diff --git a/sflphone-common/src/audio/audiortp.cpp b/sflphone-common/src/audio/audiortp.cpp
index 2335e5ce879bd5207836f43e7e97b36fa2bf1ead..3493159f72ba943985a230bd894df5794ce8df3b 100644
--- a/sflphone-common/src/audio/audiortp.cpp
+++ b/sflphone-common/src/audio/audiortp.cpp
@@ -154,6 +154,7 @@ AudioRtpRTX::AudioRtpRTX (SIPCall *sipcall, bool sym) : time(new ost::Time()), _
_debug ("%i\n", _ca->getLocalAudioPort());
_session = new ost::SymmetricRTPSession (local_ip, _ca->getLocalAudioPort());
+ // _session = new ost::RTPSessionBase(local_ip, _ca->getLocalAudioPort());
_sessionRecv = NULL;
_sessionSend = NULL;
@@ -195,6 +196,8 @@ AudioRtpRTX::~AudioRtpRTX () {
delete converter; converter = NULL;
+ // _session->terminate();
+
delete _session; _session = NULL;
_debug("AudioRtpRTX instance deleted\n");
@@ -222,9 +225,9 @@ AudioRtpRTX::initBuffers()
AudioRtpRTX::initAudioRtpSession (void)
{
- try {
+ try {
- _session->setSchedulingTimeout(10000);
+ _session->setSchedulingTimeout(100000);
_session->setExpireTimeout(1000000);
@@ -252,10 +255,8 @@ AudioRtpRTX::setRtpSessionMedia(void)
_codecSampleRate = _audiocodec->getClockRate();
_codecFrameSize = _audiocodec->getFrameSize();
-
- if (_audiocodec->getPayload() == 9){
- _debug("WE ARE G722\n");
- _payloadIsSet = _session->setPayloadFormat(ost::DynamicPayloadFormat((ost::StaticPayloadType) _audiocodec->getPayload(), 16000));
+ if( _audiocodec->getPayload() == 9 ) {
+ _payloadIsSet = _session->setPayloadFormat(ost::DynamicPayloadFormat((ost::PayloadType) _audiocodec->getPayload(), _audiocodec->getClockRate()));
}
else if ( _audiocodec->hasDynamicPayload() ) {
_payloadIsSet = _session->setPayloadFormat(ost::DynamicPayloadFormat((ost::PayloadType) _audiocodec->getPayload(), _audiocodec->getClockRate()));
@@ -322,6 +323,8 @@ int
AudioRtpRTX::processDataEncode()
{
+ _debug("processDataEncode\n");
+
// compute codec framesize in ms
float fixed_codec_framesize = computeCodecFrameSize(_audiocodec->getFrameSize(), _audiocodec->getClockRate());
@@ -373,7 +376,7 @@ AudioRtpRTX::processDataEncode()
void
AudioRtpRTX::processDataDecode(unsigned char* spkrData, unsigned int size, int& countTime)
{
-
+ _debug("processDataDecode\n");
if (_audiocodec != NULL) {
// Return the size of data in bytes
@@ -427,7 +430,7 @@ AudioRtpRTX::sendSessionFromMic(int timestamp)
// 3. encode it
// 4. send it
-
+ _debug("sendSessionForSpeaker\n");
timestamp += time->getSecond();
// no call, so we do nothing
if (_ca==0) { _debug(" !ARTP: No call associated (mic)\n"); return; }
@@ -439,7 +442,7 @@ AudioRtpRTX::sendSessionFromMic(int timestamp)
int compSize = processDataEncode();
-
+ _debug("compSize: %i\n", compSize);
// putData put the data on RTP queue, sendImmediate bypass this queue
_session->putData(timestamp, micDataEncoded, compSize);
// _session->sendImmediate(timestamp, micDataEncoded, compSize);
@@ -451,6 +454,8 @@ AudioRtpRTX::sendSessionFromMic(int timestamp)
void
AudioRtpRTX::receiveSessionForSpkr (int& countTime)
{
+
+ _debug("receiveSessionForSpkr\n");
if (_ca == 0) { return; }
@@ -460,11 +465,9 @@ AudioRtpRTX::receiveSessionForSpkr (int& countTime)
const ost::AppDataUnit* adu = NULL;
- if (!_sym) {
- adu = _sessionRecv->getData(_sessionRecv->getFirstTimestamp());
- } else {
- adu = _session->getData(_session->getFirstTimestamp());
- }
+
+ adu = _session->getData(_session->getFirstTimestamp());
+
if (adu == NULL) {
// _debug("No RTP audio stream\n");
return;
@@ -498,8 +501,6 @@ void
AudioRtpRTX::run () {
int sessionWaiting;
-
- _session->startRunning();
initBuffers();
initAudioRtpSession();
@@ -522,6 +523,8 @@ AudioRtpRTX::run () {
_audiolayer->startStream();
+ _session->startRunning();
+
_debug("- ARTP Action: Start call %s\n",_ca->getCallId().c_str());
while (!testCancel()) {
@@ -529,8 +532,8 @@ AudioRtpRTX::run () {
sessionWaiting = _session->isWaiting();
sendSessionFromMic(timestamp);
- timestamp += timestep;
-
+ // timestamp += timestep;
+ timestamp = _session->getCurrentTimestamp();
// Recv session
receiveSessionForSpkr(countTime);
diff --git a/sflphone-common/src/audio/codecs/audiocodec.h b/sflphone-common/src/audio/codecs/audiocodec.h
index c7ad5ccf1468a78c025c5dbecee0887a68d960d9..1487fb465dce56cd9a5821ba8cd3a4695398adca 100644
--- a/sflphone-common/src/audio/codecs/audiocodec.h
+++ b/sflphone-common/src/audio/codecs/audiocodec.h
@@ -34,12 +34,30 @@ public:
: _codecName(codecName), _clockRate(8000), _channel(1), _bitrate(0.0),_bandwidth(0),_payload(payload), _hasDynamicPayload(false),_state(true) {
_hasDynamicPayload = (_payload >= 96 && _payload <= 127) ? true : false;
+
+ // If g722 (payload 9), we need to init libccrtp symetric sessions with using
+ // dynamic payload format. This way we get control on rtp clockrate.
+
+ if(_payload == 9)
+ {
+ _hasDynamicPayload = true;
+ }
+
}
AudioCodec( const AudioCodec& codec )
: _codecName(codec._codecName), _clockRate(codec._clockRate), _channel(codec._channel), _bitrate(codec._bitrate),_bandwidth(codec._bandwidth),_payload(codec._payload), _hasDynamicPayload(false),_state(true) {
_hasDynamicPayload = (_payload >= 96 && _payload <= 127) ? true : false;
+
+ // If g722 (payload 9), we need to init libccrtp symetric sessions with using
+ // dynamic payload format. This way we get control on rtp clockrate.
+
+ if(_payload == 9)
+ {
+ _hasDynamicPayload = true;
+ }
+
}
virtual ~AudioCodec() {
diff --git a/sflphone-common/src/sipvoiplink.cpp b/sflphone-common/src/sipvoiplink.cpp
index 897aded5b5d5b1c355475f19608c5371e19291d0..f1e8f2cf9fe5201eee26c1fc1c95deeb130d5141 100644
--- a/sflphone-common/src/sipvoiplink.cpp
+++ b/sflphone-common/src/sipvoiplink.cpp
@@ -508,14 +508,15 @@ SIPVoIPLink::newOutgoingCall ( const CallID& id, const std::string& toUrl )
call->setPeerNumber ( getSipTo ( toUrl, account->getHostname() ) );
setCallAudioLocal ( call, getLocalIPAddress(), useStun(), getStunServer() );
- /*
+
try {
_debug("CREATE NEW RTP SESSION FROM NEWOUTGOINGCALL\n");
_audiortp->createNewSession (call);
} catch (...) {
_debug("Failed to create rtp thread from newOutGoingCall\n");
}
- */
+
+
call->initRecFileName();
@@ -539,7 +540,7 @@ SIPVoIPLink::newOutgoingCall ( const CallID& id, const std::string& toUrl )
// _audiortp->start();
- call->setAudioStart ( true );
+ // call->setAudioStart ( true );
}
@@ -596,9 +597,7 @@ SIPVoIPLink::answer ( const CallID& id )
call->setConnectionState ( Call::Connected );
call->setState ( Call::Active );
- _audiortp->start();
-
- call->setAudioStart ( true );
+ ;
return true;
}
@@ -1233,6 +1232,8 @@ void
SIPVoIPLink::SIPCallAnswered ( SIPCall *call, pjsip_rx_data *rdata )
{
+ _debug ( "SIPCallAnswered\n" );
+
pjmedia_sdp_session *r_sdp;
if ( !call )
@@ -1260,20 +1261,23 @@ SIPVoIPLink::SIPCallAnswered ( SIPCall *call, pjsip_rx_data *rdata )
call->setState ( Call::Active );
Manager::instance().peerAnsweredCall ( call->getCallId() );
+ if (Manager::instance().isCurrentCall(call->getCallId()))
+ {
+ /*
+ _debug("CREATE NEW RTP SESSION FROM SIPCALLANSWERED\n");
+ try {
+ _audiortp->createNewSession (call);
+ call->setAudioStart(true);
+ } catch (...) {
+ _debug("Failed to create rtp thread from answer\n");
+ }
+ */
-
- _debug("CREATE NEW RTP SESSION FROM SIPCALLANSWERED\n");
- try {
- _audiortp->createNewSession (call);
- call->setAudioStart(true);
- } catch (...) {
- _debug("Failed to create rtp thread from answer\n");
- }
-
-
- call->setAudioStart(true);
+ call->setAudioStart(true);
- _audiortp->start();
+ _audiortp->start();
+
+ }
}
else
@@ -1346,13 +1350,13 @@ bool SIPVoIPLink::new_ip_to_ip_call ( const CallID& id, const std::string& to )
// Building the local SDP offer
call->getLocalSDP()->set_ip_address ( getLocalIP() );
call->getLocalSDP()->create_initial_offer();
- /*
+
try {
_audiortp->createNewSession (call);
} catch (...) {
_debug ( "! SIP Failure: Unable to create RTP Session in SIPVoIPLink::new_ip_to_ip_call (%s:%d)\n", __FILE__, __LINE__ );
}
- */
+
// Generate the contact URI
// uri_contact << "<" << uri_from << ":" << call->getLocalSDP()->get_local_extern_audio_port() << ">";
@@ -1391,9 +1395,9 @@ bool SIPVoIPLink::new_ip_to_ip_call ( const CallID& id, const std::string& to )
call->setState ( Call::Active );
addCall ( call );
- _audiortp->start();
+ // _audiortp->start();
- call->setAudioStart ( true );
+ // call->setAudioStart ( true );
return true;
}
@@ -1912,8 +1916,7 @@ void SIPVoIPLink::handle_reinvite ( SIPCall *call )
_audiortp->start();
call->setAudioStart ( true );
- // _audiortp->getRTX()->setRtpSessionRemoteIp();
- // _audiortp->getRTX()->setRtpSessionMedia();
+
}