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/account.cpp b/src/account.cpp
index cb2d710a750c9a1da5229683608e6f289f4aece5..b50fc71fc49d5648d875ac8bc4cb99758b82a9d3 100644
--- a/src/account.cpp
+++ b/src/account.cpp
@@ -25,7 +25,7 @@
Account::Account(const AccountID& accountID, std::string type) :
_accountID(accountID), _link(NULL), _enabled(false), _type(type)
{
- setRegistrationState(VoIPLink::Unregistered);
+ setRegistrationState(Unregistered);
}
Account::~Account()
@@ -47,3 +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 a71ffd775e31038c45a4fbead66696a4e4ec866b..764d4628e1c2d780eb18b9ce4cdc23046f150f95 100644
--- a/src/account.h
+++ b/src/account.h
@@ -39,7 +39,7 @@ class VoIPLink;
typedef std::string AccountID;
/** Contains all the state an Voip can be in */
- typedef enum RegistrationState {
+typedef enum RegistrationState {
Unregistered,
Trying,
Registered,
@@ -49,7 +49,7 @@ typedef std::string AccountID;
ErrorHost,
ErrorExistStun,
ErrorConfStun
- } RegistrationState;
+} RegistrationState;
#define AccountNULL ""
@@ -121,13 +121,9 @@ class Account{
*/
inline RegistrationState getRegistrationState() { return _registrationState; }
- inline void setRegistrationState( RegistrationState state ) {
- _registrationState = state;
-
- // Notify the client
- Manager::instance().connectionStatusNotification( );
- }
+ void setRegistrationState( RegistrationState state );
+ //TODO inline?
inline std::string getUsername( void ) { return _username; }
inline void setUsername( std::string username) { _username = username; }
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/global.h b/src/global.h
index e752d41b2f5ee4be78392cd0dd67308e19bf2b93..7a2a1b5793dcdb59213d5f3f684a57c864b40272 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 */
@@ -98,6 +101,7 @@ typedef short int16;
#define WINDOW_POPUP 1 /** Popup mode */
#define NOTIFY_ALL 1 /** Desktop notification level 0: never notify */
#define NOTIFY_MAILS 1 /** Desktop mail notification level 0: never notify */
+#define STUN_ENABLED 1
// Error codes for error handling
#define NO_ERROR 0x0000 /** No error - Everything alright */
diff --git a/src/iaxaccount.cpp b/src/iaxaccount.cpp
index 22672437cb0e2bbaec42f7f7be1390dc4f4eb5ff..4de19c54cc995f4d39954775851299b67860451a 100644
--- a/src/iaxaccount.cpp
+++ b/src/iaxaccount.cpp
@@ -49,7 +49,7 @@ int IAXAccount::registerVoIPLink()
thislink->setPass(Manager::instance().getConfigString(_accountID, PASSWORD));
}
- _link->sendRegister();
+ _link->sendRegister( _accountID );
return SUCCESS;
}
@@ -57,7 +57,7 @@ int 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 035cb3675b7f0629ee857264541f0eac774a9203..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(-1);
+ sendRegister("");
}
// thread wait 3 millisecond
@@ -308,48 +309,52 @@ IAXVoIPLink::getIAXCall(const CallID& id)
int
-IAXVoIPLink::sendRegister(AccountID id UNUSED)
+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();
+
+ // Always use a brand new session
+ if (_regSession) {
+ iax_destroy(_regSession);
+ }
- _regSession = iax_session_new();
+ _regSession = iax_session_new();
- if (!_regSession) {
- _debug("Error when generating new session for register");
- } else {
- // refresh
- // last reg
- char host[_host.length()+1];
- strcpy(host, _host.c_str());
- char user[_user.length()+1];
- strcpy(user, _user.c_str());
- char pass[_pass.length()+1];
- strcpy(pass, _pass.c_str());
- // iax_register doesn't use const char*
-
- _debug("IAX Sending registration to %s with user %s\n", host, user);
- int val = iax_register(_regSession, host, user, pass, 120);
- _debug ("Return value: %d\n", val);
- // set the time-out to 15 seconds, after that, resend a registration request.
- // until we unregister.
- _nextRefreshStamp = time(NULL) + 10;
- result = true;
-
- setRegistrationState(Trying);
+ 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(AccountID id UNUSED)
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);
}
}
diff --git a/src/iaxvoiplink.h b/src/iaxvoiplink.h
index bae6df69f55ca1e3dfd4fca245f14485c5a7ae3e..8ad4082882f41b731a610714172167bb94e15f95 100644
--- a/src/iaxvoiplink.h
+++ b/src/iaxvoiplink.h
@@ -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 619d494894b9ee5376732405a537ceb4ec938f1d..c8c25d2b8432821c06f587c7b17ae0d3b38fa651 100644
--- a/src/managerimpl.cpp
+++ b/src/managerimpl.cpp
@@ -192,27 +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
@@ -971,15 +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);
@@ -1361,7 +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 );
@@ -1833,7 +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);
@@ -1842,15 +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")))))))))
)
);
@@ -1875,6 +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;
@@ -1888,16 +1915,25 @@ 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((*details.find(SIP_USE_STUN)).second == "TRUE")
{
- //TODO Replace: _userAgent->setStunServer((*details.find(SIP_STUN_SERVER)).second.data());
+ link->setStunServer((*details.find(SIP_STUN_SERVER)).second.data());
}
else
{
- //TODO: replace: _userAgent->setStunServer(NULL);
+ link->setStunServer("");
}
//restartPjsip();
}
@@ -1916,6 +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
diff --git a/src/managerimpl.h b/src/managerimpl.h
index a556b9175cdda6677dc9624757a27d0cd8e27589..222c7154788fd86d4cc79d2fd3ad16c048584f82 100644
--- a/src/managerimpl.h
+++ b/src/managerimpl.h
@@ -1020,6 +1020,11 @@ public:
void setSipPort( int port );
+ std::string getStunServer (void);
+ void setStunServer (const std::string &server);
+
+ int isStunEnabled (void);
+ void enableStun (void);
private:
// Copy Constructor
diff --git a/src/sipaccount.cpp b/src/sipaccount.cpp
index 5270e39df7a3aea627dd51bfac68e1b43e55f225..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
@@ -28,18 +26,28 @@ SIPAccount::SIPAccount(const AccountID& accountID)
: Account(accountID, "sip")
, _cred(NULL)
, _contact("")
+ , _bRegister(false)
+ , _regc()
{
- _link = new SIPVoIPLink(accountID);
- //_link = SIPVoIPLink::instance( 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()
@@ -56,12 +64,7 @@ int SIPAccount::registerVoIPLink()
/* STUN configuration is attached to a voiplink because it is applied to every accounts (PJSIP limitation)*/
thislink = dynamic_cast<SIPVoIPLink*> (_link);
if (thislink) {
- useStun = Manager::instance().getConfigInt(_accountID,SIP_USE_STUN);
- thislink->setStunServer(Manager::instance().getConfigString(_accountID,SIP_STUN_SERVER));
- thislink->setUseStun( useStun!=0 ? true : false);
}
- /* Link initialization */
- _link->init();
/* Start registration */
status = _link->sendRegister( _accountID );
@@ -73,7 +76,7 @@ int SIPAccount::registerVoIPLink()
int SIPAccount::unregisterVoIPLink()
{
_debug("SIPAccount: unregister account %s\n" , getAccountID().c_str());
- return _link->sendUnregister();
+ return _link->sendUnregister( _accountID );
}
void SIPAccount::loadConfig()
diff --git a/src/sipaccount.h b/src/sipaccount.h
index dfbc55774f4a9fcb19327ebdbb303f0ca0b97bc2..b716086fbe5fc94fe57146228d20d4605a92169d 100644
--- a/src/sipaccount.h
+++ b/src/sipaccount.h
@@ -26,8 +26,6 @@
#include "account.h"
#include "sipvoiplink.h"
-struct pjsip_cred_info;
-
class SIPVoIPLink;
/**
@@ -91,13 +89,31 @@ class SIPAccount : public Account
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
*/
diff --git a/src/sipcall.cpp b/src/sipcall.cpp
index bcc596ccf7f1f3a1b33a9befc4fccc2bc03167ce..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
@@ -68,11 +66,6 @@ SIPCall::SIPCallInvite(pjsip_rx_data *rdata, pj_pool_t *pool)
_localSDP->time.start = _localSDP->time.stop = 0;
sdpAddMediaDescription(pool);
- _debug("SDP: addr: %s\nuser: %s\nid: %i\nversion: %i\naddr: %s\nattr count: %i\n",
- _ipAddr.c_str(), _localSDP->origin.user, _localSDP->origin.id, _localSDP->origin.version, _localSDP->origin.addr.ptr, _localSDP->attr_count );
-
- _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/sipvoiplink.cpp b/src/sipvoiplink.cpp
index 42e9887e837d97ce43845f29bef36d9fa0f61958..a37e68318704433019f1836d553157ab4f0910c0 100644
--- a/src/sipvoiplink.cpp
+++ b/src/sipvoiplink.cpp
@@ -18,29 +18,45 @@
* 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 "audio/audiortp.h"
-#include "manager.h"
-#include "user_cfg.h" // SIGNALISATION / PULSE #define
+/**************** EXTERN VARIABLES AND FUNCTIONS (callbacks) **************************/
-/** PJSIP related variables */
+/*
+ * The global pool factory
+ */
pj_caching_pool _cp;
+
+/*
+ * The pool to allocate memory
+ */
pj_pool_t *_pool;
+
+/*
+ * The SIP endpoint
+ */
pjsip_endpoint *_endpt;
-pjsip_module _mod_ua; /** PJSIP module. */
+
+/*
+ * 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 );
/**
@@ -49,84 +65,146 @@ void set_voicemail_info( AccountID account, pjsip_msg_body *body );
* @param call a SIPCall valid pointer
* @return bool True
*/
-bool setCallAudioLocal(SIPCall* call);
-
-
-/** Do we use stun? */
-bool _useStun;
+bool setCallAudioLocal(SIPCall* call, std::string localIP, bool stun, std::string server);
-/** The current STUN server address */
-std::string _stunServer;
-
-std::string _localIPAddress;
+// Documentated from the PJSIP Developer's Guide, available on the pjsip website/
-/** PJSIP callbacks */
+/*
+ * 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("")
+ , _nbTryListenAddr(2) // number of times to try to start SIP listener
+ , _stunServer("")
, _localExternAddress("")
, _localExternPort(0)
, _audiortp(new AudioRtp())
- , _regc()
- , _bRegister(false)
,_regPort(DEFAULT_SIP_PORT)
+ , _useStun(false)
+ , _clients(0)
{
// to get random number for RANDOM_PORT
srand (time(NULL));
- _useStun = false;
- _localIPAddress = "127.0.0.1";
+ /* Instanciate the C++ thread */
_evThread = new EventThread(this);
+
+ /* Start pjsip initialization step */
+ init();
}
SIPVoIPLink::~SIPVoIPLink()
{
- delete _evThread; _evThread = NULL;
terminate();
}
-SIPVoIPLink* SIPVoIPLink::instance( const AccountID& id){
- /*if(!_instance ){
- _instance = new SIPVoIPLink( id );
- }
- return _instance;*/
+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()
{
- if(_initDone)
+ if(initDone())
return false;
/* Initialize the pjsip library */
- _regc = NULL;
- if(!pjsip_init())
- return false;
+ pjsip_init();
+ initDone(true);
- _initDone = true;
return true;
}
void
SIPVoIPLink::terminate()
{
+ delete _evThread; _evThread = NULL;
+
/* Clean shutdown of pjsip library */
- if( _initDone )
+ if( initDone() )
+ {
pjsip_shutdown();
- _initDone = false;
+ }
+ initDone(false);
}
void
@@ -152,17 +230,11 @@ SIPVoIPLink::getEvent()
{
// 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 );
- }
-
- _mutexSIP.enterMutex();
// PJSIP polling
pj_time_val timeout = {0, 10};
pjsip_endpt_handle_events( _endpt, &timeout);
-
- _mutexSIP.leaveMutex();
}
int SIPVoIPLink::sendRegister( AccountID id )
@@ -174,6 +246,7 @@ int SIPVoIPLink::sendRegister( AccountID id )
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();
@@ -182,30 +255,32 @@ int SIPVoIPLink::sendRegister( AccountID id )
_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;
+ if(regc) {
+ status = pjsip_regc_destroy(regc);
+ regc = NULL;
PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
}
- setRegister(true);
+ account->setRegister(true);
/* Set the expire value of the message from the config file */
expire_value = Manager::instance().getRegistrationExpireValue();
/* Update the state of the voip link */
- setRegistrationState(Trying);
+ account->setRegistrationState(Trying);
if (!validStunServer) {
- setRegistrationState(VoIPLink::ErrorExistStun);
- setRegister(false);
+ account->setRegistrationState(ErrorExistStun);
+ account->setRegister(false);
_mutexSIP.leaveMutex();
return false;
}
/* Create the registration according to the account ID */
- status = pjsip_regc_create(_endpt, (void*)new AccountID(id), ®c_cb, &_regc);
+ status = pjsip_regc_create(_endpt, (void*)account, ®c_cb, ®c);
if (status != PJ_SUCCESS) {
_debug("UserAgent: Unable to create regc.\n");
_mutexSIP.leaveMutex();
@@ -222,7 +297,7 @@ int SIPVoIPLink::sendRegister( AccountID id )
pj_strdup2(_pool, &contact, contactTmp);
account->setContact(contactTmp);
- status = pjsip_regc_init(_regc, &svr, &aor, &aor, 1, &contact, 600); //timeout);
+ 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();
@@ -240,48 +315,55 @@ int SIPVoIPLink::sendRegister( AccountID id )
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);
+ pjsip_regc_set_credentials(regc, 1, cred);
account->setCredInfo(cred);
- status = pjsip_regc_register(_regc, PJ_TRUE, &tdata);
+ 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);
+ 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()
+SIPVoIPLink::sendUnregister( AccountID id )
{
pj_status_t status = 0;
pjsip_tx_data *tdata = NULL;
+ SIPAccount *account;
+ pjsip_regc *regc;
- if(!isRegister()){
- setRegistrationState(VoIPLink::Unregistered);
+ 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(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 );
+ status = pjsip_regc_send( regc, tdata );
if(status != PJ_SUCCESS) {
_debug("UserAgent: Unable to send regc request.\n");
return false;
@@ -291,7 +373,8 @@ SIPVoIPLink::sendUnregister()
return false;
}
- setRegister(false);
+ account->setRegistrationInfo(regc);
+ account->setRegister(false);
return true;
}
@@ -302,9 +385,17 @@ SIPVoIPLink::newOutgoingCall(const CallID& id, const std::string& toUrl)
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());
@@ -426,12 +517,17 @@ SIPVoIPLink::onhold(const CallID& 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();
+
if( local_sdp == NULL ){
_debug("! SIP Failure: unable to find local_sdp\n");
return false;
@@ -703,7 +799,7 @@ SIPVoIPLink::SIPStartCall(SIPCall* call, const std::string& subject UNUSED)
id = Manager::instance().getAccountFromCall(call->getCallId());
// Get the basic information about the callee account
account = dynamic_cast<SIPAccount *>(Manager::instance().getAccount(id));
-
+
strTo = getSipTo(call->getPeerNumber(), account->getHostname());
_debug(" To: %s\n", strTo.data());
@@ -723,7 +819,7 @@ SIPVoIPLink::SIPStartCall(SIPCall* call, const std::string& subject UNUSED)
&dialog);
PJ_ASSERT_RETURN(status == PJ_SUCCESS, false);
- setCallAudioLocal(call);
+ setCallAudioLocal(call, getLocalIPAddress(), useStun(), getStunServer());
call->setIp(getLocalIP());
// Building the local SDP offer
@@ -767,14 +863,14 @@ std::string SIPVoIPLink::getSipTo(const std::string& to_url, std::string hostnam
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)
@@ -782,27 +878,27 @@ std::string SIPVoIPLink::SIPToHeader(const std::string& to)
return true;
}
-bool 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();
+ 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);
+ _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);
+ // Set local audio port for SIPCall(id)
+ call->setLocalIp(localIP);
+ call->setLocalAudioPort(callLocalAudioPort);
+ call->setLocalExternAudioPort(callLocalExternAudioPort);
- return true;
- }
+ return true;
+ }
void
SIPVoIPLink::SIPCallServerFailure(SIPCall *call)
@@ -901,18 +997,20 @@ bool setCallAudioLocal(SIPCall* call)
return NULL;
}
- ///////////////////////////////////////////////////////////////////////////////
- // Private functions
- ///////////////////////////////////////////////////////////////////////////////
-
- pj_str_t SIPVoIPLink::string2PJStr(const std::string &value)
+ void SIPVoIPLink::setStunServer( const std::string &server )
{
- char tmp[256];
-
- strcpy(tmp, value.data());
- return pj_str(tmp);
+ if(server != "") {
+ useStun(true);
+ _stunServer = server;
+ } else {
+ useStun(false);
+ _stunServer = std::string("");
+ }
}
+ ///////////////////////////////////////////////////////////////////////////////
+ // Private functions
+ ///////////////////////////////////////////////////////////////////////////////
bool SIPVoIPLink::pjsip_init()
{
@@ -922,7 +1020,7 @@ bool setCallAudioLocal(SIPCall* call)
pjsip_inv_callback inv_cb;
pj_str_t accepted;
std::string name_mod;
-
+ bool useStun;
validStunServer = true;
name_mod = "sflphone";
@@ -968,9 +1066,14 @@ bool setCallAudioLocal(SIPCall* call)
port = _regPort;
- if (_useStun && !Manager::instance().behindNat(_stunServer, port)) {
+ /* 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(_stunServer, 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...
@@ -978,7 +1081,7 @@ bool setCallAudioLocal(SIPCall* call)
}
_localPort = port;
- if (_useStun) {
+ if (useStun) {
// set by last behindNat() call (ish)...
stunServerResolve();
_localExternAddress = Manager::instance().getFirewallAddress();
@@ -1037,17 +1140,6 @@ bool setCallAudioLocal(SIPCall* call)
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 );
@@ -1096,6 +1188,9 @@ bool setCallAudioLocal(SIPCall* call)
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));
@@ -1103,13 +1198,13 @@ bool setCallAudioLocal(SIPCall* call)
stun_status = PJ_EPENDING;
// Init STUN socket
- pos = _stunServer.find(':');
+ pos = stun_server.find(':');
if(pos == std::string::npos) {
- pj_strdup2(_pool, &stun_adr, _stunServer.data());
+ 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 = _stunServer.substr(0, pos);
- serverPort = _stunServer.substr(pos + 1);
+ 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);
@@ -1219,120 +1314,110 @@ bool setCallAudioLocal(SIPCall* call)
return _mod_ua.id;
}
-void set_voicemail_info( AccountID account, pjsip_msg_body *body ){
+ void 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;
+ 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
+ _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 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);
+ // 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 {
+ // 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;
+ 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);
}
- // We need now to notify the manager
- if( voicemail != 0 )
- Manager::instance().startVoiceMessageNotification(account, voicemail);
-}
+ /*******************************/
+ /* CALLBACKS IMPLEMENTATION */
+ /*******************************/
-bool SIPVoIPLink::useStun( void ){
- return _useStun;
-}
+ void call_on_state_changed( pjsip_inv_session *inv, pjsip_event *e){
-void SIPVoIPLink::setUseStun( bool use )
-{
- _useStun = use;
-}
+ PJ_UNUSED_ARG(inv);
+ SIPCall *call = reinterpret_cast<SIPCall*> (inv->mod_data[_mod_ua.id]);
+ if(!call)
+ return;
- /*******************************/
- /* CALLBACKS IMPLEMENTATION */
- /*******************************/
+ /* 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;
-void call_on_state_changed( pjsip_inv_session *inv, pjsip_event *e){
-
- PJ_UNUSED_ARG(inv);
+ 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;
- SIPCall *call = reinterpret_cast<SIPCall*> (inv->mod_data[_mod_ua.id]);
- if(!call)
- return;
+ 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;
- /* 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;
- }
+ case PJSIP_INV_STATE_DISCONNECTED:
+ st_code = e->body.tsx_state.tsx->status_code;
+ ev_state = PJSIP_EVSUB_STATE_TERMINATED;
+ break;
- if (st_code != -1) {
- pjsip_tx_data *tdata;
- pj_status_t status;
+ case PJSIP_INV_STATE_INCOMING:
+ /* Nothing to do. Just to keep gcc from complaining about
+ * unused enums.
+ */
+ break;
+ }
- 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 (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 send NOTIFY -- %d\n", status);
+ _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");
@@ -1342,136 +1427,136 @@ void call_on_state_changed( 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){
+ 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;
+ 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);
+ _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;
- }
+ 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;
+ //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;
- 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");
+ 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;
+ call->setConnectionState(Call::Ringing);
+ Manager::instance().peerRingingCall(call->getCallId());
}
- accId = Manager::instance().getAccountFromCall(call->getCallId());
- link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink(accId));
- if (link) {
- link->SIPCallServerFailure(call);
+ 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
+ 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;
+ } 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);
}
-
- accId = Manager::instance().getAccountFromCall(call->getCallId());
- link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink(accId));
- if (link) {
- link->SIPCallClosed(call);
+ 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
+ break;
+ default:
+ break;
+ } // end of switch
+ }
}
-}
void regc_cb(struct pjsip_regc_cbparam *param){
- AccountID *id = static_cast<AccountID *> (param->token);
- SIPVoIPLink *voipLink;
+ //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);
- voipLink = dynamic_cast<SIPVoIPLink *>(Manager::instance().getAccountLink(*id));
- if(!voipLink)
+ account = static_cast<SIPAccount *>(param->token);
+ if(!account)
return;
if (param->status == PJ_SUCCESS) {
@@ -1483,584 +1568,602 @@ void call_on_tsx_changed(pjsip_inv_session *inv, pjsip_transaction *tsx, pjsip_e
switch(param->code) {
case 408:
case 606:
- voipLink->setRegistrationState(VoIPLink::ErrorConfStun);
+ account->setRegistrationState(ErrorConfStun);
break;
case 503:
- voipLink->setRegistrationState(VoIPLink::ErrorHost);
+ account->setRegistrationState(ErrorHost);
break;
case 401:
case 403:
case 404:
- voipLink->setRegistrationState(VoIPLink::ErrorAuth);
+ account->setRegistrationState(ErrorAuth);
break;
default:
- voipLink->setRegistrationState(VoIPLink::Error);
+ account->setRegistrationState(Error);
break;
}
- voipLink->setRegister(false);
+ account->setRegister(false);
} else {
// Registration/Unregistration is success
- if(voipLink->isRegister())
- voipLink->setRegistrationState(VoIPLink::Registered);
+ if(account->isRegister())
+ account->setRegistrationState(Registered);
else {
- voipLink->setRegistrationState(VoIPLink::Unregistered);
- voipLink->setRegister(false);
+ account->setRegistrationState(Unregistered);
+ account->setRegister(false);
}
}
} else {
- voipLink->setRegistrationState(VoIPLink::ErrorAuth);
- voipLink->setRegister(false);
+ account->setRegistrationState(ErrorAuth);
+ account->setRegister(false);
}
}
- pj_bool_t mod_on_rx_request(pjsip_rx_data *rdata){
+ 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;
- //pjmedia_sdp_session *r_sdp;
- AccountID account_id;
- pjsip_uri *uri;
- pjsip_sip_uri *sip_uri;
- std::string userName, server, caller, callerServer, peerNumber;
-
- // 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(account_id == AccountNULL) {
- _debug("UserAgent: Username %s doesn't match any account!\n",userName.c_str());
- 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());
+ 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;
+ }
- /* 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);
+ _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());
- caller = sip_uri->user.ptr;
- callerServer = sip_uri->host.ptr;
- peerNumber = caller + "@" + callerServer;
+ /* 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);
- // 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 );
+ /* 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;
+ }
}
- 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 ");
+ // 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;
}
- }
- // 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!
- 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 false;
- }
-
- // Set the codec map, IP, peer number and so on... for the SIPCall object
- setCallAudioLocal(call);
- call->setCodecMap(Manager::instance().getCodecDescriptorMap());
- call->setConnectionState(Call::Progressing);
- call->setIp("127.0.0.1");
- 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)) {
+ // 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;
+ }
- // 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");
+ // 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 notify UI
+ // Fail to collect call information
delete call;
call = NULL;
- _debug("UserAgent: Fail to notify UI!\n");
+ _debug("UserAgent: Call SIPCallInvite failed!\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;
- }
+ /* 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);
+ // 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;
+ // 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);
+ // 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);
+ // Associate invite session to the current call
+ call->setInvSession(inv);
- // Update the connection state
- call->setConnectionState(Call::Ringing);
+ // Update the connection state
+ call->setConnectionState(Call::Ringing);
- /* Done */
- return true;
+ /* 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.
- */
+ void onCallTransfered(pjsip_inv_session *inv, pjsip_rx_data *rdata)
+ {
+ pj_status_t status;
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);
+ 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;
}
- /* 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);
-
+ /* 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);
- /* 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;
+ if (refer_sub) {
+ if (!pj_strnicmp2(&refer_sub->hvalue, "true", 4)==0)
+ no_refer_sub = PJ_TRUE;
}
- /* Don't have subscription */
- sub = NULL;
-
- } else {
- struct pjsip_evsub_user xfer_cb;
- pjsip_hdr hdr_list;
+ /* 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);
- /* Init callback */
- pj_bzero(&xfer_cb, sizeof(xfer_cb));
- xfer_cb.on_evsub_state = &xfer_svr_cb;
+ /* Notify callback */
+ code = PJSIP_SC_ACCEPTED;
- /* Init addiTHIS_FILE, THIS_FILE, tional header list to be sent with REFER response */
- pj_list_init(&hdr_list);
+ _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);
- /* 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 };
+ 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(inv->dlg->pool,
- &str_refer_sub,
- &str_true);
- pj_list_push_back(&hdr_list, hdr);
+ pjsip_generic_string_hdr_create(tdata->pool, &str_refer_sub,
+ &str_false);
+ pjsip_msg_add_hdr(tdata->msg, hdr);
- }
- /* Accept the REFER request, send 2xx. */
- pjsip_xfer_accept(sub, rdata, code, &hdr_list);
+ /* 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;
+ }
- /* 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;
- }
+ /* Don't have subscription */
+ sub = NULL;
- /* 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;
- }
- }
+ } else {
+ struct pjsip_evsub_user xfer_cb;
+ pjsip_hdr hdr_list;
- /* 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';
+ /* Init callback */
+ pj_bzero(&xfer_cb, sizeof(xfer_cb));
+ xfer_cb.on_evsub_state = &xfer_svr_cb;
- /* Now make the outgoing call. */
- tmp = std::string(uri);
+ /* Init addiTHIS_FILE, THIS_FILE, tional header list to be sent with REFER response */
+ pj_list_init(&hdr_list);
- if(existing_call == NULL) {
- _debug("UserAgent: Call doesn't exist!\n");
- return;
- }
+ /* 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;
+ }
- AccountID accId = Manager::instance().getAccountFromCall(existing_call->getCallId());
- CallID newCallId = Manager::instance().getNewCallID();
+ /* 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;
- if(!Manager::instance().outgoingCall(accId, newCallId, tmp)) {
+ hdr = (pjsip_hdr*)
+ pjsip_generic_string_hdr_create(inv->dlg->pool,
+ &str_refer_sub,
+ &str_true);
+ pj_list_push_back(&hdr_list, hdr);
- /* Notify xferer about the error (if we have subscription) */
- if (sub) {
- status = pjsip_xfer_notify(sub, PJSIP_EVSUB_STATE_TERMINATED,
- 500, NULL, &tdata);
+ }
+
+ /* 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\n", status);
+ _debug("UserAgent: Unable to create NOTIFY to REFER -- %d", status);
return;
}
- status = pjsip_xfer_send_request(sub, tdata);
+
+ /* 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;
}
}
- 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");
+ /* 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;
}
- }
- 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);
- }
-}
+ 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;
+ }
+ }
-void xfer_func_cb( pjsip_evsub *sub, pjsip_event *event){
+ 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);
- PJ_UNUSED_ARG(event);
+ /* Put the invite_data in the subscription. */
+ pjsip_evsub_set_mod_data(sub, _mod_ua.id,
+ newCall);
+ }
+ }
- _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;
+ void xfer_func_cb( pjsip_evsub *sub, pjsip_event *event){
- /* Find Refer-Sub header */
- refer_sub = (pjsip_generic_string_hdr*)
- pjsip_msg_find_hdr_by_name(rdata->msg_info.msg,
- &REFER_SUB, NULL);
+ PJ_UNUSED_ARG(event);
- /* 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();
- }
+ _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) {
- /* Yes, subscription is suppressed.
- * Terminate our subscription now.
- */
- _debug("UserAgent: Xfer subscription suppressed, terminating event subcription...\n");
- pjsip_evsub_terminate(sub, PJ_TRUE);
+ pjsip_rx_data *rdata;
+ pjsip_generic_string_hdr *refer_sub;
+ const pj_str_t REFER_SUB = {(char*)"Refer-Sub", 9 };
- } 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));
- 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;
- /* 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");
+ /* 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();
+ }
- if (!link || !event) {
- /* Application is not interested with call progress status */
- _debug("UserAgent: Either link or event is empty!\n");
- return;
- }
+ /* Yes, subscription is suppressed.
+ * Terminate our subscription now.
+ */
+ _debug("UserAgent: Xfer subscription suppressed, terminating event subcription...\n");
+ pjsip_evsub_terminate(sub, PJ_TRUE);
- // Get current call
- SIPCall *call = dynamic_cast<SIPCall *>(link->getCall(Manager::instance().getCurrentCallId()));
- if(!call) {
- _debug("UserAgent: Call doesn't exit!\n");
- return;
+ } else {
+ /* Notify application about call transfer progress.
+ * Initially notify with 100/Accepted status.
+ */
+ _debug("UserAgent: Xfer subscription 100/Accepted received...\n");
+ }
}
-
- /* This better be a NOTIFY request */
- if (event->type == PJSIP_EVENT_TSX_STATE &&
- event->body.tsx_state.type == PJSIP_EVENT_RX_MSG)
+ /*
+ * 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_rx_data *rdata;
+ 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;
- rdata = event->body.tsx_state.src.rdata;
+ 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");
- /* 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");
+ if (!link || !event) {
+ /* Application is not interested with call progress status */
+ _debug("UserAgent: Either link or event is empty!\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");
+ // Get current call
+ SIPCall *call = dynamic_cast<SIPCall *>(link->getCall(Manager::instance().getCurrentCallId()));
+ if(!call) {
+ _debug("UserAgent: Call doesn't exit!\n");
return;
}
- } else {
- _debug("UserAgent: Set code to 500!\n");
- status_line.code = 500;
- status_line.reason = *pjsip_get_status_text(500);
- }
+ /* 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;
- /* Notify application */
- is_last = (pjsip_evsub_get_state(sub)==PJSIP_EVSUB_STATE_TERMINATED);
- cont = !is_last;
+ rdata = event->body.tsx_state.src.rdata;
- if(status_line.code/100 == 2) {
- _debug("UserAgent: Try to stop rtp!\n");
- pjsip_tx_data *tdata;
+ /* 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;
+ }
- 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");
+ _debug("UserAgent: Set code to 500!\n");
+ status_line.code = 500;
+ status_line.reason = *pjsip_get_status_text(500);
}
- link->transferStep2();
- cont = PJ_FALSE;
- }
+ /* 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");
+ }
- if (!cont) {
- pjsip_evsub_set_mod_data(sub, _mod_ua.id, NULL);
+ 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);
+ 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;
+ /*
+ * 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;
+ 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);
+ pjsip_evsub_set_mod_data(sub, _mod_ua.id, NULL);
+ call->setXferSub(NULL);
- _debug("UserAgent: Xfer server subscription terminated\n");
- }
-}
+ _debug("UserAgent: Xfer server subscription terminated\n");
+ }
+ }
diff --git a/src/sipvoiplink.h b/src/sipvoiplink.h
index cf67cd7ca47c3692bf3f557646a9ea5e3efd3e8d..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
@@ -24,14 +24,15 @@
#include "voiplink.h"
+//////////////////////////////
+/* PJSIP imports */
#include <pjsip.h>
-#include <pjlib-util.h>
#include <pjlib.h>
-#include <pjnath/stun_config.h>
-#include <pjsip_simple.h>
#include <pjsip_ua.h>
-#include <pjmedia/sdp.h>
-#include <pjmedia/sdp_neg.h>
+#include <pjlib-util.h>
+#include <pjnath/stun_config.h>
+///////////////////////////////
+
class EventThread;
class SIPCall;
@@ -39,11 +40,15 @@ 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
@@ -51,11 +56,9 @@ class SIPVoIPLink : public VoIPLink
public:
/**
- * Constructor
- * @param accountID The account identifier
+ * Singleton method. Enable to retrieve the unique static instance
+ * @return SIPVoIPLink* A pointer on the object
*/
- SIPVoIPLink(const AccountID& accountID);
-
static SIPVoIPLink* instance( const AccountID& id );
/**
@@ -75,6 +78,9 @@ class SIPVoIPLink : public VoIPLink
*/
bool init(void);
+ /**
+ * Shut the library and clean up
+ */
void terminate( void );
/**
@@ -94,7 +100,7 @@ class SIPVoIPLink : public VoIPLink
* @return bool True on success
* false otherwise
*/
- int sendUnregister(void);
+ int sendUnregister(AccountID id);
/**
* Place a new call
@@ -169,19 +175,17 @@ class SIPVoIPLink : public VoIPLink
* If set to true, we check for a firewall
* @param use true if we use STUN
*/
- void setUseStun(bool use);
+ inline void useStun(bool use) { _useStun=use; }
- bool useStun( void );
+ inline bool useStun( void ) { return _useStun; }
/**
* The name of the STUN server
* @param server Server FQDN/IP
*/
- void setStunServer(const std::string& server) { _stunServer = server; }
+ void setStunServer(const std::string& server);
- bool isRegister() {return _bRegister;}
-
- void setRegister(bool result) {_bRegister = result;}
+ std::string getStunServer (void) { return _stunServer; }
/**
* Terminate every call not hangup | brutal | Protected by mutex
@@ -264,12 +268,24 @@ class SIPVoIPLink : public VoIPLink
/** Starting sound */
AudioRtp* _audiortp;
- pj_str_t string2PJStr(const std::string &value);
+ /** 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
@@ -283,6 +299,9 @@ class SIPVoIPLink : public VoIPLink
*/
bool pjsip_shutdown(void);
+ /** Do we use stun? */
+ bool _useStun;
+
pj_status_t stunServerResolve();
/** Create SIP UDP Listener */
@@ -292,8 +311,6 @@ class SIPVoIPLink : public VoIPLink
std::string getLocalIP() {return _localExternAddress;}
- pjsip_regc *_regc;
-
/** For registration use only */
int _regPort;
@@ -313,7 +330,9 @@ class SIPVoIPLink : public VoIPLink
EventThread* _evThread;
ost::Mutex _mutexSIP;
- bool _bRegister;
+ /* 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 59a370dfab70abe8b6f6f911cdbad76fd6da3948..0000000000000000000000000000000000000000
--- a/src/useragent.cpp
+++ /dev/null
@@ -1,906 +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;
- }
-
-
-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
-}
-
-
-
-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);
-}
-
-void UserAgent::setStunServer(const char *server) {
- if(server != NULL) {
- _useStun = true;
- _stunServer = std::string(server);
- } else {
- _useStun = false;
- _stunServer = std::string("");
- }
-}
-
-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;
-}
-
-
-
-
-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);
- }
- }
- }
- }
-
-}
-
-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, _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/voiplink.cpp b/src/voiplink.cpp
index 46b69582329a1302a77a1c705d3713df98ba0ec8..e4ff653d48211f0912f639bc8da37a75ada65593 100644
--- a/src/voiplink.cpp
+++ b/src/voiplink.cpp
@@ -76,10 +76,3 @@ bool VoIPLink::clearCallMap()
return true;
}
-void VoIPLink::setRegistrationState(AccountID id, const RegistrationState state)
-{
- Account *account;
-
- account = dynamic_cast<SIPAccount *>(Manager::instance().getAccount(id));
- account->setRegistrationState( state );
-}
diff --git a/src/voiplink.h b/src/voiplink.h
index 2261211edc5ad86a1e0ca65f5251478a63dd92cf..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;
@@ -38,197 +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);
-
-
- /**
- * 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 (void) = 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;
-
- /** 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);
-
- /**
- * Get an account connection status
- * @return Connection status
- */
- RegistrationState getRegistrationState( void );
-
- /**
- * Set new registration state
- * @param state The registration state
- */
- void setRegistrationState(AccountID id, const RegistrationState state);
-
- private:
- /**
- * ID of parent's Account
- */
- AccountID _accountID;
-
- /**
- * State of registration
- */
- RegistrationState _registrationState;
-
-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;
+ /**
+ * 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__