diff --git a/sflphone-common/src/audio/audiortp.cpp b/sflphone-common/src/audio/audiortp.cpp
index 86837fb88a898ef034b86ff6fc8702a5b03b8ad7..16766e710bd3ecb86accf3c95e7ccf6493774978 100644
--- a/sflphone-common/src/audio/audiortp.cpp
+++ b/sflphone-common/src/audio/audiortp.cpp
@@ -52,7 +52,7 @@ AudioRtp::~AudioRtp (void) {
delete _RTXThread; _RTXThread = 0;
}
-int
+void
AudioRtp::createNewSession (SIPCall *ca) {
ost::MutexLock m(_threadMutex);
@@ -70,6 +70,17 @@ AudioRtp::createNewSession (SIPCall *ca) {
// Start RTP Send/Receive threads
_symmetric = Manager::instance().getConfigInt(SIGNALISATION,SYMMETRIC) ? true : false;
_RTXThread = new AudioRtpRTX (ca, _symmetric);
+
+}
+
+int
+AudioRtp::start(void)
+{
+ if(_RTXThread == 0) {
+ _debug("! ARTP Failure: Cannot start audiortp thread since not yet created\n");
+ throw AudioRtpException();
+ }
+
try {
if (_RTXThread->start() != 0) {
_debug("! ARTP Failure: unable to start RTX Thread\n");
@@ -211,7 +222,7 @@ AudioRtpRTX::initAudioRtpSession (void)
try {
- setRtpSessionMedia();
+ // setRtpSessionMedia();
/*
if (_ca == 0) { return; }
@@ -228,7 +239,7 @@ AudioRtpRTX::initAudioRtpSession (void)
_debug("! ARTP Thread Error: Target IP address [%s] is not correct!\n", _ca->getLocalSDP()->get_remote_ip().data());
return;
}
-
+ */
if (!_sym) {
_sessionRecv->setSchedulingTimeout (10000);
@@ -240,9 +251,9 @@ AudioRtpRTX::initAudioRtpSession (void)
_session->setSchedulingTimeout(10000);
_session->setExpireTimeout(1000000);
}
- */
+
- setRtpSessionRemoteIp();
+ // setRtpSessionRemoteIp();
/*
if (!_sym) {
diff --git a/sflphone-common/src/audio/audiortp.h b/sflphone-common/src/audio/audiortp.h
index d31d3bcb7929feb56aba4447440b6bd7b56dc83f..17bfdb6569773dccbe4e51139d88266ee22308b2 100644
--- a/sflphone-common/src/audio/audiortp.h
+++ b/sflphone-common/src/audio/audiortp.h
@@ -35,6 +35,17 @@
#define UP_SAMPLING 0
#define DOWN_SAMPLING 1
+
+
+class AudioRtpException: public std::exception
+{
+ virtual const char* what() const throw()
+ {
+ return "AudioRtpException occured";
+ }
+};
+
+
/**
* @file audiortp.h
* @brief Manage the real-time data transport in a SIP call
@@ -225,7 +236,12 @@ class AudioRtp {
* Create a brand new RTP session by calling the AudioRtpRTX constructor
* @param ca A pointer on a SIP call
*/
- int createNewSession (SIPCall *ca);
+ void createNewSession (SIPCall *ca);
+
+ /**
+ * Start the AudioRtpRTX thread created with createNewSession
+ */
+ int start(void);
/**
* Close a RTP session and kills the remaining threads
diff --git a/sflphone-common/src/sipvoiplink.cpp b/sflphone-common/src/sipvoiplink.cpp
index c44726f772eeacfd490263982de676a2cc1cd846..22988b69573fda62c36036d974891fb9f49a4f69 100644
--- a/sflphone-common/src/sipvoiplink.cpp
+++ b/sflphone-common/src/sipvoiplink.cpp
@@ -506,6 +506,12 @@ SIPVoIPLink::newOutgoingCall ( const CallID& id, const std::string& toUrl )
call->setPeerNumber ( getSipTo ( toUrl, account->getHostname() ) );
setCallAudioLocal ( call, getLocalIPAddress(), useStun(), getStunServer() );
+ try {
+ _audiortp->createNewSession (call);
+ } catch (...) {
+ _debug("Failed to create rtp thread from newOutGoingCall\n");
+ }
+
call->initRecFileName();
_debug ( "Try to make a call to: %s with call ID: %s\n", toUrl.data(), id.data() );
@@ -554,6 +560,13 @@ SIPVoIPLink::answer ( const CallID& id )
}
local_sdp = call->getLocalSDP();
+
+ try {
+ _audiortp->createNewSession (call);
+ } catch (...) {
+ _debug("Failed to create rtp thread from newOutGoingCall\n");
+ }
+
inv_session = call->getInvSession();
status = local_sdp->start_negociation ();
@@ -568,7 +581,8 @@ SIPVoIPLink::answer ( const CallID& id )
// Start the RTP sessions
_debug ( "SIPVoIPLink::answer: Starting AudioRTP when answering : call %s \n", call->getCallId().c_str() );
- if ( _audiortp->createNewSession ( call ) >= 0 )
+ // if ( _audiortp->createNewSession ( call ) >= 0 )
+ if ( _audiortp->start() >= 0 )
{
call->setAudioStart ( true );
call->setConnectionState ( Call::Connected );
@@ -774,6 +788,12 @@ SIPVoIPLink::offhold ( const CallID& id )
return false;
}
+ try {
+ _audiortp->createNewSession (call);
+ } catch (...) {
+ _debug ( "! SIP Failure: Unable to create RTP Session (%s:%d)\n", __FILE__, __LINE__ );
+ }
+
/* Create re-INVITE with new offer */
status = inv_session_reinvite ( call, "sendrecv" );
if ( status != PJ_SUCCESS )
@@ -783,11 +803,15 @@ SIPVoIPLink::offhold ( const CallID& id )
_debug ( "* SIP Info: Starting AudioRTP when offhold\n" );
call->setState ( Call::Active );
// it's sure that this is the current call id...
- if ( _audiortp->createNewSession ( call ) < 0 )
+
+
+ // if ( _audiortp->createNewSession ( call ) < 0 )
+ if ( _audiortp->start() < 0 )
{
_debug ( "! SIP Failure: Unable to start sound (%s:%d)\n", __FILE__, __LINE__ );
return false;
}
+
return true;
}
@@ -1224,6 +1248,8 @@ SIPVoIPLink::SIPCallAnswered ( SIPCall *call, pjsip_rx_data *rdata )
_debug ( "Get remote media information from offer\n" );
call->getLocalSDP()->fetch_media_transport_info_from_remote_sdp ( r_sdp );
+ _audiortp->getRTX()->setRtpSessionRemoteIp();
+
_debug ( "Update call state , id = %s\n", call->getCallId().c_str() );
call->setConnectionState ( Call::Connected );
call->setState ( Call::Active );
@@ -1232,7 +1258,8 @@ SIPVoIPLink::SIPCallAnswered ( SIPCall *call, pjsip_rx_data *rdata )
if ( Manager::instance().isCurrentCall ( call->getCallId() ) )
{
_debug ( "* SIP Info: Starting AudioRTP when answering\n" );
- if ( _audiortp->createNewSession ( call ) < 0 )
+ // if ( _audiortp->createNewSession ( call ) < 0 )
+ if ( _audiortp->start() < 0 )
{
_debug ( "RTP Failure: unable to create new session\n" );
}
@@ -1852,18 +1879,29 @@ void set_voicemail_info ( AccountID account, pjsip_msg_body *body )
void SIPVoIPLink::handle_reinvite ( SIPCall *call )
{
+ /*
_debug("handle_reinvite\n");
-
+
// Close the previous RTP session
_audiortp->closeRtpSession ();
call->setAudioStart ( false );
_debug("create new rtp session from handle_reinvite \n");
// Create a new one with new info
- if ( _audiortp->createNewSession ( call ) >= 0 )
+ // if ( _audiortp->createNewSession ( call ) >= 0 )
+ try {
+ _audiortp->createNewSession (call);
+ } catch (...) {
+ _debug ( "! SIP Failure: Unable to create RTP Session (%s:%d)\n", __FILE__, __LINE__ );
+ }
+ if ( _audiortp->start() >= 0 )
{
call->setAudioStart ( true );
}
+ */
+
+ _audiortp->getRTX()->setRtpSessionRemoteIp();
+ _audiortp->getRTX()->setRtpSessionMedia();
}
@@ -1879,6 +1917,8 @@ void call_on_state_changed ( pjsip_inv_session *inv, pjsip_event *e )
SIPVoIPLink *link;
pjsip_rx_data *rdata;
+ _debug("---------------------- call_on_state_changed --------------------------\n");
+
/* Retrieve the call information */
call = reinterpret_cast<SIPCall*> ( inv->mod_data[_mod_ua.id] );
if ( !call )
@@ -2025,6 +2065,11 @@ void call_on_state_changed ( pjsip_inv_session *inv, pjsip_event *e )
void call_on_media_update ( pjsip_inv_session *inv, pj_status_t status )
{
+ _debug("---------------------- call_on_media_update --------------------------\n");
+
+ AccountID accId;
+ SIPVoIPLink *link;
+
const pjmedia_sdp_session *r_sdp;
SIPCall *call;
@@ -2042,6 +2087,12 @@ void call_on_media_update ( pjsip_inv_session *inv, pj_status_t status )
call->getLocalSDP()->clean_session_media();
// Set the fresh negociated one
call->getLocalSDP()->set_negociated_offer ( r_sdp );
+
+ accId = Manager::instance().getAccountFromCall ( call->getCallId() );
+ link = dynamic_cast<SIPVoIPLink *> ( Manager::instance().getAccountLink ( accId ) );
+
+ link->_audiortp->getRTX()->setRtpSessionMedia();
+
}
void call_on_forked ( pjsip_inv_session *inv, pjsip_event *e )
diff --git a/sflphone-common/test/rtpTest.cpp b/sflphone-common/test/rtpTest.cpp
index 0b164e76833dced5d2d2dce591f477e5f7b21165..a33e1df141517c7ea61c27db2f3944678439e0c0 100644
--- a/sflphone-common/test/rtpTest.cpp
+++ b/sflphone-common/test/rtpTest.cpp
@@ -83,7 +83,7 @@ void RtpTest::testRtpInitClose()
try {
_debug("-------- Open Rtp Session ----------\n");
- CPPUNIT_ASSERT(audiortp->createNewSession(sipcall) == 0);
+ audiortp->createNewSession(sipcall);
} catch(...) {
@@ -125,7 +125,7 @@ void RtpTest::testRtpThread()
_debug("-------- Open Rtp Session ----------\n");
try {
- CPPUNIT_ASSERT(audiortp->createNewSession(sipcall) == 0);
+ audiortp->createNewSession(sipcall);
} catch(...) {
@@ -176,7 +176,7 @@ void RtpTest::testRtpResampling()
_debug("-------- Open Rtp Session ----------\n");
try {
- CPPUNIT_ASSERT(audiortp->createNewSession(sipcall) == 0);
+ audiortp->createNewSession(sipcall);
} catch(...) {