ /**
+ * The call pointer was released
+ * If the call was not cleared before, report an error
+ * @param
+ */
+ void SIPCallReleased(SIPCall *call);
+
+ /**
+ * SIPCall accessor
+ * @param id The call identifier
+ * @return SIPCall* A pointer on SIPCall object
+ */
+ SIPCall* getSIPCall(const CallID& id);
+
+ /** when we init the listener, how many times we try to bind a port? */
+ int _nbTryListenAddr;
+
+ /** Starting sound */
+ AudioRtp* _audiortp;
+
+ /** Increment the number of SIP account connected to this link */
+ void incrementClients (void) { _clients++; }
+
+ /** Decrement the number of SIP account connected to this link */
+ void decrementClients (void);
+
+ private:
+ /**
+ * Constructor
+ * @param accountID The account identifier
+ */
+ SIPVoIPLink(const AccountID& accountID);
+
+ /* The singleton instance */
+ static SIPVoIPLink* _instance;
+
+ int getModId();
+
+ /**
+ * Initialize the PJSIP library
+ * Must be called before any other calls to the SIP layer
+ *
+ * @return bool True on success
+ */
+ bool pjsip_init();
+
+ /**
+ * Delete link-related stuuf like calls
+ */
+ bool pjsip_shutdown(void);
+
+ /** Do we use stun? */
+ bool _useStun;
+
+ pj_status_t stunServerResolve();
+
+ /** Create SIP UDP Listener */
+ int createUDPServer();
+
+ bool loadSIPLocalIP();
+
+ std::string getLocalIP() {return _localExternAddress;}
+
+ /** For registration use only */
+ int _regPort;
+
+ /* Flag to check if the STUN server is valid or not */
+ bool validStunServer;
+
+ /** The current STUN server address */
+ std::string _stunServer;
+
+ /** Local Extern Address is the IP address seen by peers for SIP listener */
+ std::string _localExternAddress;
+
+ /** Local Extern Port is the port seen by peers for SIP listener */
+ unsigned int _localExternPort;
+
+ /** Threading object */
+ EventThread* _evThread;
+ ost::Mutex _mutexSIP;
+
+ /* Number of SIP accounts connected to the link */
+ int _clients;
+
};
#endif
diff --git a/src/user_cfg.h b/src/user_cfg.h
index 57201f084be7353e84750920d6b4aa74909b64f6..438911822771e894d8c42decabc4a88c8c46be28 100644
--- a/src/user_cfg.h
+++ b/src/user_cfg.h
@@ -69,6 +69,8 @@
#define PULSE_LENGTH "DTMF.pulseLength" /** Length of the DTMF in millisecond */
#define SEND_DTMF_AS "DTMF.sendDTMFas" /** DTMF send mode */
#define SYMMETRIC "VoIPLink.symmetric" /** VoIP link type */
+#define STUN_ENABLE "STUN.enable"
+#define STUN_SERVER "STUN.server"
#define EMPTY_FIELD "" /** Default value for empty field */
#define DFT_STUN_SERVER "stun.fwdnet.net:3478" /** Default STUN server address */
@@ -89,6 +91,7 @@
#define DFT_EXPIRE_VALUE "180" /** Default expire value for registration */
#define DFT_AUDIO_MANAGER "1" /** Default audio manager */
#define DFT_SIP_PORT "5060"
+#define DFT_STUN_ENABLE "0"
#ifdef USE_ZEROCONF
#define CONFIG_ZEROCONF_DEFAULT_STR "1" /** Default Zero configuration networking module value */
diff --git a/src/useragent.cpp b/src/useragent.cpp
deleted file mode 100644
index fb527784e0a41d0754228debb329860589370d0c..0000000000000000000000000000000000000000
--- a/src/useragent.cpp
+++ /dev/null
@@ -1,1704 +0,0 @@
-/*
- * Copyright (C) 2004-2005 Savoir-Faire Linux inc.
- * Author: Yun Liu <yun.liu@savoirfairelinux.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
- */
-
-#include <string>
-#include <iostream>
-
-#include "manager.h"
-#include "sipcall.h"
-#include "useragent.h"
-#include "sipvoiplink.h"
-#include "sipaccount.h"
-
-#define RANDOM_SIP_PORT rand() % 64000 + 1024
-#define RANDOM_LOCAL_PORT ((rand() % 27250) + 5250)*2
-
-UserAgent *UserAgent::_current;
-
-UserAgent::UserAgent():_endpt(NULL) ,_sock(NULL), _cp(), _pool(NULL), _mutex(NULL), _mod(), _useStun(false), _stunHost(),
- _stunServer(""), _localExternAddress(""), _localIPAddress("127.0.0.1"), _localExternPort(0), _localPort(0), _regPort(DEFAULT_SIP_PORT), _thread(NULL) {
- //_useStun = false;
- //_localIPAddress = "127.0.0.1";
- UserAgent::_current = this;
-}
-
-UserAgent::~UserAgent() {
- _debug("UserAgent: In dtor!\n");
- sipDestory();
-}
-
-pj_status_t UserAgent::sipCreate() {
-
- pj_status_t status;
-
- // Init PJLIB: must be called before any call to the pjsip library
- status = pj_init();
- // Use pjsip macros for sanity check
- PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
-
- // Init PJLIB-UTIL library
- status = pjlib_util_init();
- PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
-
- // Set the pjsip log level
- pj_log_set_level( PJ_LOG_LEVEL );
-
- // Init PJNATH
- status = pjnath_init();
- PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
-
- // Create a pool factory to allocate memory
- pj_caching_pool_init(&_cp, &pj_pool_factory_default_policy, 0);
-
- // Create memory pool for application.
- _pool = pj_pool_create(&_cp.factory, "sflphone", 4000, 4000, NULL);
-
- if (!_pool) {
- _debug("UserAgent: Could not initialize memory pool\n");
- return PJ_ENOMEM;
- }
-
- // Create a recursive mutex. Simple wrapper for pj_mutex_create
- status = pj_mutex_create_recursive(_pool, "sflphone", &_mutex);
- PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
-
- // Create the SIP endpoint
- status = pjsip_endpt_create(&_cp.factory, pj_gethostname()->ptr, &_endpt);
- PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
-
- return PJ_SUCCESS;
-}
-
-pj_status_t UserAgent::sipInit() {
-
- pj_status_t status;
- int errPjsip = 0;
- int port;
-
- validStunServer = true;
- /* Init SIP UA: */
-
- //FIXME! DNS initialize here! */
-
- /* Start resolving STUN server */
- // if we useStun and we failed to receive something on port 5060, we try a random port
- // If use STUN server, firewall address setup
- if (!loadSIPLocalIP()) {
- _debug("UserAgent: Unable to determine network capabilities\n");
- return false;
- }
- errPjsip = 0;
- port = _regPort;
-
- //_debug("stun host is %s\n", _stunHost.ptr);
- if (_useStun && !Manager::instance().behindNat(_stunServer, port)) {
- port = RANDOM_SIP_PORT;
- if (!Manager::instance().behindNat(_stunServer, port)) {
- _debug("UserAgent: Unable to check NAT setting\n");
- validStunServer = false;
- return false; // hoho we can't use the random sip port too...
- }
- }
-
- _localPort = port;
- if (_useStun) {
- // set by last behindNat() call (ish)...
- stunServerResolve();
- _localExternAddress = Manager::instance().getFirewallAddress();
- _localExternPort = Manager::instance().getFirewallPort();
- errPjsip = createUDPServer();
- if (errPjsip != 0) {
- _debug("UserAgent: Could not initialize SIP listener on port %d\n", port);
- return errPjsip;
- }
- } else {
- _localExternAddress = _localIPAddress;
- _localExternPort = _localPort;
- errPjsip = createUDPServer();
- if (errPjsip != 0) {
- _debug("UserAgent: Could not initialize SIP listener on port %d\n", _localExternPort);
- _localExternPort = _localPort = RANDOM_SIP_PORT;
- _debug("UserAgent: Try to initialize SIP listener on port %d\n", _localExternPort);
- errPjsip = createUDPServer();
- if (errPjsip != 0) {
- _debug("UserAgent: Fail to initialize SIP listener on port %d\n", _localExternPort);
- return errPjsip;
- }
- }
- }
-
- _debug("UserAgent: SIP Init -- listening on port %d\n", _localExternPort);
-
- // Initialize transaction layer
- status = pjsip_tsx_layer_init_module(_endpt);
- PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
-
- // Initialize UA layer module
- status = pjsip_ua_init_module(_endpt, NULL);
- PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
-
- // Initialize Replaces support. See the Replaces specification in RFC 3891
- status = pjsip_replaces_init_module(_endpt);
- PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
-
- // Initialize 100rel support
- status = pjsip_100rel_init_module(_endpt);
- PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
-
- // Initialize and register sflphone module
- {
- const pjsip_module mod_initializer ={
- NULL, NULL, // prev, next.
- { (char*)"mod-sflphone", 9}, // Name.
- -1, // Id
- PJSIP_MOD_PRIORITY_APPLICATION, // Priority
- NULL, // load()
- NULL, // start()
- NULL, // stop()
- NULL, // unload()
- &mod_on_rx_request, // on_rx_request()
- &mod_on_rx_response, // on_rx_response()
- NULL, // on_tx_request.
- NULL, // on_tx_response()
- NULL, // on_tsx_state()
- };
-
- _mod = mod_initializer;
-
- status = pjsip_endpt_register_module(_endpt, &_mod);
- PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
-
- }
-
- // Init the event subscription module.
- // It extends PJSIP by supporting SUBSCRIBE and NOTIFY methods
- status = pjsip_evsub_init_module(_endpt);
- PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
-
- // Init presence module.
- // TODO We probably do not need that extension
- status = pjsip_pres_init_module(_endpt, pjsip_evsub_instance());
- PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
-
- // Init PUBLISH module
- // Provide an implementation of SIP Extension for Event State Publication (RFC 3903)
- // TODO Check if it is necessary
- status = pjsip_publishc_init_module(_endpt);
- PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
-
- // Init xfer/REFER module
- status = pjsip_xfer_init_module(_endpt);
- PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
-
- // Initialize invite session module
- // These callbacks will be called on incoming requests, media session state, etc.
- {
- pjsip_inv_callback inv_cb;
-
- // Init the callback for INVITE session:
- pj_bzero(&inv_cb, sizeof (inv_cb));
-
- inv_cb.on_state_changed = &call_on_state_changed;
- inv_cb.on_new_session = &call_on_forked;
- inv_cb.on_media_update = &call_on_media_update;
- inv_cb.on_tsx_state_changed = &call_on_tsx_changed;
-
- _debug("UserAgent: VOIP callbacks initialized\n");
-
- // Initialize session invite module
- status = pjsip_inv_usage_init(_endpt, &inv_cb);
- PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
- }
-
-
- // Add endpoint capabilities (INFO, OPTIONS, etc) for this UA
- {
- pj_str_t allowed[] = {
- {(char*)"INFO", 4},
- {(char*)"REGISTER", 8}
- }; // //{"INVITE", 6}, {"ACK",3}, {"BYE",3}, {"CANCEL",6}, {"OPTIONS", 7},
- pj_str_t accepted = {(char*)"application/sdp", 15};
-
- // Register supported methods
- pjsip_endpt_add_capability(_endpt, &_mod, PJSIP_H_ALLOW, NULL, PJ_ARRAY_SIZE(allowed), allowed);
-
- // Register "application/sdp" in ACCEPT header
- pjsip_endpt_add_capability(_endpt, &_mod, PJSIP_H_ACCEPT, NULL, 1, &accepted);
- }
-
- _debug("UserAgent: pjsip version %s for %s initialized\n", pj_get_version(), PJ_OS_NAME);
-
- Manager::instance().setSipThreadStatus(false);
-
- // Create the secondary thread to poll sip events
- status = pj_thread_create(_pool, "sflphone", &start_thread, NULL, PJ_THREAD_DEFAULT_STACK_SIZE, 0,
- &_thread);
- PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
-
- /* Done! */
- return PJ_SUCCESS;
-
-}
-
-void UserAgent::sipDestory() {
- /* Signal threads to quit: */
- Manager::instance().setSipThreadStatus(true);
-
- /* Wait worker thread to quit: */
- if (_thread) {
- pj_thread_join(_thread);
- pj_thread_destroy(_thread);
- _thread = NULL;
- }
-
- if (_endpt) {
- /* Terminate all presence subscriptions. */
- //pjsua_pres_shutdown();
-
- /* Wait for some time to allow unregistration to complete: */
- _debug("UserAgent: Shutting down...\n");
- busy_sleep(1000);
- }
-
- /* Destroy endpoint. */
- if (_endpt) {
- pjsip_endpt_destroy(_endpt);
- _endpt = NULL;
- }
-
- /* Destroy mutex */
- if (_mutex) {
- pj_mutex_destroy(_mutex);
- _mutex = NULL;
- }
-
- /* Destroy pool and pool factory. */
- if (_pool) {
- pj_pool_release(_pool);
- _pool = NULL;
- pj_caching_pool_destroy(&_cp);
-
- /* Shutdown PJLIB */
- pj_shutdown();
- }
-
- /* Done. */
-}
-
-void UserAgent::busy_sleep(unsigned msec)
-{
-#if defined(PJ_SYMBIAN) && PJ_SYMBIAN != 0
- /* Ideally we shouldn't call pj_thread_sleep() and rather
- * CActiveScheduler::WaitForAnyRequest() here, but that will
- * drag in Symbian header and it doesn't look pretty.
- */
- pj_thread_sleep(msec);
-#else
- pj_time_val timeout, now, tv;
-
- pj_gettimeofday(&timeout);
- timeout.msec += msec;
- pj_time_val_normalize(&timeout);
-
- tv.sec = 0;
- tv.msec = 10;
- pj_time_val_normalize(&tv);
-
- do {
- pjsip_endpt_handle_events(_endpt, &tv);
- pj_gettimeofday(&now);
- } while (PJ_TIME_VAL_LT(now, timeout));
-#endif
-}
-
-bool UserAgent::addAccount(AccountID id, pjsip_regc **regc2, const std::string& server, const std::string& user, const std::string& passwd, const int& timeout UNUSED) {
-
- pj_status_t status;
- AccountID *currentId = new AccountID(id);
- char contactTmp[256];
- pjsip_regc *regc;
- pj_str_t svr;
- pj_str_t aor;
- pj_str_t contact;
- pjsip_tx_data *tdata;
-
- //pj_mutex_lock(_mutex);
- std::string tmp;
-
-
- SIPAccount *account;
-
- if (!validStunServer) {
-
- SIPVoIPLink *voipLink;
- voipLink = dynamic_cast<SIPVoIPLink *>(Manager::instance().getAccountLink(id));
- Manager::instance().getAccountLink(id)->setRegistrationState(VoIPLink::ErrorExistStun);
- voipLink->setRegister(false);
- return false;
- }
-
- status = pjsip_regc_create(_endpt, (void *) currentId, ®c_cb, ®c);
- if (status != PJ_SUCCESS) {
- _debug("UserAgent: Unable to create regc.\n");
- return false;
- }
-
- tmp = "sip:" + server;
- pj_strdup2(_pool, &svr, tmp.data());
-
- tmp = "<sip:" + user + "@" + server + ">";
- pj_strdup2(_pool, &aor, tmp.data());
-
-
- sprintf(contactTmp, "<sip:%s@%s:%d>", user.data(), _localExternAddress.data(), _localExternPort);
- pj_strdup2(_pool, &contact, contactTmp);
-
- //_debug("UserAgent: Get in %s %d %s\n", svr.ptr, svr.slen, aor.ptr);
- status = pjsip_regc_init(regc, &svr, &aor, &aor, 1, &contact, 600); //timeout);
- if (status != PJ_SUCCESS) {
- _debug("UserAgent: Unable to initialize regc. %d\n", status); //, regc->str_srv_url.ptr);
- //pj_mutex_unlock(_mutex);
- return false;
- }
-
-
- account = dynamic_cast<SIPAccount *> (Manager::instance().getAccount(id));
- pjsip_cred_info *cred = account->getCredInfo();
-
- if(!cred)
- cred = new pjsip_cred_info();
-
- pj_bzero(cred, sizeof (pjsip_cred_info));
- pj_strdup2(_pool, &cred->username, user.data());
- cred->data_type = PJSIP_CRED_DATA_PLAIN_PASSWD;
- pj_strdup2(_pool, &cred->data, passwd.data());
- pj_strdup2(_pool, &cred->realm, "*");
- pj_strdup2(_pool, &cred->scheme, "digest");
- pjsip_regc_set_credentials(regc, 1, cred);
-
- account->setCredInfo(cred);
-
- status = pjsip_regc_register(regc, PJ_TRUE, &tdata);
- if (status != PJ_SUCCESS) {
- _debug("UserAgent: Unable to register regc.\n");
- //pj_mutex_unlock(_mutex);
- return false;
- }
-
- status = pjsip_regc_send(regc, tdata);
- if (status != PJ_SUCCESS) {
- _debug("UserAgent: Unable to send regc request.\n");
- //pj_mutex_unlock(_mutex);
- return false;
- }
-
- account->setUserName(user);
- account->setServer(server);
- account->setContact(contactTmp);
-
- // associate regc with account
- *regc2 = regc;
-
- //pj_mutex_unlock(_mutex);
-
- return true;
-}
-
-bool UserAgent::removeAccount(pjsip_regc *regc)
-{
- pj_status_t status = 0;
- pjsip_tx_data *tdata = NULL;
-
- //pj_mutex_lock(_mutex);
- if(regc) {
- status = pjsip_regc_unregister(regc, &tdata);
- if(status != PJ_SUCCESS) {
- _debug("UserAgent: Unable to unregister regc.\n");
- //pj_mutex_unlock(_mutex);
- return false;
- }
-
- status = pjsip_regc_send( regc, tdata );
- if(status != PJ_SUCCESS) {
- _debug("UserAgent: Unable to send regc request.\n");
- //pj_mutex_unlock(_mutex);
- return false;
- }
- } else {
- _debug("UserAgent: regc is null!\n");
- //pj_mutex_unlock(_mutex);
- return false;
- }
-
- //pj_mutex_unlock(_mutex);
- return true;
-}
-
-pj_str_t UserAgent::buildContact(char *userName) {
- //pj_str_t contact;
- char tmp[256];
-
- //FIXME: IPV6 issue!!
- _debug("In build Contact %s %s %d\n", userName, _localExternAddress.data(), _localExternPort);
- sprintf(tmp, "<sip:%s@%s:%d>", userName, _localExternAddress.data(), _localExternPort);
- //_debug("get tmp\n");
- return pj_str(tmp);
-}
-
-pj_status_t UserAgent::stunServerResolve() {
- pj_str_t stun_adr;
- pj_hostent he;
- pj_stun_config stunCfg;
- pj_status_t stun_status;
- pj_sockaddr stun_srv;
-
- // Initialize STUN configuration
- pj_stun_config_init(&stunCfg, &_cp.factory, 0, pjsip_endpt_get_ioqueue(_endpt), pjsip_endpt_get_timer_heap(_endpt));
-
- stun_status = PJ_EPENDING;
-
- // Init STUN socket
- size_t pos = _stunServer.find(':');
- if(pos == std::string::npos) {
- pj_strdup2(_pool, &stun_adr, _stunServer.data());
- stun_status = pj_sockaddr_in_init(&stun_srv.ipv4, &stun_adr, (pj_uint16_t) 3478);
- } else {
- std::string serverName = _stunServer.substr(0, pos);
- std::string serverPort = _stunServer.substr(pos + 1);
- int nPort = atoi(serverPort.data());
- pj_strdup2(_pool, &stun_adr, serverName.data());
- stun_status = pj_sockaddr_in_init(&stun_srv.ipv4, &stun_adr, (pj_uint16_t) nPort);
- }
-
- if (stun_status != PJ_SUCCESS) {
- _debug("UserAgent: Unresolved stun server!\n");
- stun_status = pj_gethostbyname(&stun_adr, &he);
-
- if (stun_status == PJ_SUCCESS) {
- pj_sockaddr_in_init(&stun_srv.ipv4, NULL, 0);
- stun_srv.ipv4.sin_addr = *(pj_in_addr*) he.h_addr;
- stun_srv.ipv4.sin_port = pj_htons((pj_uint16_t) 3478);
- }
- }
-
- return stun_status;
-}
-
-int UserAgent::createUDPServer() {
- pj_status_t status;
- //pj_str_t ipAddr;
- pj_sockaddr_in bound_addr;
- pjsip_host_port a_name;
- char tmpIP[32];
-
- // Init bound address to ANY
- pj_memset(&bound_addr, 0, sizeof (bound_addr));
- bound_addr.sin_addr.s_addr = PJ_INADDR_ANY;
-
- // Create UDP server socket
- status = pj_sock_socket(PJ_AF_INET, PJ_SOCK_DGRAM, 0, &_sock);
- if (status != PJ_SUCCESS) {
- _debug("UserAgent: (%d) UDP socket() error\n", status);
- return status;
- }
-
- status = pj_sock_bind_in(_sock, pj_ntohl(bound_addr.sin_addr.s_addr), (pj_uint16_t) _localPort);
- if (status != PJ_SUCCESS) {
- _debug("UserAgent: (%d) UDP bind() error\n", status);
- pj_sock_close(_sock);
- return status;
- }
-
- _debug("UserAgent: Use IP: %s\n", _localExternAddress.data());
-
- // Create UDP-Server (default port: 5060)
- strcpy(tmpIP, _localExternAddress.data());
- pj_strdup2(_pool, &a_name.host, tmpIP);
- a_name.port = (pj_uint16_t) _localExternPort;
-
- status = pjsip_udp_transport_attach(_endpt, _sock, &a_name, 1, NULL);
- if (status != PJ_SUCCESS) {
- _debug("UserAgent: (%d) Unable to start UDP transport!\n", status);
- return -1;
- } else {
- _debug("UserAgent: UDP server listening on port %d\n", _localExternPort);
- }
-
- return 0;
-}
-
-void UserAgent::setStunServer(const char *server) {
- if(server != NULL) {
- _useStun = true;
- _stunServer = std::string(server);
- } else {
- _useStun = false;
- _stunServer = std::string("");
- }
-}
-
-void UserAgent::regc_cb(struct pjsip_regc_cbparam *param) {
-
- AccountID *id = static_cast<AccountID *> (param->token);
- SIPVoIPLink *voipLink;
-
- _debug("UserAgent: Account ID is %s, Register result: %d, Status: %d\n", id->data(), param->status, param->code);
- voipLink = dynamic_cast<SIPVoIPLink *>(Manager::instance().getAccountLink(*id));
- if(!voipLink)
- return;
-
- if (param->status == PJ_SUCCESS) {
- if (param->code < 0 || param->code >= 300) {
- /* Sometimes, the status is OK, but we still failed.
- * So checking the code for real result
- */
- _debug("UserAgent: The error is: %d\n", param->code);
- switch(param->code) {
- case 408:
- case 606:
- Manager::instance().getAccountLink(*id)->setRegistrationState(VoIPLink::ErrorConfStun);
- break;
- case 503:
- Manager::instance().getAccountLink(*id)->setRegistrationState(VoIPLink::ErrorHost);
- break;
- case 401:
- case 403:
- case 404:
- Manager::instance().getAccountLink(*id)->setRegistrationState(VoIPLink::ErrorAuth);
- break;
- default:
- Manager::instance().getAccountLink(*id)->setRegistrationState(VoIPLink::Error);
- break;
- }
- voipLink->setRegister(false);
- } else {
- // Registration/Unregistration is success
-
- if(voipLink->isRegister())
- Manager::instance().getAccountLink(*id)->setRegistrationState(VoIPLink::Registered);
- else {
- Manager::instance().getAccountLink(*id)->setRegistrationState(VoIPLink::Unregistered);
- voipLink->setRegister(false);
- }
- }
- } else {
- Manager::instance().getAccountLink(*id)->setRegistrationState(VoIPLink::ErrorAuth);
- voipLink->setRegister(false);
- }
-}
-
-bool
-UserAgent::loadSIPLocalIP() {
- bool returnValue = true;
- if (_localIPAddress == "127.0.0.1") {
- pj_sockaddr ip_addr;
- if (pj_gethostip(pj_AF_INET(), &ip_addr) != PJ_SUCCESS) {
- // Update the registration state if no network capabilities found
- _debug("UserAgent: Get host ip failed!\n");
- returnValue = false;
- } else {
- _localIPAddress = std::string(pj_inet_ntoa(ip_addr.ipv4.sin_addr));
- _debug("UserAgent: Checking network, setting local IP address to: %s\n", _localIPAddress.data());
- }
- }
- return returnValue;
-}
-
-/* Thread entry point function. */
-int UserAgent::start_thread(void *arg) {
-
- PJ_UNUSED_ARG(arg);
-
- // FIXME! maybe we should add a flag for exiting!
- // TODO Add the flag. We have to stop the thread when destroying the instance
- while (!Manager::instance().getSipThreadStatus()) {
- pj_time_val timeout = {0, 10};
- pjsip_endpt_handle_events(getInstance()->getEndPoint(), &timeout);
- }
-
- return 0;
-}
-
-void UserAgent::set_voicemail_info( AccountID account, pjsip_msg_body *body ){
-
- int voicemail, pos_begin, pos_end;
- std::string voice_str = "Voice-Message: ";
- std::string delimiter = "/";
- std::string msg_body, voicemail_str;
-
- _debug("UserAgent: checking the voice message!\n");
- // The voicemail message is formated like that:
- // Voice-Message: 1/0 . 1 is the number we want to retrieve in this case
-
- // We get the notification body
- msg_body = (char*)body->data;
-
- // We need the position of the first character of the string voice_str
- pos_begin = msg_body.find(voice_str);
- // We need the position of the delimiter
- pos_end = msg_body.find(delimiter);
-
- // So our voicemail number between the both index
- try {
-
- voicemail_str = msg_body.substr(pos_begin + voice_str.length(), pos_end - ( pos_begin + voice_str.length()));
- std::cout << "voicemail number : " << voicemail_str << std::endl;
- voicemail = atoi( voicemail_str.c_str() );
- }
- catch( std::out_of_range& e ){
- std::cerr << e.what() << std::endl;
- }
-
- // We need now to notify the manager
- if( voicemail != 0 )
- Manager::instance().startVoiceMessageNotification(account, voicemail);
-}
-
-
-pj_bool_t UserAgent::mod_on_rx_request(pjsip_rx_data *rdata) {
-
- pj_status_t status;
- pj_str_t reason;
- unsigned options = 0;
- pjsip_dialog* dialog;
- pjsip_tx_data *tdata;
- //pjmedia_sdp_session *r_sdp;
- AccountID account_id;
-
- // voicemail part
- std::string method_name;
- std::string request;
-
- // Handle the incoming call invite in this function
- _debug("UserAgent: Callback on_rx_request is involved!\n");
-
- /* First, let's got the username and server name from the invite.
- * We will use them to detect which account is the callee.
- */
- pjsip_uri *uri = rdata->msg_info.to->uri;
- pjsip_sip_uri *sip_uri = (pjsip_sip_uri *) pjsip_uri_get_uri(uri);
-
- std::string userName = std::string(sip_uri->user.ptr, sip_uri->user.slen);
- std::string server = std::string(sip_uri->host.ptr, sip_uri->host.slen);
-
- // Get the account id of callee from username and server
- account_id = Manager::instance().getAccountIdFromNameAndServer(userName, server);
- if(account_id == AccountNULL) {
- _debug("UserAgent: Username %s doesn't match any account!\n",userName.c_str());
- return PJ_FALSE;
- }
- _debug("UserAgent: The receiver is : %s@%s\n", userName.data(), server.data());
- _debug("UserAgent: The callee account id is %s\n", account_id.c_str());
-
- /* Now, it is the time to find the information of the caller */
- uri = rdata->msg_info.from->uri;
- sip_uri = (pjsip_sip_uri *) pjsip_uri_get_uri(uri);
-
- std::string caller = std::string(sip_uri->user.ptr, sip_uri->user.slen);
- std::string callerServer = std::string(sip_uri->host.ptr, sip_uri->host.slen);
- std::string peerNumber = caller + "@" + callerServer;
-
-
- // Get the server voicemail notification
- // Catch the NOTIFY message
- if( rdata->msg_info.msg->line.req.method.id == PJSIP_OTHER_METHOD )
- {
- method_name = "NOTIFY";
- // Retrieve all the message. Should contains only the method name but ...
- request = rdata->msg_info.msg->line.req.method.name.ptr;
- // Check if the message is a notification
- if( request.find( method_name ) != (size_t)-1 ) {
- set_voicemail_info( account_id, rdata->msg_info.msg->body );
- }
- pjsip_endpt_respond_stateless(getInstance()->getEndPoint(), rdata, PJSIP_SC_OK, NULL, NULL, NULL);
- return PJ_SUCCESS;
- }
-
- // Respond statelessly any non-INVITE requests with 500
- if (rdata->msg_info.msg->line.req.method.id != PJSIP_INVITE_METHOD) {
- if (rdata->msg_info.msg->line.req.method.id != PJSIP_ACK_METHOD) {
- pj_strdup2(getInstance()->getAppPool(), &reason, "user agent unable to handle this request ");
- pjsip_endpt_respond_stateless(getInstance()->getEndPoint(), rdata, PJSIP_SC_METHOD_NOT_ALLOWED, &reason, NULL,
- NULL);
- return PJ_TRUE;
- }
- }
-
- // Verify that we can handle the request
- status = pjsip_inv_verify_request(rdata, &options, NULL, NULL, getInstance()->getEndPoint(), NULL);
- if (status != PJ_SUCCESS) {
- pj_strdup2(getInstance()->getAppPool(), &reason, "user agent unable to handle this INVITE ");
- pjsip_endpt_respond_stateless(getInstance()->getEndPoint(), rdata, PJSIP_SC_METHOD_NOT_ALLOWED, &reason, NULL,
- NULL);
- return PJ_TRUE;
- }
-
- // Generate a new call ID for the incoming call!
- CallID id = Manager::instance().getNewCallID();
-
- _debug("UserAgent: The call id of the incoming call is %s\n", id.c_str());
- SIPCall* call = new SIPCall(id, Call::Incoming);
- if (!call) {
- _debug("UserAgent: unable to create an incoming call");
- return PJ_FALSE;
- }
-
- // Set the codec map, IP, peer number and so on... for the SIPCall object
- getInstance()->setCallAudioLocal(call);
- call->setCodecMap(Manager::instance().getCodecDescriptorMap());
- call->setConnectionState(Call::Progressing);
- call->setIp(getInstance()->getLocalIP());
- call->setPeerNumber(peerNumber);
-
- /* Call the SIPCallInvite function to generate the local sdp,
- * remote sdp and negociator.
- * This function is also used to set the parameters of audio RTP, including:
- * local IP and port number
- * remote IP and port number
- * possilbe audio codec will be used in this call
- */
- if (call->SIPCallInvite(rdata, getInstance()->getAppPool())) {
-
- // Notify UI there is an incoming call
- if (Manager::instance().incomingCall(call, account_id)) {
- // Add this call to the callAccountMap in ManagerImpl
- Manager::instance().getAccountLink(account_id)->addCall(call);
- _debug("UserAgent: Notify UI success!\n");
- } else {
- // Fail to notify UI
- delete call;
- call = NULL;
- _debug("UserAgent: Fail to notify UI!\n");
- return PJ_FALSE;
- }
- } else {
- // Fail to collect call information
- delete call;
- call = NULL;
- _debug("UserAgent: Call SIPCallInvite failed!\n");
- return PJ_FALSE;
- }
-
- /* Create the local dialog (UAS) */
- status = pjsip_dlg_create_uas(pjsip_ua_instance(), rdata, NULL, &dialog);
- if (status != PJ_SUCCESS) {
- pjsip_endpt_respond_stateless(getInstance()->getEndPoint(), rdata, PJSIP_SC_INTERNAL_SERVER_ERROR, &reason, NULL,
- NULL);
- return PJ_TRUE;
- }
-
- // Specify media capability during invite session creation
- pjsip_inv_session *inv;
- status = pjsip_inv_create_uas(dialog, rdata, call->getLocalSDPSession(), 0, &inv);
- PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
-
- // Associate the call in the invite session
- inv->mod_data[getInstance()->getModId()] = call;
-
- // Send a 180/Ringing response
- status = pjsip_inv_initial_answer(inv, rdata, PJSIP_SC_RINGING, NULL, NULL, &tdata);
- PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
- status = pjsip_inv_send_msg(inv, tdata);
- PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
-
- // Associate invite session to the current call
- call->setInvSession(inv);
-
- // Update the connection state
- call->setConnectionState(Call::Ringing);
-
- /* Done */
- return PJ_SUCCESS;
-}
-
-bool UserAgent::setCallAudioLocal(SIPCall* call) {
- // Firstly, we use the local IP and port number
- unsigned int callLocalAudioPort = RANDOM_LOCAL_PORT;
- unsigned int callLocalExternAudioPort = callLocalAudioPort;
-
- if (_useStun) {
- // If use Stun server, modify them
- if (Manager::instance().behindNat(_stunServer, callLocalAudioPort)) {
- callLocalExternAudioPort = Manager::instance().getFirewallPort();
- }
- }
- _debug("UserAgent: Setting local audio port to: %d\n", callLocalAudioPort);
- _debug("UserAgent: Setting local audio port (external) to: %d\n", callLocalExternAudioPort);
-
- // Set local audio port for SIPCall(id)
- call->setLocalIp(_localIPAddress);
- call->setLocalAudioPort(callLocalAudioPort);
- call->setLocalExternAudioPort(callLocalExternAudioPort);
-
- return true;
-}
-
-int UserAgent::answer(SIPCall *call) {
- pj_status_t status;
- pjsip_tx_data *tdata;
-
- // User answered the incoming call, tell peer this news
- if (call->startNegociation(_pool)) {
- // Create and send a 200(OK) response
- _debug("UserAgent: Negociation success!\n");
- status = pjsip_inv_answer(call->getInvSession(), PJSIP_SC_OK, NULL, NULL, &tdata);
- PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
- status = pjsip_inv_send_msg(call->getInvSession(), tdata);
- PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
-
- return 0;
- }
-
- return 1;
-}
-
-bool UserAgent::makeOutgoingCall(const std::string& strTo, SIPCall* call, const AccountID& id) {
- pj_status_t status;
- pjsip_dialog *dialog;
- pjsip_tx_data *tdata;
- pj_str_t from, to, contact;
-
- _debug("*******************AccountId is %s\n", id.data());
- // Get the basic information about the callee account
- SIPAccount* account = dynamic_cast<SIPAccount *>(Manager::instance().getAccount(id));
-
- // Generate the from URI
- std::string strFrom = "sip:" + account->getUserName() + "@" + account->getServer();
-
- _debug("UserAgent: Make a new call from:%s to %s. Contact is %s\n",
- strFrom.data(), strTo.data(), account->getContact().data());
-
- // pjsip need the from and to information in pj_str_t format
- pj_strdup2(_pool, &from, strFrom.data());
- pj_strdup2(_pool, &to, strTo.data());
- pj_strdup2(_pool, &contact, account->getContact().data());
-
- // create the dialog (UAC)
- status = pjsip_dlg_create_uac(pjsip_ua_instance(), &from,
- &contact,
- &to,
- NULL,
- &dialog);
- PJ_ASSERT_RETURN(status == PJ_SUCCESS, false);
-
- setCallAudioLocal(call);
- call->setIp(getInstance()->getLocalIP());
-
- // Building the local SDP offer
- call->createInitialOffer(_pool);
-
- // Create the invite session for this call
- pjsip_inv_session *inv;
- status = pjsip_inv_create_uac(dialog, call->getLocalSDPSession(), 0, &inv);
- PJ_ASSERT_RETURN(status == PJ_SUCCESS, false);
-
- // Set auth information
- pjsip_auth_clt_set_credentials(&dialog->auth_sess, 1, account->getCredInfo());
-
- // Associate current call in the invite session
- inv->mod_data[_mod.id] = call;
-
- status = pjsip_inv_invite(inv, &tdata);
- PJ_ASSERT_RETURN(status == PJ_SUCCESS, false);
-
- // Associate current invite session in the call
- call->setInvSession(inv);
-
- status = pjsip_inv_send_msg(inv, tdata);
- //PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
- if(status != PJ_SUCCESS) {
- return false;
- }
-
- return true;
-}
-
-void UserAgent::call_on_forked(pjsip_inv_session *inv, pjsip_event *e) {
- PJ_UNUSED_ARG(inv);
- PJ_UNUSED_ARG(e);
-}
-
-void UserAgent::call_on_tsx_changed(pjsip_inv_session *inv, pjsip_transaction *tsx, pjsip_event *e) {
-
- pjsip_rx_data *rdata;
- AccountID accId;
- SIPCall *call;
- SIPVoIPLink *link;
- pjsip_msg *msg;
-
- _debug("UserAgent: TSX Changed! The tsx->state is %d; tsx->role is %d; code is %d; method id is %.*s.\n",
- tsx->state, tsx->role, tsx->status_code, (int)tsx->method.name.slen, tsx->method.name.ptr);
-
- if(pj_strcmp2(&tsx->method.name, "INFO") == 0) {
- // Receive a INFO message, ingore it!
- return;
- }
-
- //Retrieve the body message
- rdata = e->body.tsx_state.src.rdata;
-
- if (tsx->role == PJSIP_ROLE_UAC) {
- switch (tsx->state) {
- case PJSIP_TSX_STATE_TERMINATED:
- if (tsx->status_code == 200 &&
- pjsip_method_cmp(&tsx->method, pjsip_get_refer_method()) != 0) {
- // Peer answered the outgoing call
- _debug("UserAgent: Peer answered the outgoing call!\n");
- call = reinterpret_cast<SIPCall *> (inv->mod_data[getInstance()->getModId()]);
- if (call == NULL)
- return;
-
- //_debug("UserAgent: The call id is %s\n", call->getCallId().data());
-
- accId = Manager::instance().getAccountFromCall(call->getCallId());
- link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink(accId));
- if (link)
- link->SIPCallAnswered(call, rdata);
- } else if (tsx->status_code / 100 == 5) {
- _debug("UserAgent: 5xx error message received\n");
- }
- break;
- case PJSIP_TSX_STATE_PROCEEDING:
- // Peer is ringing for the outgoing call
- msg = rdata->msg_info.msg;
-
- call = reinterpret_cast<SIPCall *> (inv->mod_data[getInstance()->getModId()]);
- if (call == NULL)
- return;
-
- if (msg->line.status.code == 180) {
- _debug("UserAgent: Peer is ringing!\n");
-
- call->setConnectionState(Call::Ringing);
- Manager::instance().peerRingingCall(call->getCallId());
- }
- break;
- case PJSIP_TSX_STATE_COMPLETED:
- if (tsx->status_code == 407 || tsx->status_code == 401) //FIXME
- break;
- if (tsx->status_code / 100 == 6 || tsx->status_code / 100 == 4) {
- // We get error message of outgoing call from server
- _debug("UserAgent: Server error message is received!\n");
- call = reinterpret_cast<SIPCall *> (inv->mod_data[getInstance()->getModId()]);
- if (call == NULL) {
- _debug("UserAgent: Call has been removed!\n");
- return;
- }
- accId = Manager::instance().getAccountFromCall(call->getCallId());
- link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink(accId));
- if (link) {
- link->SIPCallServerFailure(call);
- }
- }
- break;
- default:
- break;
- } // end of switch
-
- } else {
- switch (tsx->state) {
- case PJSIP_TSX_STATE_TRYING:
- if (pjsip_method_cmp(&tsx->method, pjsip_get_refer_method()) == 0) {
- // Peer ask me to transfer call to another number.
- _debug("UserAgent: Incoming REFER request!\n");
- getInstance()->onCallTransfered(inv, e->body.tsx_state.src.rdata);
- }
- break;
- case PJSIP_TSX_STATE_COMPLETED:
- if (tsx->status_code == 200 && tsx->method.id == PJSIP_BYE_METHOD) {
- // Peer hangup the call
- _debug("UserAgent: Peer hangup(bye) message is received!\n");
- call = reinterpret_cast<SIPCall *> (inv->mod_data[getInstance()->getModId()]);
- if (call == NULL) {
- _debug("UserAgent: Call has been removed!\n");
- return;
- }
- accId = Manager::instance().getAccountFromCall(call->getCallId());
- link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink(accId));
- if (link) {
- link->SIPCallClosed(call);
- }
- } else if (tsx->status_code == 200 && tsx->method.id == PJSIP_CANCEL_METHOD) {
- // Peer refuse the call
- _debug("UserAgent: Cancel message is received!\n");
- call = reinterpret_cast<SIPCall *> (inv->mod_data[getInstance()->getModId()]);
- if (call == NULL) {
- _debug("UserAgent: Call has been removed!\n");
- return;
- }
-
- accId = Manager::instance().getAccountFromCall(call->getCallId());
- link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink(accId));
- if (link) {
- link->SIPCallClosed(call);
- }
- }
- break;
- default:
- break;
- } // end of switch
- }
-
-}
-
-void UserAgent::call_on_state_changed(pjsip_inv_session *inv, pjsip_event *e) {
-
- PJ_UNUSED_ARG(inv);
-
- SIPCall *call = reinterpret_cast<SIPCall*> (inv->mod_data[getInstance()->getModId()]);
- if(!call)
- return;
-
- /* If this is an outgoing INVITE that was created because of
- * REFER/transfer, send NOTIFY to transferer.
- */
- if (call->getXferSub() && e->type==PJSIP_EVENT_TSX_STATE) {
- int st_code = -1;
- pjsip_evsub_state ev_state = PJSIP_EVSUB_STATE_ACTIVE;
-
-
- switch (call->getInvSession()->state) {
- case PJSIP_INV_STATE_NULL:
- case PJSIP_INV_STATE_CALLING:
- /* Do nothing */
- break;
-
- case PJSIP_INV_STATE_EARLY:
- case PJSIP_INV_STATE_CONNECTING:
- st_code = e->body.tsx_state.tsx->status_code;
- ev_state = PJSIP_EVSUB_STATE_ACTIVE;
- break;
-
- case PJSIP_INV_STATE_CONFIRMED:
- /* When state is confirmed, send the final 200/OK and terminate
- * subscription.
- */
- st_code = e->body.tsx_state.tsx->status_code;
- ev_state = PJSIP_EVSUB_STATE_TERMINATED;
- break;
-
- case PJSIP_INV_STATE_DISCONNECTED:
- st_code = e->body.tsx_state.tsx->status_code;
- ev_state = PJSIP_EVSUB_STATE_TERMINATED;
- break;
-
- case PJSIP_INV_STATE_INCOMING:
- /* Nothing to do. Just to keep gcc from complaining about
- * unused enums.
- */
- break;
- }
-
- if (st_code != -1) {
- pjsip_tx_data *tdata;
- pj_status_t status;
-
- status = pjsip_xfer_notify( call->getXferSub(),
- ev_state, st_code,
- NULL, &tdata);
- if (status != PJ_SUCCESS) {
- _debug("UserAgent: Unable to create NOTIFY -- %d\n", status);
- } else {
- status = pjsip_xfer_send_request(call->getXferSub(), tdata);
- if (status != PJ_SUCCESS) {
- _debug("UserAgent: Unable to send NOTIFY -- %d\n", status);
- }
- }
- }
- }
-
-}
-
-bool UserAgent::onhold(SIPCall *call) {
-
- pj_status_t status;
- pjsip_tx_data *tdata;
- pjmedia_sdp_attr *attr;
- pjmedia_sdp_session* local_sdp;
-
- local_sdp = call->getLocalSDPSession();
- if( local_sdp == NULL ){
- _debug("! SIP Failure: unable to find local_sdp\n");
- return false;
- }
-
- /* Create re-INVITE with new offer */
- // Remove all the attributes with the specified name
- pjmedia_sdp_media_remove_all_attr(local_sdp->media[0], "sendrecv");
- attr = pjmedia_sdp_attr_create(_pool, "sendonly", NULL);
- pjmedia_sdp_media_add_attr(local_sdp->media[0], attr);
-
- status = pjsip_inv_reinvite( call->getInvSession(), NULL, local_sdp, &tdata);
- if( status != PJ_SUCCESS )
- {
- _debug("On hold: creation of the Re-invite request failed\n");
- return false;
- }
- /* Send the request */
- status = pjsip_inv_send_msg( call->getInvSession(), tdata);
-
- return (status == PJ_SUCCESS);
-}
-
-bool UserAgent::offhold(SIPCall *call) {
-
- pj_status_t status;
- pjsip_tx_data *tdata;
- pjmedia_sdp_attr *attr;
- pjmedia_sdp_session* local_sdp;
-
- local_sdp = call->getLocalSDPSession();
- if( local_sdp == NULL ){
- _debug("! SIP Failure: unable to find local_sdp\n");
- return false;
- }
-
- /* Create re-INVITE with new offer */
- // Remove all the attributes with the specified name
- pjmedia_sdp_media_remove_all_attr(local_sdp->media[0], "sendonly");
- attr = pjmedia_sdp_attr_create(_pool, "sendrecv", NULL);
- pjmedia_sdp_media_add_attr(local_sdp->media[0], attr);
-
- status = pjsip_inv_reinvite( call->getInvSession(), NULL, local_sdp , &tdata);
- if( status != PJ_SUCCESS )
- {
- _debug("Off hold: creation of the Re-invite request failed\n");
- return false;
- }
-
- /* Send the request */
- status = pjsip_inv_send_msg( call->getInvSession(), tdata);
-
- return (status == PJ_SUCCESS);
-}
-
-bool UserAgent::hangup(SIPCall* call) {
- pj_status_t status;
- pjsip_tx_data *tdata = NULL;
-
- // User hangup current call. Notify peer
- status = pjsip_inv_end_session(call->getInvSession(), 404, NULL, &tdata);
- if(status != PJ_SUCCESS)
- return false;
-
- _debug("UserAgent: Before send msg!\n");
-
- if(tdata == NULL)
- return true;
-
- status = pjsip_inv_send_msg(call->getInvSession(), tdata);
- if(status != PJ_SUCCESS)
- return false;
-
- call->getInvSession()->mod_data[getInstance()->getModId()] = NULL;
- return true;
-}
-
-bool UserAgent::refuse(SIPCall* call)
-{
- pj_status_t status;
- pjsip_tx_data *tdata;
-
- // User refuse current call. Notify peer
- status = pjsip_inv_end_session(call->getInvSession(), PJSIP_SC_DECLINE, NULL, &tdata); //603
- if(status != PJ_SUCCESS)
- return false;
-
- status = pjsip_inv_send_msg(call->getInvSession(), tdata);
- if(status != PJ_SUCCESS)
- return false;
-
- call->getInvSession()->mod_data[getInstance()->getModId()] = NULL;
- return true;
-}
-
-
-bool UserAgent::carryingDTMFdigits(SIPCall* call, char *msgBody)
-{
- pj_status_t status;
- pjsip_tx_data *tdata;
- pj_str_t methodName, content;
- pjsip_method method;
- pjsip_media_type ctype;
-
- pj_strdup2(_pool, &methodName, "INFO");
- pjsip_method_init_np(&method, &methodName);
-
- /* Create request message. */
- status = pjsip_dlg_create_request( call->getInvSession()->dlg, &method,
- -1, &tdata);
- if (status != PJ_SUCCESS) {
- _debug("UserAgent: Unable to create INFO request -- %d\n", status);
- return false;
- }
-
- /* Get MIME type */
- pj_strdup2(_pool, &ctype.type, "application");
- pj_strdup2(_pool, &ctype.subtype, "dtmf-relay");
-
- /* Create "application/dtmf-relay" message body. */
- pj_strdup2(_pool, &content, msgBody);
- tdata->msg->body = pjsip_msg_body_create( tdata->pool, &ctype.type,
- &ctype.subtype, &content);
- if (tdata->msg->body == NULL) {
- _debug("UserAgent: Unable to create msg body!\n");
- pjsip_tx_data_dec_ref(tdata);
- return false;
- }
-
- /* Send the request. */
- status = pjsip_dlg_send_request( call->getInvSession()->dlg, tdata,
- _mod.id, NULL);
- if (status != PJ_SUCCESS) {
- _debug("UserAgent: Unable to send MESSAGE request -- %d\n", status);
- return false;
- }
-
- return true;
-
-}
-
-bool UserAgent::transfer(SIPCall *call, const std::string& to)
-{
- pjsip_evsub *sub;
- pjsip_tx_data *tdata;
- struct pjsip_evsub_user xfer_cb;
- pj_status_t status;
- pj_str_t dest;
-
- pj_strdup2(_pool, &dest, to.data());
-
- /* Create xfer client subscription. */
- pj_bzero(&xfer_cb, sizeof(xfer_cb));
- xfer_cb.on_evsub_state = &xfer_func_cb;
-
- status = pjsip_xfer_create_uac(call->getInvSession()->dlg, &xfer_cb, &sub);
- if (status != PJ_SUCCESS) {
- _debug("UserAgent: Unable to create xfer -- %d\n", status);
- return false;
- }
-
- /* Associate this voiplink of call with the client subscription
- * We can not just associate call with the client subscription
- * because after this function, we can not find the cooresponding
- * voiplink from the call any more. But the voiplink is useful!
- */
- AccountID accId = Manager::instance().getAccountFromCall(call->getCallId());
- SIPVoIPLink *link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink(accId));
- pjsip_evsub_set_mod_data(sub, _mod.id, link);
-
- /*
- * Create REFER request.
- */
- status = pjsip_xfer_initiate(sub, &dest, &tdata);
- if (status != PJ_SUCCESS) {
- _debug("UserAgent: Unable to create REFER request -- %d\n", status);
- return false;
- }
-
- /* Send. */
- status = pjsip_xfer_send_request(sub, tdata);
- if (status != PJ_SUCCESS) {
- _debug("UserAgent: Unable to send REFER request -- %d\n", status);
- return false;
- }
-
- return true;
-}
-
-void UserAgent::xfer_func_cb( pjsip_evsub *sub, pjsip_event *event)
-{
- PJ_UNUSED_ARG(event);
-
- _debug("UserAgent: Transfer callback is involved!\n");
- /*
- * When subscription is accepted (got 200/OK to REFER), check if
- * subscription suppressed.
- */
- if (pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_ACCEPTED) {
-
- pjsip_rx_data *rdata;
- pjsip_generic_string_hdr *refer_sub;
- const pj_str_t REFER_SUB = {(char*)"Refer-Sub", 9 };
-
- SIPVoIPLink *link = reinterpret_cast<SIPVoIPLink *> (pjsip_evsub_get_mod_data(sub,
- getInstance()->getModId()));
-
- /* Must be receipt of response message */
- pj_assert(event->type == PJSIP_EVENT_TSX_STATE &&
- event->body.tsx_state.type == PJSIP_EVENT_RX_MSG);
- rdata = event->body.tsx_state.src.rdata;
-
- /* Find Refer-Sub header */
- refer_sub = (pjsip_generic_string_hdr*)
- pjsip_msg_find_hdr_by_name(rdata->msg_info.msg,
- &REFER_SUB, NULL);
-
- /* Check if subscription is suppressed */
- if (refer_sub && pj_stricmp2(&refer_sub->hvalue, "false")==0) {
- /* Since no subscription is desired, assume that call has been
- * transfered successfully.
- */
- if (link) {
- // It's the time to stop the RTP
- link->transferStep2();
- }
-
- /* Yes, subscription is suppressed.
- * Terminate our subscription now.
- */
- _debug("UserAgent: Xfer subscription suppressed, terminating event subcription...\n");
- pjsip_evsub_terminate(sub, PJ_TRUE);
-
- } else {
- /* Notify application about call transfer progress.
- * Initially notify with 100/Accepted status.
- */
- _debug("UserAgent: Xfer subscription 100/Accepted received...\n");
- }
- }
- /*
- * On incoming NOTIFY, notify application about call transfer progress.
- */
- else if (pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_ACTIVE ||
- pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_TERMINATED)
- {
- pjsip_msg *msg;
- pjsip_msg_body *body;
- pjsip_status_line status_line;
- pj_bool_t is_last;
- pj_bool_t cont;
- pj_status_t status;
-
- SIPVoIPLink *link = reinterpret_cast<SIPVoIPLink *> (pjsip_evsub_get_mod_data(sub,
- getInstance()->getModId()));
-
- /* When subscription is terminated, clear the xfer_sub member of
- * the inv_data.
- */
- if (pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_TERMINATED) {
- pjsip_evsub_set_mod_data(sub, getInstance()->getModId(), NULL);
- _debug("UserAgent: Xfer client subscription terminated\n");
-
- }
-
- if (!link || !event) {
- /* Application is not interested with call progress status */
- _debug("UserAgent: Either link or event is empty!\n");
- return;
- }
-
- // Get current call
- SIPCall *call = dynamic_cast<SIPCall *>(link->getCall(Manager::instance().getCurrentCallId()));
- if(!call) {
- _debug("UserAgent: Call doesn't exit!\n");
- return;
- }
-
- /* This better be a NOTIFY request */
- if (event->type == PJSIP_EVENT_TSX_STATE &&
- event->body.tsx_state.type == PJSIP_EVENT_RX_MSG)
- {
- pjsip_rx_data *rdata;
-
- rdata = event->body.tsx_state.src.rdata;
-
- /* Check if there's body */
- msg = rdata->msg_info.msg;
- body = msg->body;
- if (!body) {
- _debug("UserAgent: Warning! Received NOTIFY without message body\n");
- return;
- }
-
- /* Check for appropriate content */
- if (pj_stricmp2(&body->content_type.type, "message") != 0 ||
- pj_stricmp2(&body->content_type.subtype, "sipfrag") != 0)
- {
- _debug("UserAgent: Warning! Received NOTIFY with non message/sipfrag content\n");
- return;
- }
-
- /* Try to parse the content */
- status = pjsip_parse_status_line((char*)body->data, body->len,
- &status_line);
- if (status != PJ_SUCCESS) {
- _debug("UserAgent: Warning! Received NOTIFY with invalid message/sipfrag content\n");
- return;
- }
-
- } else {
- _debug("UserAgent: Set code to 500!\n");
- status_line.code = 500;
- status_line.reason = *pjsip_get_status_text(500);
- }
-
- /* Notify application */
- is_last = (pjsip_evsub_get_state(sub)==PJSIP_EVSUB_STATE_TERMINATED);
- cont = !is_last;
-
- if(status_line.code/100 == 2) {
- _debug("UserAgent: Try to stop rtp!\n");
- pjsip_tx_data *tdata;
-
- status = pjsip_inv_end_session(call->getInvSession(), PJSIP_SC_GONE, NULL, &tdata);
- if(status != PJ_SUCCESS) {
- _debug("UserAgent: Fail to create end session msg!\n");
- } else {
- status = pjsip_inv_send_msg(call->getInvSession(), tdata);
- if(status != PJ_SUCCESS)
- _debug("UserAgent: Fail to send end session msg!\n");
- }
-
- link->transferStep2();
- cont = PJ_FALSE;
- }
-
- if (!cont) {
- pjsip_evsub_set_mod_data(sub, getInstance()->getModId(), NULL);
- }
- }
-
-}
-
-void UserAgent::onCallTransfered(pjsip_inv_session *inv, pjsip_rx_data *rdata)
-{
- pj_status_t status;
- pjsip_tx_data *tdata;
- SIPCall *existing_call;
- const pj_str_t str_refer_to = { (char*)"Refer-To", 8};
- const pj_str_t str_refer_sub = { (char*)"Refer-Sub", 9 };
- const pj_str_t str_ref_by = { (char*)"Referred-By", 11 };
- pjsip_generic_string_hdr *refer_to;
- pjsip_generic_string_hdr *refer_sub;
- pjsip_hdr *ref_by_hdr;
- pj_bool_t no_refer_sub = PJ_FALSE;
- char *uri;
- std::string tmp;
- pjsip_status_code code;
- pjsip_evsub *sub;
-
- existing_call = (SIPCall *) inv->mod_data[getInstance()->getModId()];
-
- /* Find the Refer-To header */
- refer_to = (pjsip_generic_string_hdr*)
- pjsip_msg_find_hdr_by_name(rdata->msg_info.msg, &str_refer_to, NULL);
-
- if (refer_to == NULL) {
- /* Invalid Request.
- * No Refer-To header!
- */
- _debug("UserAgent: Received REFER without Refer-To header!\n");
- pjsip_dlg_respond( inv->dlg, rdata, 400, NULL, NULL, NULL);
- return;
- }
-
- /* Find optional Refer-Sub header */
- refer_sub = (pjsip_generic_string_hdr*)
- pjsip_msg_find_hdr_by_name(rdata->msg_info.msg, &str_refer_sub, NULL);
-
- if (refer_sub) {
- if (!pj_strnicmp2(&refer_sub->hvalue, "true", 4)==0)
- no_refer_sub = PJ_TRUE;
- }
-
- /* Find optional Referred-By header (to be copied onto outgoing INVITE
- * request.
- */
- ref_by_hdr = (pjsip_hdr*)
- pjsip_msg_find_hdr_by_name(rdata->msg_info.msg, &str_ref_by,
- NULL);
-
- /* Notify callback */
- code = PJSIP_SC_ACCEPTED;
-
- _debug("UserAgent: Call to %.*s is being transfered to %.*s\n",
- (int)inv->dlg->remote.info_str.slen,
- inv->dlg->remote.info_str.ptr,
- (int)refer_to->hvalue.slen,
- refer_to->hvalue.ptr);
-
- if (no_refer_sub) {
- /*
- * Always answer with 2xx.
- */
- pjsip_tx_data *tdata;
- const pj_str_t str_false = { (char*)"false", 5};
- pjsip_hdr *hdr;
-
- status = pjsip_dlg_create_response(inv->dlg, rdata, code, NULL,
- &tdata);
- if (status != PJ_SUCCESS) {
- _debug("UserAgent: Unable to create 2xx response to REFER -- %d\n", status);
- return;
- }
-
- /* Add Refer-Sub header */
- hdr = (pjsip_hdr*)
- pjsip_generic_string_hdr_create(tdata->pool, &str_refer_sub,
- &str_false);
- pjsip_msg_add_hdr(tdata->msg, hdr);
-
-
- /* Send answer */
- status = pjsip_dlg_send_response(inv->dlg, pjsip_rdata_get_tsx(rdata),
- tdata);
- if (status != PJ_SUCCESS) {
- _debug("UserAgent: Unable to create 2xx response to REFER -- %d\n", status);
- return;
- }
-
- /* Don't have subscription */
- sub = NULL;
-
- } else {
- struct pjsip_evsub_user xfer_cb;
- pjsip_hdr hdr_list;
-
- /* Init callback */
- pj_bzero(&xfer_cb, sizeof(xfer_cb));
- xfer_cb.on_evsub_state = &xfer_svr_cb;
-
- /* Init addiTHIS_FILE, THIS_FILE, tional header list to be sent with REFER response */
- pj_list_init(&hdr_list);
-
- /* Create transferee event subscription */
- status = pjsip_xfer_create_uas( inv->dlg, &xfer_cb, rdata, &sub);
- if (status != PJ_SUCCESS) {
- _debug("UserAgent: Unable to create xfer uas -- %d\n", status);
- pjsip_dlg_respond( inv->dlg, rdata, 500, NULL, NULL, NULL);
- return;
- }
-
- /* If there's Refer-Sub header and the value is "true", send back
- * Refer-Sub in the response with value "true" too.
- */
- if (refer_sub) {
- const pj_str_t str_true = { (char*)"true", 4 };
- pjsip_hdr *hdr;
-
- hdr = (pjsip_hdr*)
- pjsip_generic_string_hdr_create(inv->dlg->pool,
- &str_refer_sub,
- &str_true);
- pj_list_push_back(&hdr_list, hdr);
-
- }
-
- /* Accept the REFER request, send 2xx. */
- pjsip_xfer_accept(sub, rdata, code, &hdr_list);
-
- /* Create initial NOTIFY request */
- status = pjsip_xfer_notify( sub, PJSIP_EVSUB_STATE_ACTIVE,
- 100, NULL, &tdata);
- if (status != PJ_SUCCESS) {
- _debug("UserAgent: Unable to create NOTIFY to REFER -- %d", status);
- return;
- }
-
- /* Send initial NOTIFY request */
- status = pjsip_xfer_send_request( sub, tdata);
- if (status != PJ_SUCCESS) {
- _debug("UserAgent: Unable to send NOTIFY to REFER -- %d\n", status);
- return;
- }
- }
-
- /* We're cheating here.
- * We need to get a null terminated string from a pj_str_t.
- * So grab the pointer from the hvalue and NULL terminate it, knowing
- * that the NULL position will be occupied by a newline.
- */
- uri = refer_to->hvalue.ptr;
- uri[refer_to->hvalue.slen] = '\0';
-
- /* Now make the outgoing call. */
- tmp = std::string(uri);
-
- if(existing_call == NULL) {
- _debug("UserAgent: Call doesn't exist!\n");
- return;
- }
-
- AccountID accId = Manager::instance().getAccountFromCall(existing_call->getCallId());
- CallID newCallId = Manager::instance().getNewCallID();
-
- if(!Manager::instance().outgoingCall(accId, newCallId, tmp)) {
-
- /* Notify xferer about the error (if we have subscription) */
- if (sub) {
- status = pjsip_xfer_notify(sub, PJSIP_EVSUB_STATE_TERMINATED,
- 500, NULL, &tdata);
- if (status != PJ_SUCCESS) {
- _debug("UserAgent: Unable to create NOTIFY to REFER -- %d\n", status);
- return;
- }
- status = pjsip_xfer_send_request(sub, tdata);
- if (status != PJ_SUCCESS) {
- _debug("UserAgent: Unable to send NOTIFY to REFER -- %d\n", status);
- return;
- }
- }
- return;
- }
-
- SIPCall* newCall;
- SIPVoIPLink *link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink(accId));
- if(link) {
- newCall = dynamic_cast<SIPCall *>(link->getCall(newCallId));
- if(!newCall) {
- _debug("UserAgent: can not find the call from sipvoiplink!\n");
- return;
- }
- }
-
- if (sub) {
- /* Put the server subscription in inv_data.
- * Subsequent state changed in pjsua_inv_on_state_changed() will be
- * reported back to the server subscription.
- */
- newCall->setXferSub(sub);
-
- /* Put the invite_data in the subscription. */
- pjsip_evsub_set_mod_data(sub, _mod.id,
- newCall);
- }
-}
-
-void UserAgent::xfer_svr_cb(pjsip_evsub *sub, pjsip_event *event)
-{
- PJ_UNUSED_ARG(event);
-
- /*
- * When subscription is terminated, clear the xfer_sub member of
- * the inv_data.
- */
- if (pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_TERMINATED) {
- SIPCall *call;
-
- call = (SIPCall*) pjsip_evsub_get_mod_data(sub, getInstance()->getModId());
- if (!call)
- return;
-
- pjsip_evsub_set_mod_data(sub, getInstance()->getModId(), NULL);
- call->setXferSub(NULL);
-
- _debug("UserAgent: Xfer server subscription terminated\n");
- }
-}
diff --git a/src/useragent.h b/src/useragent.h
deleted file mode 100644
index b37257aeae33bd46bf9fae99a5a841fc655e013e..0000000000000000000000000000000000000000
--- a/src/useragent.h
+++ /dev/null
@@ -1,166 +0,0 @@
-/*
- * Copyright (C) 2004-2005 Savoir-Faire Linux inc.
- * Author: Yun Liu <yun.liu@savoirfairelinux.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
- */
-
-
-#ifndef _SIPMANAGER_H
-#define _SIPMANAGER_H
-
-#include <pjsip.h>
-#include <pjlib-util.h>
-#include <pjlib.h>
-#include <pjnath/stun_config.h>
-
-//TODO Remove this include if we don't need anything from it
-#include <pjsip_simple.h>
-
-#include <pjsip_ua.h>
-#include <pjmedia/sdp.h>
-#include <pjmedia/sdp_neg.h>
-
-#include <string>
-#include <vector>
-
-#define PJ_LOG_LEVEL 1
-
-typedef std::string AccountID;
-
-class SIPCall;
-
-class UserAgent
-{
-private:
- /** PJSIP Endpoint */
- pjsip_endpoint *_endpt;
- pj_sock_t _sock;
- //pjsip_module _appMod;
- pj_caching_pool _cp;
- pj_pool_t *_pool;
- pj_mutex_t *_mutex; /** Mutex protection for this data */
- pjsip_module _mod; /** PJSIP module. */
- pjsip_module _options_handler;
- bool _useStun;
- pj_str_t _stunHost;
- std::string _stunServer;
- bool validStunServer;
-
- /** Local Extern Address is the IP address seen by peers for SIP listener */
- std::string _localExternAddress;
- std::string _localIPAddress;
-
- /** Local Extern Port is the port seen by peers for SIP listener */
- unsigned int _localExternPort;
- unsigned int _localPort;
-
- /** For registration use only */
- int _regPort;
-
- pj_thread_t *_thread;
-
- static UserAgent *_current;
-
- /* Sleep with polling */
- void busy_sleep(unsigned msec);
-public:
- UserAgent();
- ~UserAgent();
-
- pj_status_t sipCreate();
-
- /**
- * This method is used to initialize the pjsip
- */
- pj_status_t sipInit();
-
-
- void sipDestory();
-
- /** Create SIP UDP Listener */
- int createUDPServer();
-
- /** Set whether it will use stun server */
- void setStunServer(const char *server);
-
- /** Set the port number user designated */
- void setSipPort(int port) { _regPort = port; }
-
- int getSipPort() { return _regPort; }
-
- pj_str_t getStunServer() { return _stunHost; }
-
- bool addAccount(AccountID id, pjsip_regc **regc, const std::string& server, const std::string& user, const std::string& passwd
- , const int& timeout);
- bool removeAccount(pjsip_regc *regc);
-
- pj_str_t buildContact(char *userName);
-
- bool loadSIPLocalIP();
-
- pj_status_t stunServerResolve();
-
- pjsip_endpoint* getEndPoint() {return _endpt;}
-
- std::string getLocalIP() {return _localExternAddress;}
-
- int getModId() {return _mod.id;}
-
- bool setCallAudioLocal(SIPCall* call);
-
- int answer(SIPCall* call);
-
- bool hangup(SIPCall* call);
-
- bool refuse(SIPCall* call);
-
- bool onhold(SIPCall *call);
- bool offhold(SIPCall *call);
-
- bool transfer(SIPCall *call, const std::string& to);
-
- void onCallTransfered(pjsip_inv_session *inv, pjsip_rx_data *rdata);
-
- bool makeOutgoingCall(const std::string& to, SIPCall* call, const AccountID& id);
-
- bool carryingDTMFdigits(SIPCall* call, char *msgBody);
-
- pj_pool_t *getAppPool() {return _pool;}
- static pj_bool_t mod_on_rx_request(pjsip_rx_data *rdata);
- static pj_bool_t mod_on_rx_response(pjsip_rx_data *rdata UNUSED) {return PJ_SUCCESS;}
- static void regc_cb(struct pjsip_regc_cbparam *param);
- static void xfer_func_cb( pjsip_evsub *sub, pjsip_event *event);
- static void xfer_svr_cb(pjsip_evsub *sub, pjsip_event *event);
- static void call_on_media_update( pjsip_inv_session *inv UNUSED, pj_status_t status UNUSED) {}
- static void call_on_state_changed( pjsip_inv_session *inv, pjsip_event *e);
- static void call_on_forked(pjsip_inv_session *inv, pjsip_event *e);
- static void call_on_tsx_changed(pjsip_inv_session *inv, pjsip_transaction *tsx, pjsip_event *e);
- static int start_thread(void *arg);
- static UserAgent* getInstance() {return _current;}
-
- static void set_voicemail_info( AccountID account, pjsip_msg_body *body );
-private:
-
- // Copy Constructor
- UserAgent(const UserAgent& rh);
-
- // Assignment Operator
- UserAgent& operator=( const UserAgent& rh);
-};
-
-
-#endif /* _SIPMANAGER_H */
-
diff --git a/src/voiplink.cpp b/src/voiplink.cpp
index fe62b20467a1047fa2a1965618de1bff4abeea79..77a0c95ed11f7cc05646e0baa21a311501310ab8 100644
--- a/src/voiplink.cpp
+++ b/src/voiplink.cpp
@@ -26,7 +26,6 @@
VoIPLink::VoIPLink(const AccountID& accountID) : _accountID(accountID), _localIPAddress("127.0.0.1"), _localPort(0), _initDone(false)
{
- setRegistrationState(VoIPLink::Unregistered);
}
VoIPLink::~VoIPLink (void)
@@ -64,8 +63,7 @@ Call* VoIPLink::getCall(const CallID& id)
return 0;
}
-bool
-VoIPLink::clearCallMap()
+bool VoIPLink::clearCallMap()
{
ost::MutexLock m(_callMapMutex);
CallMap::iterator iter = _callMap.begin();
@@ -78,9 +76,11 @@ VoIPLink::clearCallMap()
return true;
}
-void VoIPLink::setRegistrationState(const RegistrationState state)
+Account* VoIPLink::getAccountPtr(void)
{
- _registrationState = state;
- // Notify the client
- Manager::instance().connectionStatusNotification( );
+ Account* account;
+ AccountID id;
+
+ id = getAccountID();
+ return Manager::instance().getAccount(id);
}
diff --git a/src/voiplink.h b/src/voiplink.h
index e095c1922ccc24af99754eddd5ed79b1d2ebc83a..4038394ccf6446663bba7e5d0d465d7cd955c2a8 100644
--- a/src/voiplink.h
+++ b/src/voiplink.h
@@ -24,8 +24,10 @@
#define __VOIP_LINK_H__
#include "call.h"
+#include "account.h"
class AudioCodec;
+class Account;
/** Define AccountID type */
typedef std::string AccountID;
@@ -38,218 +40,189 @@ typedef std::map<CallID, Call*> CallMap;
* @brief Listener and manager interface for each VoIP protocol
*/
class VoIPLink {
- public:
-
- /**
- * Constructor
- * @param accountID The account identifier
- */
- VoIPLink(const AccountID& accountID);
-
- /**
- * Virtual destructor
- */
- virtual ~VoIPLink (void);
-
- /** Contains all the state an Voip can be in */
- enum RegistrationState {Unregistered, Trying, Registered, Error, ErrorAuth , ErrorNetwork , ErrorHost, ErrorExistStun, ErrorConfStun};
- typedef enum RegistrationState RegistrationState;
-
- /**
- * Virtual method
- * Event listener. Each event send by the call manager is received and handled from here
- */
- virtual void getEvent (void) = 0;
-
- /**
- * Virtual method
- * Try to initiate the pjsip engine/thread and set config
- * @return bool True if OK
- */
- virtual bool init (void) = 0;
-
- /**
- * Virtual method
- * Delete link-related stuuf like calls
- */
- virtual void terminate (void) = 0;
-
- /**
- * Virtual method
- * Build and send SIP registration request
- * @return bool True on success
- * false otherwise
- */
- virtual int sendRegister (void) = 0;
-
- /**
- * Virtual method
- * Build and send SIP unregistration request
- * @return bool True on success
- * false otherwise
- */
- virtual int sendUnregister (void) = 0;
-
- /**
- * Place a new call
- * @param id The call identifier
- * @param toUrl The Sip address of the recipient of the call
- * @return Call* The current call
- */
- virtual Call* newOutgoingCall(const CallID& id, const std::string& toUrl) = 0;
- /**
- * Answer the call
- * @param id The call identifier
- * @return bool True on success
- */
- virtual bool answer(const CallID& id) = 0;
-
- /**
- * Hang up a call
- * @param id The call identifier
- * @return bool True on success
- */
- virtual bool hangup(const CallID& id) = 0;
-
- /**
- * Cancel the call dialing
- * @param id The call identifier
- * @return bool True on success
- */
- virtual bool cancel(const CallID& id) = 0;
-
- /**
- * Put a call on hold
- * @param id The call identifier
- * @return bool True on success
- */
- virtual bool onhold(const CallID& id) = 0;
-
- /**
- * Resume a call from hold state
- * @param id The call identifier
- * @return bool True on success
- */
- virtual bool offhold(const CallID& id) = 0;
-
- /**
- * Transfer a call to specified URI
- * @param id The call identifier
- * @param to The recipient of the call
- * @return bool True on success
- */
- virtual bool transfer(const CallID& id, const std::string& to) = 0;
-
- /**
- * Refuse incoming call
- * @param id The call identifier
- * @return bool True on success
- */
- virtual bool refuse(const CallID& id) = 0;
-
- /**
- * Send DTMF
- * @param id The call identifier
- * @param code The char code
- * @return bool True on success
- */
- virtual bool carryingDTMFdigits(const CallID& id, char code) = 0;
-
- /* Accessors */
- std::string& getFullName (void) { return _fullname; }
- void setFullName (const std::string& fullname) { _fullname = fullname; }
-
- std::string& getHostname (void) { return _hostname; }
- void setHostname (const std::string& hostname) { _hostname = hostname; }
-
- std::string& getUsername (void) { return _username; }
- void setUsername (const std::string& username) { _username = username; }
-
- std::string& getPassword (void) { return _password; }
- void setPassword (const std::string& password) { _password = password; }
-
- /**
- * @return AccountID parent Account's ID
- */
- AccountID& getAccountID(void) { return _accountID; }
-
- /**
- * @param accountID The account identifier
- */
- void setAccountID( const AccountID& accountID) { _accountID = accountID; }
-
- /** Get the call pointer from the call map (protected by mutex)
- * @param id A Call ID
- * @return Call* Call pointer or 0
- */
- Call* getCall(const CallID& id);
-
- /**
- * Get connection status
- * @return Connection status
- */
- RegistrationState getRegistrationState() { return _registrationState; }
-
- /**
- * Set new registration state
- * @param state The registration state
- */
- void setRegistrationState(const RegistrationState state);
-
- private:
- std::string _hostname;
- std::string _username;
- std::string _password;
- std::string _fullname;
-
- /**
- * ID of parent's Account
- */
- AccountID _accountID;
-
- /**
- * State of registration
- */
- RegistrationState _registrationState;
-
-public:
- /** Add a call to the call map (protected by mutex)
- * @param call A call pointer with a unique pointer
- * @return bool True if the call was unique and added
- */
- bool addCall(Call* call);
-
- /** Remove a call from the call map (protected by mutex)
- * @param id A Call ID
- * @return bool True if the call was correctly removed
- */
- bool removeCall(const CallID& id);
-
- /**
- * Remove all the call from the map
- * @return bool True on success
- */
- bool clearCallMap();
-
-
-protected:
- /** Contains all the calls for this Link, protected by mutex */
- CallMap _callMap;
-
- /** Mutex to protect call map */
- ost::Mutex _callMapMutex;
-
- /** Get Local IP Address (ie: 127.0.0.1, 192.168.0.1, ...) */
- std::string _localIPAddress;
-
- /** Get local listening port (5060 for SIP, ...) */
- unsigned int _localPort;
-
- /** Whether init() was called already or not
- * This should be used in [IAX|SIP]VoIPLink::init() and terminate(), to
- * indicate that init() was called, or reset by terminate().
- */
- bool _initDone;
+
+ public:
+ /**
+ * Constructor
+ * @param accountID The account identifier
+ */
+ VoIPLink(const AccountID& accountID);
+
+ /**
+ * Virtual destructor
+ */
+ virtual ~VoIPLink (void);
+
+
+ /**
+ * Virtual method
+ * Event listener. Each event send by the call manager is received and handled from here
+ */
+ virtual void getEvent (void) = 0;
+
+ /**
+ * Virtual method
+ * Try to initiate the communication layer and set config
+ * @return bool True if OK
+ */
+ virtual bool init (void) = 0;
+
+ /**
+ * Virtual method
+ * Delete link-related stuff like calls
+ */
+ virtual void terminate (void) = 0;
+
+ /**
+ * Virtual method
+ * Build and send account registration request
+ * @return bool True on success
+ * false otherwise
+ */
+ virtual int sendRegister ( AccountID id ) = 0;
+
+ /**
+ * Virtual method
+ * Build and send account unregistration request
+ * @return bool True on success
+ * false otherwise
+ */
+ virtual int sendUnregister ( AccountID id ) = 0;
+
+ /**
+ * Place a new call
+ * @param id The call identifier
+ * @param toUrl The address of the recipient of the call
+ * @return Call* The current call
+ */
+ virtual Call* newOutgoingCall(const CallID& id, const std::string& toUrl) = 0;
+
+ /**
+ * Answer the call
+ * @param id The call identifier
+ * @return bool True on success
+ */
+ virtual bool answer(const CallID& id) = 0;
+
+ /**
+ * Hang up a call
+ * @param id The call identifier
+ * @return bool True on success
+ */
+ virtual bool hangup(const CallID& id) = 0;
+
+ /**
+ * Cancel the call dialing
+ * @param id The call identifier
+ * @return bool True on success
+ */
+ virtual bool cancel(const CallID& id) = 0;
+
+ /**
+ * Put a call on hold
+ * @param id The call identifier
+ * @return bool True on success
+ */
+ virtual bool onhold(const CallID& id) = 0;
+
+ /**
+ * Resume a call from hold state
+ * @param id The call identifier
+ * @return bool True on success
+ */
+ virtual bool offhold(const CallID& id) = 0;
+
+ /**
+ * Transfer a call to specified URI
+ * @param id The call identifier
+ * @param to The recipient of the call
+ * @return bool True on success
+ */
+ virtual bool transfer(const CallID& id, const std::string& to) = 0;
+
+ /**
+ * Refuse incoming call
+ * @param id The call identifier
+ * @return bool True on success
+ */
+ virtual bool refuse(const CallID& id) = 0;
+
+ /**
+ * Send DTMF
+ * @param id The call identifier
+ * @param code The char code
+ * @return bool True on success
+ */
+ virtual bool carryingDTMFdigits(const CallID& id, char code) = 0;
+
+ bool initDone (void) { return _initDone; }
+ void initDone (bool state) { _initDone = state; }
+
+ std::string getLocalIPAddress (void) { return _localIPAddress; }
+
+ /** Add a call to the call map (protected by mutex)
+ * @param call A call pointer with a unique pointer
+ * @return bool True if the call was unique and added
+ */
+ bool addCall(Call* call);
+
+ /** Remove a call from the call map (protected by mutex)
+ * @param id A Call ID
+ * @return bool True if the call was correctly removed
+ */
+ bool removeCall(const CallID& id);
+
+ /**
+ * Remove all the call from the map
+ * @return bool True on success
+ */
+ bool clearCallMap();
+
+ /**
+ * @return AccountID parent Account's ID
+ */
+ inline AccountID& getAccountID(void) { return _accountID; }
+
+ Account* getAccountPtr(void);
+
+ /**
+ * @param accountID The account identifier
+ */
+ inline void setAccountID( const AccountID& accountID) { _accountID = accountID; }
+
+ /**
+ * Get the call pointer from the call map (protected by mutex)
+ * @param id A Call ID
+ * @return Call* Call pointer or 0
+ */
+ Call* getCall(const CallID& id);
+
+ virtual void setStunServer( const std::string &server ) = 0;
+
+ private:
+ /**
+ * ID of parent's Account
+ */
+ AccountID _accountID;
+
+ protected:
+ /** Contains all the calls for this Link, protected by mutex */
+ CallMap _callMap;
+
+ /** Mutex to protect call map */
+ ost::Mutex _callMapMutex;
+
+ /** Get Local IP Address (ie: 127.0.0.1, 192.168.0.1, ...) */
+ std::string _localIPAddress;
+
+ /** Get local listening port (5060 for SIP, ...) */
+ unsigned int _localPort;
+
+ /** Whether init() was called already or not
+ * This should be used in [IAX|SIP]VoIPLink::init() and terminate(), to
+ * indicate that init() was called, or reset by terminate().
+ */
+ bool _initDone;
};
#endif // __VOIP_LINK_H__
diff --git a/src/zeroconf/DNSQueryThread.cpp b/src/zeroconf/DNSQueryThread.cpp
deleted file mode 100644
index a5e9dd3fd2208eb5466264ce7d5c078961a9eedc..0000000000000000000000000000000000000000
--- a/src/zeroconf/DNSQueryThread.cpp
+++ /dev/null
@@ -1,69 +0,0 @@
-/**
- * Copyright (C) 2005 Savoir-Faire Linux inc.
- * Author: Yan Morin <yan.morin@savoirfairelinux.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
- */
-
-#include "DNSQueryThread.h"
-#include "DNSService.h"
-
-/**
- * Construct a DNSQueryThread and initialize the cancel to deferred
- */
-DNSQueryThread::DNSQueryThread(DNSService *parent, const char *regtype) : ost::Thread()
-{
- _parent = parent;
- _regtype = regtype;
- _serviceRef = NULL;
- setCancel(cancelDeferred);
-}
-
-/**
- * Destruct a DNSQueryThread
- */
-DNSQueryThread::~DNSQueryThread()
-{
- if (_serviceRef) {
- DNSServiceRefDeallocate(_serviceRef);
- }
- terminate();
- _parent = NULL;
- _regtype = NULL;
- _serviceRef = NULL;
-}
-
-/**
- * Running loop
- */
-void
-DNSQueryThread::run() {
- DNSServiceErrorType theErr=0; // NULL;
- DNSServiceFlags resultFlags=0;
-
- theErr = DNSServiceBrowse(&_serviceRef,
- resultFlags,
- 0, // all interfaces
- _regtype,
- NULL,
- DNSServiceAddServicesCallback,
- (void*)_parent);
-
- if (theErr == kDNSServiceErr_NoError) {
- while(!testCancel()) {
- DNSServiceProcessResult(_serviceRef); // blockage if none...
- }
- }
-}
diff --git a/src/zeroconf/DNSQueryThread.h b/src/zeroconf/DNSQueryThread.h
deleted file mode 100644
index 2ea0bc2cab61d49f1493cccee103e8e8fddd9d33..0000000000000000000000000000000000000000
--- a/src/zeroconf/DNSQueryThread.h
+++ /dev/null
@@ -1,40 +0,0 @@
-/**
- * Copyright (C) 2005 Savoir-Faire Linux inc.
- * Author: Yan Morin <yan.morin@savoirfairelinux.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
- */
-#ifndef __DNSSD_DNSQUERYTHREAD_H__
-#define __DNSSD_DNSQUERYTHREAD_H__
-
-#include <cc++/thread.h>
-#include <dns_sd.h>
-
-class DNSService;
-class DNSQueryThread : public ost::Thread
-{
-public:
- DNSQueryThread(DNSService *parent, const char *regtype);
- ~DNSQueryThread();
- virtual void run(); // looking for services
-
-private:
- DNSService *_parent; // parent service
- DNSServiceRef _serviceRef; // service reference
- const char *_regtype; // service type and socket type (_sip._udp by example)
-};
-
-
-#endif // __DNSSD_DNSQUERYTHREAD_H__
diff --git a/src/zeroconf/DNSService.cpp b/src/zeroconf/DNSService.cpp
deleted file mode 100644
index e2919fe27d93ec7c431ec7e30cb53d1469b462f2..0000000000000000000000000000000000000000
--- a/src/zeroconf/DNSService.cpp
+++ /dev/null
@@ -1,237 +0,0 @@
-/**
- * Copyright (C) 2005 Savoir-Faire Linux inc.
- * Author: Yan Morin <yan.morin@savoirfairelinux.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
- */
-
- /**
- * Inspired by http://braden.machacking.net/zerobrowse.cpp and
- * http://developer.kde.org/documentation/library/3.4-api/dnssd/html/remoteservice_8cpp-source.html
- */
-#include "DNSService.h"
-#include "DNSServiceTXTRecord.h"
-#include "DNSQueryThread.h"
-
-#include <cc++/thread.h>
-
-/**
- * Simple Empty Constructor
- */
-DNSService::DNSService()
-{
- _start = false;
- _regtypeList.push_back("_sip._udp");
-#ifdef USE_IAX2
- _regtypeList.push_back("_iax._udp");
-#endif
-
- // for the thread, the ifdef add a dynamic _regtypeList problem
- for (std::list<std::string>::iterator iterThread=_regtypeList.begin();
- iterThread!=_regtypeList.end();
- iterThread++) {
- _queryThread.push_back(new DNSQueryThread(this, (*iterThread).c_str()));
- }
-}
-
-/**
- * Simple Empty Destructor
- */
-DNSService::~DNSService()
-{
- int cntThread = _queryThread.size();
- for (int iThread=0;iThread<cntThread;iThread++) {
- delete _queryThread[iThread]; _queryThread[iThread] = NULL;
- }
-}
-
-/**
- * Look for zeroconf services and add them to _services
- */
-void
-DNSService::startScanServices()
-{
- for (std::vector<DNSQueryThread *>::iterator iter = _queryThread.begin();iter!=_queryThread.end();iter++) {
- (*iter)->start();
- }
- _start = true;
-}
-
-/**
- * Add one service to the list of actual services
- * @param service Service to add to the list
- */
-void DNSService::addService(const std::string &service)
-{
- DNSServiceTXTRecord txtRecord;
- _mutex.enterMutex();
- _services[service] = txtRecord;
- // we leave before the queryService since, each
- // thread will modify a DNSServiceTXTRecord of a difference services
- _mutex.leaveMutex();
- notify();
- queryService(service);
-}
-
-/**
- * Remove one service to the list of actual services
- * @param service Service to remove to the list
- */
-void DNSService::removeService(const std::string &service)
-{
- _mutex.enterMutex();
- _services.erase(service);
- _mutex.leaveMutex();
- notify();
-}
-
-/**
- * Return every services
- */
-DNSServiceMap
-DNSService::getServices()
-{
- ost::MutexLock m(_mutex);
- return _services;
-}
-
-
-/**
- * Query a service and wait for the anwser
- * the queryCallback will show the result
- * @param service The service full adress
- */
-void
-DNSService::queryService(const std::string &service)
-{
- DNSServiceErrorType theErr=0;
- DNSServiceRef myServRef=0;
- DNSServiceFlags resultFlags=0;
-
- theErr = DNSServiceQueryRecord(&myServRef,
- resultFlags,
- 0,
- service.c_str(),
- kDNSServiceType_TXT,
- kDNSServiceClass_IN,
- DNSServiceQueryRecordCallback,
- (void*)this);
- if (theErr == kDNSServiceErr_NoError) {
- DNSServiceProcessResult(myServRef); // blockage...
- DNSServiceRefDeallocate(myServRef);
- }
-}
-
-/**
- * Overloadding queryService
- * @param service service name
- * @param regtype registred type of service
- * @param domain domain (habitually local.)
- */
-void
-DNSService::queryService(const char *service, const char *regtype, const char *domain)
-{
- char serviceName[kDNSServiceMaxDomainName+1];
- DNSServiceConstructFullName(serviceName, service, regtype, domain);
- queryService(std::string(serviceName));
-}
-
-/**
- * Add a txt record with the queryService callback answser data
- * @param rdlen the length of the txt record data
- * @param rdata txt record data
- */
-void
-DNSService::addTXTRecord(const char *fullname, uint16_t rdlen, const void *rdata)
-{
- char key[256];
-
- const char *value;
- uint8_t valueLen; // 0 to 256 by type restriction
- char valueTab[256];
-
-
- uint16_t keyCount = TXTRecordGetCount(rdlen, rdata);
- for (int iKey=0; iKey<keyCount; iKey++) {
- TXTRecordGetItemAtIndex (rdlen, rdata, iKey, 256, key, &valueLen, (const void **)(&value));
- if (value) {
- bcopy(value, valueTab, valueLen);
- valueTab[valueLen]='\0';
- _mutex.enterMutex(); // extra-careful
- _services[fullname].addKeyValue(key, valueTab);
- _mutex.leaveMutex();
- } else {
- _mutex.enterMutex();
- _services[fullname].removeKey(key);
- _mutex.leaveMutex();
- }
- }
-
- notify();
-}
-
-void
-DNSServiceAddServicesCallback(DNSServiceRef,
- DNSServiceFlags flags,
- uint32_t,
- DNSServiceErrorType errorCode,
- const char *serviceName,
- const char *replyType,
- const char *replyDomain,
- void *context)
-{
- if (errorCode==kDNSServiceErr_NoError) {
-
- if (flags) {
- DNSService *service = (DNSService*)context;
- std::string tempService;
- tempService = std::string(serviceName) + "." + std::string(replyType) + std::string(replyDomain);
- if (flags&kDNSServiceFlagsAdd) {
-// _debug("DNSServiceAddServicesCallback call addService\n");
- service->addService(tempService);
- } else {
-// _debug("DNSServiceAddServicesCallback call removeService\n");
- service->removeService(tempService);
- }
- }
- } else {
- // TODO: error handling
- }
-}
-
-void
-DNSServiceQueryRecordCallback(
- DNSServiceRef,
- DNSServiceFlags flags,
- uint32_t,
- DNSServiceErrorType errorCode,
- const char *fullname,
- uint16_t,
- uint16_t,
- uint16_t rdlen,
- const void *rdata,
- uint32_t,
- void *context)
-{
- if (errorCode==kDNSServiceErr_NoError) {
- if (flags&kDNSServiceFlagsAdd) {
-// _debug("DNSServiceQueryRecordCallback call addTXTRecord\n");
- ((DNSService *)context)->addTXTRecord(fullname, rdlen, rdata);
- } else {
-// _debug("DNSServiceQueryRecordCallback call removeService\n");
- ((DNSService *)context)->removeService(fullname);
- }
- }
-}
diff --git a/src/zeroconf/DNSService.h b/src/zeroconf/DNSService.h
deleted file mode 100644
index b1a840bf1cf6bf5b8076d9fb79c2785f84cb6ae4..0000000000000000000000000000000000000000
--- a/src/zeroconf/DNSService.h
+++ /dev/null
@@ -1,92 +0,0 @@
-/**
- * Copyright (C) 2005 Savoir-Faire Linux inc.
- * Author: Yan Morin <yan.morin@savoirfairelinux.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
- */
-#ifndef __DNSSD_DNSSERVICE_H__
-#define __DNSSD_DNSSERVICE_H__
-
-#include <string>
-#include <map>
-#include <list>
-#include <vector>
-
-#include <dns_sd.h>
-#include <cc++/thread.h>
-#include "../observer.h"
-
-class DNSQueryThread;
-class DNSServiceTXTRecord;
-
-typedef std::map<std::string, DNSServiceTXTRecord> DNSServiceMap;
-class DNSService : public Pattern::Subject
-{
-public:
- DNSService();
- ~DNSService();
-
- void startScanServices(); // looking for services
- void addService(const std::string &service); // adding every services
- void removeService(const std::string &service); // remove a service
- DNSServiceMap getServices(); // get all DNS Service
- void stop(); // after the browsing loop stop
-
- void queryService(const std::string &service); // query the TXT record of a service
- void queryService(const char *service, const char *regtype, const char *domain);
- void addTXTRecord(const char *fullname, uint16_t rdlen, const void *rdata);
- //void removeTXTRecord(const char *fullname);
-
- bool isStart() const { return _start; }
-
-private:
- DNSServiceMap _services; //map
-
- std::vector<DNSQueryThread *> _queryThread;
- /**
- * Mutex to protect access to _services on add/erase
- */
- ost::Mutex _mutex;
- /**
- * RegType List contains zeroconf services to register, like sip, iax2, ...
- * It will be use to initialize the DNSQueryThread
- */
- std::list<std::string> _regtypeList;
-
- bool _start;
-};
-
-void DNSServiceAddServicesCallback(DNSServiceRef sdRef,
- DNSServiceFlags flags,
- uint32_t interfaceIndex,
- DNSServiceErrorType errorCode,
- const char *serviceName,
- const char *replyType,
- const char *replyDomain,
- void *context);
-
-void DNSServiceQueryRecordCallback(DNSServiceRef DNSServiceRef,
- DNSServiceFlags flags,
- uint32_t interfaceIndex,
- DNSServiceErrorType errorCode,
- const char *fullname,
- uint16_t rrtype,
- uint16_t rrclass,
- uint16_t rdlen,
- const void *rdata,
- uint32_t ttl,
- void *context);
-
-#endif // __DNSSD_DNSSERVICE_H__
diff --git a/src/zeroconf/DNSServiceTXTRecord.cpp b/src/zeroconf/DNSServiceTXTRecord.cpp
deleted file mode 100644
index f12ed15f5b2b1f9468ffa72f8b71d3567102f5a2..0000000000000000000000000000000000000000
--- a/src/zeroconf/DNSServiceTXTRecord.cpp
+++ /dev/null
@@ -1,76 +0,0 @@
-/**
- * Copyright (C) 2005 Savoir-Faire Linux inc.
- * Author: Yan Morin <yan.morin@savoirfairelinux.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
- */
-#include "DNSServiceTXTRecord.h"
-
-/**
- * Simple constructor
- */
-DNSServiceTXTRecord::DNSServiceTXTRecord()
-{
-}
-
-/**
- * Simple destructor
- */
-DNSServiceTXTRecord::~DNSServiceTXTRecord()
-{
-}
-
-/**
- * add a pair of key/value inside the associative std::map
- * @param key unique key inside the std::map
- * @param value value associated to the key
- */
-void
-DNSServiceTXTRecord::addKeyValue(const std::string &key, const std::string &value)
-{
- _map[key] = value;
-}
-
-/**
- * remove a key inside the map or do nothing if it doesn't exist
- * @param key unique key inside the std::map
- */
-void
-DNSServiceTXTRecord::removeKey(const std::string &key)
-{
- _map.erase(key);
-}
-
-/**
- * get a value from a key
- * @param key unique key inside the std::map
- * @return the value or empty
- */
-const std::string &
-DNSServiceTXTRecord::getValue(const std::string &key)
-{
- return _map[key]; // return std::string("") if it's not there
-}
-
-/**
- * get a value from a key
- * @param key unique key inside the std::map
- * @return the value or empty
- */
-const std::string &
-DNSServiceTXTRecord::getValue(const char* key)
-{
- return getValue(std::string(key));
-}
diff --git a/src/zeroconf/DNSServiceTXTRecord.h b/src/zeroconf/DNSServiceTXTRecord.h
deleted file mode 100644
index d6d2742fd6efa5091dc04f63064f3eb28e555397..0000000000000000000000000000000000000000
--- a/src/zeroconf/DNSServiceTXTRecord.h
+++ /dev/null
@@ -1,43 +0,0 @@
-/**
- * Copyright (C) 2005 Savoir-Faire Linux inc.
- * Author: Yan Morin <yan.morin@savoirfairelinux.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
- */
-#ifndef __DNSSD_DNSSERVICETXTRECORD_H__
-#define __DNSSD_DNSSERVICETXTRECORD_H__
-
-#include <string>
-#include <map>
-
-typedef std::map<std::string, std::string> TXTRecordMap;
-class DNSServiceTXTRecord
-{
-public:
- DNSServiceTXTRecord();
- ~DNSServiceTXTRecord();
- void addKeyValue(const std::string &key, const std::string &value);
- void removeKey(const std::string &key);
- const std::string &getValue(const std::string &key);
- const std::string &getValue(const char *key);
- inline void clear(void) { _map.clear(); };
- inline int size(void) { return _map.size(); };
- TXTRecordMap getTXTRecords() { return _map; }
-
-private:
- TXTRecordMap _map;
-};
-
-#endif // __DNSSD_DNSSERVICETXTRECORD_H__
diff --git a/src/zeroconf/Makefile.am b/src/zeroconf/Makefile.am
deleted file mode 100644
index d2ed5b4e4a56d0330b7bad354ea7f2afec291086..0000000000000000000000000000000000000000
--- a/src/zeroconf/Makefile.am
+++ /dev/null
@@ -1,12 +0,0 @@
-SUBDIRS =
-
-noinst_LTLIBRARIES = libzeroconf.la
-
-libzeroconf_la_SOURCES = \
- DNSQueryThread.cpp DNSQueryThread.h \
- DNSService.cpp DNSService.h \
- DNSServiceTXTRecord.cpp DNSServiceTXTRecord.h
-
-AM_CXXFLAGS = $(libccext2_CFLAGS)
-libzeroconf_la_LIBADD = $(LIB_DNSSD)
-