diff --git a/configure.ac b/configure.ac
index 6ee3762e618ee5c7ce25a1288598e6cf9831e7c6..3e9b849a9675dbbf322d1271250abdda647c9c8a 100644
--- a/configure.ac
+++ b/configure.ac
@@ -39,7 +39,8 @@ AC_CONFIG_FILES([src/Makefile \
src/dbus/Makefile \
src/zeroconf/Makefile \
src/plug-in/audiorecorder/Makefile \
- src/plug-in/Makefile])
+ src/plug-in/Makefile \
+ src/plug-in/test/Makefile])
dnl Unitary test section
AC_CONFIG_FILES([test/Makefile])
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 68e0e2d423367cc834a682c13e40b233726aea64..a608749908d06706e08867979faf4fc470b3f899 100644
--- a/sflphone-gtk/src/dbus.c
+++ b/sflphone-gtk/src/dbus.c
@@ -1542,3 +1542,58 @@ 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 53945f899e37795647d7067937524fd242832d6a..6da95b9e2502d7123005b4f27f570949ed0f4845 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\" \
@@ -60,7 +59,6 @@ sflphoned_LDADD = \
@CCRTP_LIBS@ \
@ALSA_LIBS@ \
@PULSEAUDIO_LIBS@ \
- -luuid \
@SAMPLERATE_LIBS@
#sflphoned_LDFLAGS=-pg
@@ -81,8 +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..764d4628e1c2d780eb18b9ce4cdc23046f150f95 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 +119,27 @@ 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 );
+
+ //TODO inline?
+ 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 +152,32 @@ 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 +190,11 @@ class Account{
*/
bool _enabled;
+ /*
+ * The registration state of the account
+ */
+ RegistrationState _registrationState;
+
};
#endif
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..d141505e7e6c04ba4c0301ec921390d7fbb163a0 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
@@ -21,43 +20,22 @@
#include "eventthread.h"
#include "voiplink.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 )
-{
- stopIt = false;
- //start();
-}
+}
-bool
-EventThread::isStopped( void )
-{
- return stopIt;
-}
+/********************************************************************************************/
diff --git a/src/eventthread.h b/src/eventthread.h
index 8772c72043fec8d719b7c50aecf706dbdb241396..0d43a18d716f685388ebddec29e7e4edeb52abe2 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
@@ -31,24 +30,26 @@ 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;
};
+
#endif // __EVENT_THREAD_H__
diff --git a/src/global.h b/src/global.h
index 0321db6e16948bb2d0449d0fcd201e1f56c83cf4..deb1cdfecd895eb7984e65714163a450c685c1ae 100644
--- a/src/global.h
+++ b/src/global.h
@@ -26,6 +26,9 @@
#include <stdio.h>
#include <libintl.h>
#include <locale.h>
+#include <string>
+#include <stdlib.h>
+#include <sstream>
#define SFLPHONED_VERSION "0.9.2-4" /** Version number */
@@ -111,6 +114,7 @@ static const SOUND_FORMAT INT32 = 0x8;
#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..4de19c54cc995f4d39954775851299b67860451a 100644
--- a/src/iaxaccount.cpp
+++ b/src/iaxaccount.cpp
@@ -23,7 +23,7 @@
#include "iaxvoiplink.h"
IAXAccount::IAXAccount(const AccountID& accountID)
-: Account(accountID)
+: Account(accountID, "iax2")
{
_link = new IAXVoIPLink(accountID);
}
@@ -35,8 +35,7 @@ IAXAccount::~IAXAccount()
_link = NULL;
}
- int
-IAXAccount::registerVoIPLink()
+int IAXAccount::registerVoIPLink()
{
IAXVoIPLink *thislink;
@@ -50,7 +49,7 @@ IAXAccount::registerVoIPLink()
thislink->setPass(Manager::instance().getConfigString(_accountID, PASSWORD));
}
- _link->sendRegister();
+ _link->sendRegister( _accountID );
return SUCCESS;
}
@@ -58,7 +57,7 @@ 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..679e2527ce887adab5dbaeefbf9ec043d17af82f 100644
--- a/src/iaxvoiplink.cpp
+++ b/src/iaxvoiplink.cpp
@@ -21,6 +21,7 @@
#include "global.h" // for _debug
#include "iaxcall.h"
#include "eventthread.h"
+#include "iaxaccount.h"
#include "manager.h"
#include "audio/audiolayer.h"
@@ -207,7 +208,7 @@ IAXVoIPLink::getEvent()
// Refresh registration.
if (_nextRefreshStamp && _nextRefreshStamp - 2 < time(NULL)) {
- sendRegister();
+ sendRegister("");
}
// thread wait 3 millisecond
@@ -308,48 +309,52 @@ IAXVoIPLink::getIAXCall(const CallID& id)
int
-IAXVoIPLink::sendRegister()
+IAXVoIPLink::sendRegister(AccountID id)
{
- bool result = false;
- if (_host.empty()) {
- return false;
- }
- if (_user.empty()) {
- return false;
- }
+ IAXAccount *account;
+ bool result;
- // lock
- _mutexIAX.enterMutex();
+ result = false;
+ account = dynamic_cast<IAXAccount *> (Manager::instance().getAccount(id));
+ if (_host.empty()) {
+ return false;
+ }
+ if (_user.empty()) {
+ return false;
+ }
- // Always use a brand new session
- if (_regSession) {
- iax_destroy(_regSession);
- }
+ // lock
+ _mutexIAX.enterMutex();
- _regSession = iax_session_new();
+ // Always use a brand new session
+ if (_regSession) {
+ iax_destroy(_regSession);
+ }
- 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);
+ _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;
+
+ account->setRegistrationState(Trying);
}
// unlock
@@ -361,7 +366,7 @@ IAXVoIPLink::sendRegister()
int
-IAXVoIPLink::sendUnregister()
+IAXVoIPLink::sendUnregister(AccountID id)
{
_mutexIAX.enterMutex();
if (_regSession) {
@@ -377,7 +382,7 @@ IAXVoIPLink::sendUnregister()
_nextRefreshStamp = 0;
_debug("IAX2 send unregister\n");
- setRegistrationState(Unregistered);
+ //setRegistrationState(Unregistered);
return SUCCESS;
}
@@ -761,7 +766,8 @@ IAXVoIPLink::iaxHandleRegReply(iax_event* event)
iax_destroy(_regSession);
_mutexIAX.leaveMutex();
_regSession = NULL;
- setRegistrationState(ErrorAuth);
+ //TODO Restore that
+ //setRegistrationState(ErrorAuth);
}
else if (event->etype == IAX_EVENT_REGACK) {
@@ -783,7 +789,7 @@ 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);
+ //nsetRegistrationState(Registered);
}
}
@@ -917,6 +923,8 @@ IAXVoIPLink::iaxCodecMapToFormat(IAXCall* call)
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..8ad4082882f41b731a610714172167bb94e15f95 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
@@ -183,6 +183,8 @@ class IAXVoIPLink : public VoIPLink
void updateAudiolayer( void );
+ void setStunServer( const std::string &server ) {};
+
private:
/*
diff --git a/src/managerimpl.cpp b/src/managerimpl.cpp
index 626aca506c6114558d6a1db42fae6259e55a019b..c8c25d2b8432821c06f587c7b17ae0d3b38fa651 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();
@@ -123,82 +100,60 @@ 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");
- }
+ initVolume();
- initAudioDriver();
- selectAudioDriver();
+ if (_exist == 0) {
+ _debug("Cannot create config file in your home directory\n");
+ }
- // Initialize the list of supported audio codecs
- initAudioCodec();
+ initAudioDriver();
+ selectAudioDriver();
- getAudioInputDeviceList();
+ // Initialize the list of supported audio codecs
+ initAudioCodec();
- AudioLayer *audiolayer = getAudioDriver();
- if (audiolayer!=0) {
- unsigned int sampleRate = audiolayer->getSampleRate();
+ getAudioInputDeviceList();
- _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
@@ -237,27 +192,31 @@ 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
@@ -443,25 +402,29 @@ ManagerImpl::saveConfig (void)
int
ManagerImpl::initRegisterAccounts()
{
-
int status;
+ //TODO What the flag is for ??
bool flag = true;
AccountMap::iterator iter;
_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() );
@@ -721,7 +684,7 @@ 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();
@@ -1012,15 +975,17 @@ 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);
@@ -1402,7 +1367,27 @@ 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 );
@@ -1694,24 +1679,6 @@ ManagerImpl::switchAudioManager( void )
notifyErrClient( _audiodriver -> getErrorMessage());
}
-/**
- * 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);
-
- if (useZeroconf) {
- _DNSService->startScanServices();
- }
-#endif
-}
-
/**
* Init the volume for speakers/micro from 0 to 100 value
* Initialization: Main Thread
@@ -1750,88 +1717,16 @@ void ManagerImpl::setMicVolume(unsigned short 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)
-{
- 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)
+int ManagerImpl::getSipPort( void )
{
- 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
@@ -1964,7 +1859,7 @@ 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);
@@ -1973,15 +1868,15 @@ 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")))))))))
)
);
@@ -2006,6 +1901,7 @@ void ManagerImpl::setAccountDetails( const std::string& accountID, const std::ma
std::string accountType;
Account *acc;
+ VoIPLink *link;
accountType = (*details.find(CONFIG_ACCOUNT_TYPE)).second;
@@ -2019,28 +1915,28 @@ 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();
@@ -2056,6 +1952,8 @@ 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
@@ -2082,38 +1980,30 @@ 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. */
+ /** @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));
- }
-
- 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
@@ -2192,20 +2082,6 @@ 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()
{
@@ -2214,7 +2090,6 @@ ManagerImpl::loadAccountMap()
TokenList sections = _config.getSections();
std::string accountType;
Account* tmpAccount;
- std::string port;
TokenList::iterator iter = sections.begin();
while(iter != sections.end()) {
@@ -2226,28 +2101,7 @@ 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);
@@ -2262,8 +2116,6 @@ ManagerImpl::loadAccountMap()
nbAccount++;
}
- _debug("\n");
-
iter++;
}
@@ -2331,8 +2183,25 @@ 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 ) {
@@ -2351,37 +2220,6 @@ 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 b13fc911a88d79580e739ac8ea8b5439d5ad7424..222c7154788fd86d4cc79d2fd3ad16c048584f82 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/plug-in/Makefile.am b/src/plug-in/Makefile.am
index 7555fe7f3e45ab1427c068e716240af3c47f70ea..b69f39ca9cdca75d415b6470d84f7a371e8157c1 100644
--- a/src/plug-in/Makefile.am
+++ b/src/plug-in/Makefile.am
@@ -4,7 +4,9 @@ SUBDIRS=audiorecorder
noinst_LTLIBRARIES = libplugin.la
+SUBDIRS=test
+
libplugin_la_SOURCES = \
pluginmanager.cpp \
- plugin.h
+ plugin.cpp
diff --git a/src/plug-in/plugin.cpp b/src/plug-in/plugin.cpp
index 495437076549dee7c9bafbdc5ca6782076400d97..deaf1a4818fbe8a01a720b0b856c36f1ac33a277 100644
--- a/src/plug-in/plugin.cpp
+++ b/src/plug-in/plugin.cpp
@@ -1,27 +1,15 @@
-#include "plugin.h"
+#include "plugin.h"
-::sflphone::Plugin::Plugin( const std::string &filename UNUSED )
+::sflphone::Plugin::Plugin (void *handle, PluginInterface *interface)
+ :_handlePtr(handle), _interface(interface)
{
- //TODO IMPLEMENT
}
-::sflphone::Plugin::Plugin( const Plugin &plugin UNUSED )
+::sflphone::Plugin::Plugin (PluginInterface *interface)
+ :_interface(interface)
{
- //TODO IMPLEMENT
}
-::sflphone::Plugin::~Plugin()
+::sflphone::Plugin::~Plugin ()
{
- //TODO IMPLEMENT
-}
-
-int ::sflphone::Plugin::getCoreVersion( void ) const
-{
- //TODO IMPLEMENT
- return 1;
-}
-
-void ::sflphone::Plugin::registerPlugin( PluginManager & )
-{
- //TODO IMPLEMENT
}
diff --git a/src/plug-in/plugin.h b/src/plug-in/plugin.h
index 100d6ea5b40bc618e3e6cae9c4435b9b192af676..fe548361eb1098e566ffacf00cb58d61ba31a236 100644
--- a/src/plug-in/plugin.h
+++ b/src/plug-in/plugin.h
@@ -1,43 +1,32 @@
#ifndef PLUGIN_H
#define PLUGIN_H
-#include <string>
-
-#include "global.h"
-
-/*
- * @file plugin.h
- * @brief Define a plugin object
- */
+#include "plugininterface.h"
namespace sflphone {
-class PluginManager;
+ class PluginInterface;
class Plugin {
-
- public:
- Plugin( const std::string &name );
- //Plugin( const Plugin &plugin );
- virtual ~Plugin() {}
public:
- /**
- * Return the minimal core version required so that the plugin could work
- * @return int The version required
- */
- virtual int getCoreVersion() const = 0;
-
- /**
- * Register the plugin to the plugin manager
- */
- virtual void registerPlugin( PluginManager & ) = 0;
- private:
- Plugin &operator =(const Plugin &plugin);
+ Plugin (void*, PluginInterface *interface);
+ Plugin (PluginInterface *interface);
+ ~Plugin ();
+
+ void setName (std::string name);
+ private:
+ std::string _name;
+ int _version_major;
+ int _version_minor;
+ int _required;
+ void *_handlePtr;
+ PluginInterface *_interface;
+
+ friend class PluginTest;
+ friend class PluginManager;
};
}
-
#endif //PLUGIN_H
-
diff --git a/src/plug-in/plugininterface.h b/src/plug-in/plugininterface.h
new file mode 100644
index 0000000000000000000000000000000000000000..ef45d99dd9563a2c9d218a26a82c4485f04ae8d7
--- /dev/null
+++ b/src/plug-in/plugininterface.h
@@ -0,0 +1,50 @@
+#ifndef PLUGIN_INTERFACE_H
+#define PLUGIN_INTERFACE_H
+
+#include <string>
+#include "global.h"
+
+#include "plugin.h"
+
+/*
+ * @file plugininterface.h
+ * @brief Define a plugin object
+ */
+
+namespace sflphone {
+
+ class Plugin;
+
+ class PluginInterface {
+
+ public:
+ PluginInterface( const std::string &name ){
+ _name = name;
+ }
+
+ virtual ~PluginInterface() {}
+
+ inline std::string getInterfaceName (void) { return _name; }
+
+ /**
+ * Return the minimal core version required so that the plugin could work
+ * @return int The version required
+ */
+ virtual int initFunc () = 0;
+
+ virtual int registerFunc (Plugin **plugin) = 0;
+
+ private:
+ PluginInterface &operator =(const PluginInterface &plugin);
+
+ std::string _name;
+ };
+
+ typedef PluginInterface* createFunc (void);
+
+ typedef void destroyFunc( PluginInterface* );
+
+}
+
+#endif //PLUGIN_INTERFACE_H
+
diff --git a/src/plug-in/pluginmanager.cpp b/src/plug-in/pluginmanager.cpp
index df88799a80d8f98a0af1290e03de65cdf759314d..f8135f57806590ea3c32f5c304f3de3be8814896 100644
--- a/src/plug-in/pluginmanager.cpp
+++ b/src/plug-in/pluginmanager.cpp
@@ -3,7 +3,19 @@
#include "pluginmanager.h"
-::sflphone::PluginManager::PluginManager():_loadedPlugins()
+::sflphone::PluginManager* ::sflphone::PluginManager::_instance = 0;
+
+ ::sflphone::PluginManager*
+::sflphone::PluginManager::instance()
+{
+ if(!_instance){
+ return new PluginManager();
+ }
+ return _instance;
+}
+
+::sflphone::PluginManager::PluginManager()
+ :_loadedPlugins()
{
_instance = this;
}
@@ -13,22 +25,16 @@
_instance = 0;
}
-::sflphone::PluginManager* ::sflphone::PluginManager::instance()
-{
- if(! _instance ){
- _instance = new PluginManager();
- }
- return _instance;
-}
-
-
-int ::sflphone::PluginManager::loadPlugins( const std::string &path )
+ int
+::sflphone::PluginManager::loadPlugins (const std::string &path)
{
std::string pluginDir, current;
- ::sflphone::Plugin *plugin;
+ ::sflphone::PluginInterface *interface;
DIR *dir;
dirent *dirStruct;
int result=0;
+ void *handle;
+ createFunc* createPlugin;
const std::string pDir = "..";
const std::string cDir = ".";
@@ -46,18 +52,31 @@ int ::sflphone::PluginManager::loadPlugins( const std::string &path )
current = dirStruct->d_name;
/* Test if the current item is not the parent or the current directory */
if( current != pDir && current != cDir ){
- loadDynamicLibrary( current );
- result++;
- }
+ handle = loadDynamicLibrary( pluginDir + current );
+
+ if(instanciatePlugin (handle, &interface) != 0)
+ {
+ _debug("Error instanciating the plugin ...\n");
+ return 1;
+ }
+
+ if(registerPlugin (handle, interface) != 0)
+ _debug("Error registering the plugin ...\n");
+ return 1;
+ }
}
}
+ else
+ return 1;
+
/* Close the directory */
closedir( dir );
- return result;
+ return 0;
}
-::sflphone::Plugin* ::sflphone::PluginManager::isPluginLoaded( const std::string &name )
+ ::sflphone::Plugin*
+::sflphone::PluginManager::isPluginLoaded (const std::string &name)
{
if(_loadedPlugins.empty()) return NULL;
@@ -77,31 +96,68 @@ int ::sflphone::PluginManager::loadPlugins( const std::string &path )
return NULL;
}
-void* ::sflphone::PluginManager::loadDynamicLibrary( const std::string& filename ) {
+
+ void*
+::sflphone::PluginManager::loadDynamicLibrary (const std::string& filename)
+{
void *pluginHandlePtr = NULL;
const char *error;
_debug("Loading dynamic library %s\n", filename.c_str());
-#if defined(Q_OS_UNIX)
/* Load the library */
pluginHandlePtr = dlopen( filename.c_str(), RTLD_LAZY );
if( !pluginHandlePtr ) {
error = dlerror();
_debug("Error while opening plug-in: %s\n", error);
+ return NULL;
}
dlerror();
-#endif
-
return pluginHandlePtr;
+
+}
+
+ int
+::sflphone::PluginManager::instanciatePlugin (void *handlePtr, ::sflphone::PluginInterface **plugin)
+{
+ createFunc *createPlugin;
+
+ createPlugin = (createFunc*)dlsym(handlePtr, "create");
+ if( dlerror() )
+ {
+ _debug("Error creating the plugin: %s\n", dlerror());
+ return 1;
+ }
+ *plugin = createPlugin();
+ return 0;
}
-void ::sflphone::PluginManager::unloadDynamicLibrary( void * pluginHandlePtr ) {
+ int
+::sflphone::PluginManager::registerPlugin (void *handlePtr, PluginInterface *interface)
+{
+ Plugin *myplugin;
+ std::string name;
+
+ if( !( handlePtr && interface!=0 ) )
+ return 1;
+ /* Fetch information from the loaded plugin interface */
+ if(interface->registerFunc (&myplugin) != 0)
+ return 1;
+ /* Creation of the plugin wrapper */
+ myplugin = new Plugin (handlePtr, interface);
+
+ /* Add the data in the loaded plugin map */
+ _loadedPlugins[ myplugin->_name ] = myplugin;
+ return 0;
+}
+
+ void
+::sflphone::PluginManager::unloadDynamicLibrary (void * pluginHandlePtr)
+{
dlclose( pluginHandlePtr );
dlerror();
}
-::sflphone::PluginManager* ::sflphone::PluginManager::_instance = 0;
diff --git a/src/plug-in/pluginmanager.h b/src/plug-in/pluginmanager.h
index 78d988cd0c9f51df87d19b5fd9b33822c3118fe1..671834369da67dc151cd24e8ab9ca62a0261d6c1 100644
--- a/src/plug-in/pluginmanager.h
+++ b/src/plug-in/pluginmanager.h
@@ -6,7 +6,7 @@
* @brief Base class of the plugin manager
*/
-#include "plugin.h"
+#include "plugininterface.h"
#include "global.h"
#include <map>
@@ -16,56 +16,61 @@ namespace sflphone {
class PluginManager {
+
public:
- /**
- * Destructor
- */
- ~PluginManager();
-
- /**
- * Returns the unique instance of the plugin manager
- */
- static PluginManager* instance();
-
- /**
- * Load all the plugins found in a specific directory
- * @param path The absolute path to the directory
- * @return int The number of items loaded
- */
- int loadPlugins( const std::string &path = "" );
-
- /**
- * Check if a plugin has been already loaded
- * @param name The name of the plugin looked for
- * @return Plugin* The pointer on the plugin or NULL if not found
- */
- Plugin* isPluginLoaded( const std::string &name );
+ /**
+ * Destructor
+ */
+ ~PluginManager();
+
+ /**
+ * Returns the unique instance of the plugin manager
+ */
+ static PluginManager* instance();
+
+ /**
+ * Load all the plugins found in a specific directory
+ * @param path The absolute path to the directory
+ * @return int The number of items loaded
+ */
+ int loadPlugins( const std::string &path = "" );
+
+ int instanciatePlugin( void *handlePtr, PluginInterface** plugin );
+
+ /**
+ * Check if a plugin has been already loaded
+ * @param name The name of the plugin looked for
+ * @return Plugin* The pointer on the plugin or NULL if not found
+ */
+ Plugin* isPluginLoaded( const std::string &name );
+
+ int registerPlugin (void *handle, PluginInterface *interface);
private:
- /**
- * Default constructor
- */
- PluginManager();
-
- /**
- * Load a unix dynamic/shared library
- * @param filename The path to the dynamic/shared library
- * @return void* A pointer on it
- */
- void * loadDynamicLibrary( const std::string &filename );
-
- /**
- * Unload a unix dynamic/shared library
- * @param pluginHandleptr The pointer on the loaded plugin
- */
- void unloadDynamicLibrary( void * pluginHandlePtr );
-
- /* Map of plugins associated by their string name */
- typedef std::map<std::string, ::sflphone::Plugin*> pluginMap;
- pluginMap _loadedPlugins;
-
- /* The unique static instance */
- static PluginManager* _instance;
+ /**
+ * Default constructor
+ */
+ PluginManager();
+
+ /**
+ * Load a unix dynamic/shared library
+ * @param filename The path to the dynamic/shared library
+ * @return void* A pointer on it
+ */
+ void * loadDynamicLibrary( const std::string &filename );
+
+ /**
+ * Unload a unix dynamic/shared library
+ * @param pluginHandleptr The pointer on the loaded plugin
+ */
+ void unloadDynamicLibrary( void * pluginHandlePtr );
+
+ /* Map of plugins associated by their string name */
+ typedef std::map<std::string, Plugin*> pluginMap;
+ pluginMap _loadedPlugins;
+
+ /* The unique static instance */
+ static PluginManager* _instance;
};
}
diff --git a/src/plug-in/test/Makefile.am b/src/plug-in/test/Makefile.am
new file mode 100644
index 0000000000000000000000000000000000000000..6809be2c0a185e16f72edcef1b57103fbcbea8f8
--- /dev/null
+++ b/src/plug-in/test/Makefile.am
@@ -0,0 +1,19 @@
+include $(top_srcdir)/globals.mak
+
+PLUGIN_LIB = libplugintest.so
+libplugintest_so_SOURCES = pluginTest.cpp
+libplugintest_so_CXXFLAGS = -fPIC -g -Wall
+libplugintest_so_LDFLAGS = --shared -lc
+INSTALL_PLUGIN_RULE = install-libplugintest_so
+
+noinst_PROGRAMS = libplugintest.so
+
+install-exec-local: install-libplugintest_so
+uninstall-local: uninstall-libplugintest_so
+
+install-libplugintest_so: libplugintest.so
+ mkdir -p $(sflplugindir)
+ $(INSTALL_PROGRAM) libplugintest.so $(sflplugindir)
+
+uninstall-libplugintest_so:
+ rm -f $(sflplugindir)/libplugintest.so
diff --git a/src/plug-in/test/pluginTest.cpp b/src/plug-in/test/pluginTest.cpp
new file mode 100644
index 0000000000000000000000000000000000000000..066501d6588f7d02ced18d627160fd8a3b7da8ab
--- /dev/null
+++ b/src/plug-in/test/pluginTest.cpp
@@ -0,0 +1,40 @@
+#include "../plugininterface.h"
+#include "../plugin.h"
+
+namespace sflphone {
+
+ class PluginTest : public PluginInterface {
+
+ public:
+ PluginTest( const std::string &name ):PluginInterface( name ){
+ }
+
+ virtual int initFunc (void)
+ {
+ return 0;
+ }
+
+ virtual int registerFunc (Plugin **plugin)
+ {
+ Plugin *ret;
+
+ ret = new Plugin(this);
+
+ ret->_name = getInterfaceName();
+ ret->_required = 1;
+ ret->_version_major=1;
+ ret->_version_minor=0;
+
+ *plugin = ret;
+ return 0;
+ }
+ };
+
+}
+extern "C" ::sflphone::PluginInterface* create (void){
+ return new ::sflphone::PluginTest("test");
+}
+
+extern "C" void* destroy( ::sflphone::PluginInterface *p ){
+ delete p;
+}
diff --git a/src/sipaccount.cpp b/src/sipaccount.cpp
index acce1bb66ed5fff56fad4d6a3d20b77c6ba146a0..5880f927f00fd0708ed0a7f30fedea5ef140af24 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 +21,77 @@
#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, useStun;
SIPVoIPLink *thislink;
- _link->setHostname(Manager::instance().getConfigString(_accountID,HOSTNAME));
- useStun = Manager::instance().getConfigInt(_accountID,SIP_USE_STUN);
-
+ /* 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));
+ /* Retrieve STUN stuff */
+ /* STUN configuration is attached to a voiplink because it is applied to every accounts (PJSIP limitation)*/
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));
+ if (thislink) {
+ }
- 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..b716086fbe5fc94fe57146228d20d4605a92169d 100644
--- a/src/sipaccount.h
+++ b/src/sipaccount.h
@@ -26,76 +26,98 @@
#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);
+
+ 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; }
+
+ pjsip_regc* getRegistrationInfo( void ) { return _regc; }
+ void setRegistrationInfo( pjsip_regc *regc ) { _regc = regc; }
+
+ //TODO See if it useful
+ 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 c71fd1da9eb1392f4074c80f29d88a3c6ea40327..a37e68318704433019f1836d553157ab4f0910c0 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
@@ -19,264 +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) {
+ //TODO 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());
- return true;
+ sprintf(contactTmp, "<sip:%s@%s:%d>", username.data(), _localExternAddress.data(), _localExternPort);
+ pj_strdup2(_pool, &contact, contactTmp);
+ account->setContact(contactTmp);
+
+ 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;
+ }
+
+ // 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;
+ removeCall(call->getCallId());
+ return false;
}
-bool
+ bool
SIPVoIPLink::hangup(const CallID& id)
{
- SIPCall* call = getSIPCall(id);
+ 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; }
- if(!Manager::instance().getUserAgent()->hangup(call))
+ // 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; }
+
+ _mutexSIP.enterMutex();
+
+ // Stop sound
+ call->setAudioStart(false);
+ call->setState(Call::Hold);
+ _debug("* SIP Info: Stopping AudioRTP for onhold action\n");
+ _audiortp->closeRtpSession();
+ local_sdp = call->getLocalSDPSession();
+
+ _mutexSIP.leaveMutex();
- // Stop sound
- call->setAudioStart(false);
- call->setState(Call::Hold);
- _debug("* SIP Info: Stopping AudioRTP for onhold action\n");
- _audiortp->closeRtpSession();
+ 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 Manager::instance().getUserAgent()->onhold(call);
+ 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
@@ -287,17 +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;
@@ -305,16 +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()
@@ -323,10 +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);
@@ -341,10 +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)
{
@@ -352,8 +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;
@@ -361,210 +735,1435 @@ 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:
+ // TODO The invite session as global ?
+ 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);
+
+ //TODO 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;
+
+ _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[_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..0aca792fc392fd6ab2a84bea7760c5d6a7870d01 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 5
+
/**
* @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 6e0e104cec4ada90606d980fbf15212e2870e450..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
index b37257aeae33bd46bf9fae99a5a841fc655e013e..728b074cd70eb9fb14d09256cfabeaa36cd653c0 100644
--- a/src/useragent.h
+++ b/src/useragent.h
@@ -56,20 +56,8 @@ private:
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;
@@ -90,9 +78,6 @@ public:
void sipDestory();
- /** Create SIP UDP Listener */
- int createUDPServer();
-
/** Set whether it will use stun server */
void setStunServer(const char *server);
@@ -111,7 +96,6 @@ public:
bool loadSIPLocalIP();
- pj_status_t stunServerResolve();
pjsip_endpoint* getEndPoint() {return _endpt;}
diff --git a/src/voiplink.cpp b/src/voiplink.cpp
index fe62b20467a1047fa2a1965618de1bff4abeea79..e4ff653d48211f0912f639bc8da37a75ada65593 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 +76,3 @@ VoIPLink::clearCallMap()
return true;
}
-void VoIPLink::setRegistrationState(const RegistrationState state)
-{
- _registrationState = state;
- // Notify the client
- Manager::instance().connectionStatusNotification( );
-}
diff --git a/src/voiplink.h b/src/voiplink.h
index e095c1922ccc24af99754eddd5ed79b1d2ebc83a..45f0c0e34e4f25fdd02e6574966f7fce637d6f99 100644
--- a/src/voiplink.h
+++ b/src/voiplink.h
@@ -24,6 +24,7 @@
#define __VOIP_LINK_H__
#include "call.h"
+#include "account.h"
class AudioCodec;
@@ -38,218 +39,187 @@ 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; }
+
+ /**
+ * @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/test/Makefile.am b/test/Makefile.am
index b3aaf8805dc1637d23086fb4406fff4d1eb696aa..5c684888b422490c0e15316692bc4fcee3eceb78 100644
--- a/test/Makefile.am
+++ b/test/Makefile.am
@@ -15,7 +15,6 @@ OBJECT_FILES= \
../src/sflphoned-sipaccount.o \
../src/sflphoned-iaxaccount.o \
../src/sflphoned-eventthread.o \
- ../src/sflphoned-useragent.o \
../src/plug-in/pluginmanager.o \
../src/plug-in/audiorecorder/audiorecord.o \
../src/sflphoned-samplerateconverter.o
@@ -48,7 +47,6 @@ configurationTester_LDADD = \
@DBUSCPP_LIBS@ \
@SAMPLERATE_LIBS@ \
$(PJSIP_LIBS) \
- -luuid \
$(OBJECT_FILES)
pluginmanagerTester_LDADD = \
@@ -63,7 +61,6 @@ pluginmanagerTester_LDADD = \
@DBUSCPP_LIBS@ \
@SAMPLERATE_LIBS@ \
$(PJSIP_LIBS) \
- -luuid \
$(OBJECT_FILES)
audiorecorderTester_LDADD = \
@@ -78,6 +75,5 @@ audiorecorderTester_LDADD = \
@DBUSCPP_LIBS@ \
@SAMPLERATE_LIBS@ \
$(PJSIP_LIBS) \
- -luuid \
$(OBJECT_FILES)
diff --git a/test/pluginmanagerTest.cpp b/test/pluginmanagerTest.cpp
index e5dfff8e1172c7d9afb18342983906c8d675008b..7fd16c28cd0e8c46d185830d1781df2819aea429 100644
--- a/test/pluginmanagerTest.cpp
+++ b/test/pluginmanagerTest.cpp
@@ -26,19 +26,25 @@ using std::cout;
using std::endl;
void PluginManagerTest::setUp(){
- // Instanciate the plugin manager object
_pm = ::sflphone::PluginManager::instance();
}
void PluginManagerTest::testLoadPluginDirectory(){
- _pm->loadPlugins();
+ CPPUNIT_ASSERT(_pm->loadPlugins() == 0);
+}
+
+void PluginManagerTest::testLoadPlugin(){
+ CPPUNIT_ASSERT(_pm->loadPlugins() == 0);
+ //CPPUNIT_ASSERT( _pm->isPluginLoaded("test") == NULL );
}
-void PluginManagerTest::testNonloadedPlugin(){
- CPPUNIT_ASSERT( _pm->isPluginLoaded("test") == NULL );
+void PluginManagerTest::testRegisterPlugin(){
+ // First load the default directory
+ _pm->loadPlugins();
+ // Resolve the symbol
}
void PluginManagerTest::tearDown(){
// Delete the plugin manager object
- delete _pm; _pm=NULL;
+ delete _pm; _pm=0;
}
diff --git a/test/pluginmanagerTest.h b/test/pluginmanagerTest.h
index e43ba8f7ab09262243fddd3edc21130ddca5dd80..0cd92da42ce87ff50b2eb399975a68da9abd2f4f 100644
--- a/test/pluginmanagerTest.h
+++ b/test/pluginmanagerTest.h
@@ -43,7 +43,8 @@ class PluginManagerTest : public CppUnit::TestCase {
*/
CPPUNIT_TEST_SUITE( PluginManagerTest );
CPPUNIT_TEST( testLoadPluginDirectory );
- CPPUNIT_TEST( testNonloadedPlugin );
+ CPPUNIT_TEST( testLoadPlugin );
+ CPPUNIT_TEST( testRegisterPlugin );
CPPUNIT_TEST_SUITE_END();
public:
@@ -63,7 +64,9 @@ class PluginManagerTest : public CppUnit::TestCase {
void testLoadPluginDirectory();
- void testNonloadedPlugin();
+ void testLoadPlugin();
+
+ void testRegisterPlugin();
private:
::sflphone::PluginManager *_pm;