diff --git a/sflphone-common/src/audio/audiortp/AudioRtpFactory.cpp b/sflphone-common/src/audio/audiortp/AudioRtpFactory.cpp
index e532515d604c30dac6e10e9e3a0fe8c3ce2fe2f7..646aa89b95d65652bb446f860364e011cc0ec92f 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpFactory.cpp
+++ b/sflphone-common/src/audio/audiortp/AudioRtpFactory.cpp
@@ -125,14 +125,14 @@ void AudioRtpFactory::initAudioRtpSession (SIPCall * ca)
void AudioRtpFactory::start (void)
{
if (_rtpSession == NULL) {
- throw AudioRtpFactoryException ("_rtpSession was null when trying to start audio thread");
+ throw AudioRtpFactoryException ("RTP: Error: _rtpSession was null when trying to start audio thread");
}
switch (_rtpSessionType) {
case Sdes:
if (static_cast<AudioSrtpSession *> (_rtpSession)->startRtpThread() != 0) {
- throw AudioRtpFactoryException ("Failed to start AudioSRtpSession thread");
+ throw AudioRtpFactoryException ("RTP: Error: Failed to start AudioSRtpSession thread");
}
break;
@@ -140,7 +140,7 @@ void AudioRtpFactory::start (void)
_debug ("Starting symmetric rtp thread");
if (static_cast<AudioSymmetricRtpSession *> (_rtpSession)->startRtpThread() != 0) {
- throw AudioRtpFactoryException ("Failed to start AudioSymmetricRtpSession thread");
+ throw AudioRtpFactoryException ("RTP: Error: Failed to start AudioSymmetricRtpSession thread");
}
break;
@@ -148,9 +148,8 @@ void AudioRtpFactory::start (void)
case Zrtp:
if (static_cast<AudioZrtpSession *> (_rtpSession)->startRtpThread() != 0) {
- throw AudioRtpFactoryException ("Failed to start AudioZrtpSession thread");
+ throw AudioRtpFactoryException ("RTP: Error: Failed to start AudioZrtpSession thread");
}
-
break;
}
}
@@ -158,10 +157,10 @@ void AudioRtpFactory::start (void)
void AudioRtpFactory::stop (void)
{
ost::MutexLock mutex (_audioRtpThreadMutex);
- _debug ("Stopping audio rtp session");
+ _info("RTP: Stopping audio rtp session");
if (_rtpSession == NULL) {
- _debugException ("_rtpSession is null when trying to stop. Returning.");
+ _debugException ("RTP: Error: _rtpSession is null when trying to stop. Returning.");
return;
}
@@ -183,16 +182,16 @@ void AudioRtpFactory::stop (void)
_rtpSession = NULL;
} catch (...) {
- _debugException ("Exception caught when stopping the audio rtp session");
- throw AudioRtpFactoryException("caught exception in AudioRtpFactory::stop");
+ _debugException ("RTP: Error: Exception caught when stopping the audio rtp session");
+ throw AudioRtpFactoryException("RTP: Error: caught exception in AudioRtpFactory::stop");
}
}
void AudioRtpFactory::updateDestinationIpAddress (void)
{
- _debug ("Updating IP address");
+ _info ("RTP: Updating IP address");
if (_rtpSession == NULL) {
- throw AudioRtpFactoryException ("_rtpSession was null when trying to update IP address");
+ throw AudioRtpFactoryException ("RTP: Error: _rtpSession was null when trying to update IP address");
}
switch (_rtpSessionType) {
@@ -216,19 +215,20 @@ sfl::AudioZrtpSession * AudioRtpFactory::getAudioZrtpSession()
if ( (_rtpSessionType == Zrtp) && (_rtpSessionType != NULL)) {
return static_cast<AudioZrtpSession *> (_rtpSession);
} else {
- throw AudioRtpFactoryException("_rtpSession is NULL in getAudioZrtpSession");
+ throw AudioRtpFactoryException("RTP: Error: _rtpSession is NULL in getAudioZrtpSession");
}
}
- void AudioRtpFactory::setRemoteCryptoInfo(sfl::SdesNegotiator& nego)
+void AudioRtpFactory::setRemoteCryptoInfo(sfl::SdesNegotiator& nego)
{
if ( _rtpSession && _rtpSessionType && (_rtpSessionType == Sdes)) {
static_cast<AudioSrtpSession *> (_rtpSession)->setRemoteCryptoInfo(nego);
}
else {
- throw AudioRtpFactoryException("_rtpSession is NULL in setRemoteCryptoInfo");
+ throw AudioRtpFactoryException("RTP: Error: _rtpSession is NULL in setRemoteCryptoInfo");
}
}
+
}
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpFactory.h b/sflphone-common/src/audio/audiortp/AudioRtpFactory.h
index cfa14165f1cf0f85960eb775b4b09fd476df7062..0076c968c8262ccd743a56b53d421439b1878dd4 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpFactory.h
+++ b/sflphone-common/src/audio/audiortp/AudioRtpFactory.h
@@ -63,27 +63,27 @@ namespace sfl {
void initAudioRtpConfig(SIPCall *ca);
- /**
- * Lazy instantiation method. Create a new RTP session of a given
- * type according to the content of the configuration file.
- * @param ca A pointer on a SIP call
- * @return A new AudioRtpSession object
- */
- void initAudioRtpSession(SIPCall *ca);
-
- /**
- * Start the audio rtp thread of the type specified in the configuration
- * file. initAudioRtpSession must have been called prior to that.
- * @param None
- */
- void start();
-
- /**
- * Stop the audio rtp thread of the type specified in the configuration
- * file. initAudioRtpSession must have been called prior to that.
- * @param None
- */
- void stop();
+ /**
+ * Lazy instantiation method. Create a new RTP session of a given
+ * type according to the content of the configuration file.
+ * @param ca A pointer on a SIP call
+ * @return A new AudioRtpSession object
+ */
+ void initAudioRtpSession(SIPCall *ca);
+
+ /**
+ * Start the audio rtp thread of the type specified in the configuration
+ * file. initAudioRtpSession must have been called prior to that.
+ * @param None
+ */
+ void start();
+
+ /**
+ * Stop the audio rtp thread of the type specified in the configuration
+ * file. initAudioRtpSession must have been called prior to that.
+ * @param None
+ */
+ void stop();
/**
* Update current RTP destination address with one stored in call
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpSession.h b/sflphone-common/src/audio/audiortp/AudioRtpSession.h
index 29057bec3ec82fbcb302f8e172164d1f8a20bbf0..b6d4908445ba7cee17c3230976ec340413329f93 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpSession.h
+++ b/sflphone-common/src/audio/audiortp/AudioRtpSession.h
@@ -67,10 +67,10 @@ namespace sfl {
int startRtpThread();
- /**
- * Used mostly when receiving a reinvite
- */
- void updateDestinationIpAddress(void);
+ /**
+ * Used mostly when receiving a reinvite
+ */
+ void updateDestinationIpAddress(void);
private:
@@ -104,15 +104,16 @@ namespace sfl {
// start() with no semaphore at all.
ost::Semaphore * _mainloopSemaphore;
- // Main destination address for this rtp session.
- // Stored in case or reINVITE, which may require to forget
- // this destination and update a new one.
- ost::InetHostAddress _remote_ip;
+ // Main destination address for this rtp session.
+ // Stored in case or reINVITE, which may require to forget
+ // this destination and update a new one.
+ ost::InetHostAddress _remote_ip;
+
- // Main destination port for this rtp session.
- // Stored in case reINVITE, which may require to forget
- // this destination and update a new one
- unsigned short _remote_port;
+ // Main destination port for this rtp session.
+ // Stored in case reINVITE, which may require to forget
+ // this destination and update a new one
+ unsigned short _remote_port;
AudioCodec * _audiocodec;
@@ -163,23 +164,23 @@ namespace sfl {
*/
ManagerImpl * _manager;
- /**
- * Sampling rate of audio converter
- */
- int _converterSamplingRate;
+ /**
+ * Sampling rate of audio converter
+ */
+ int _converterSamplingRate;
- /**
- * Timestamp for this session
- */
- int _timestamp;
+ /**
+ * Timestamp for this session
+ */
+ int _timestamp;
- /**
- * Time counter used to trigger incoming call notification
- */
- int _countNotificationTime;
+ /**
+ * Time counter used to trigger incoming call notification
+ */
+ int _countNotificationTime;
protected:
- SIPCall * _ca;
+ SIPCall * _ca;
};
@@ -208,7 +209,7 @@ namespace sfl {
assert(_ca);
- _debug ("Local audio port %i will be used", _ca->getLocalAudioPort());
+ _info ("Rtp: Local audio port %i will be used", _ca->getLocalAudioPort());
//mic, we receive from soundcard in stereo, and we send encoded
_audiolayer = _manager->getAudioDriver();
@@ -223,7 +224,7 @@ namespace sfl {
template <typename D>
AudioRtpSession<D>::~AudioRtpSession()
{
- _debug ("Delete AudioRtpSession instance");
+ _debug ("Rtp: Delete AudioRtpSession instance");
try {
terminate();
@@ -232,9 +233,7 @@ namespace sfl {
throw;
}
- _debug("Unbind audio RTP stream for call id %s", _ca->getCallId().c_str());
- // _audiolayer->getMainBuffer()->unBindAll(_ca->getCallId());
- _manager->getAudioDriver()->getMainBuffer()->unBindAll(_ca->getCallId());
+ _manager->getAudioDriver()->getMainBuffer()->unBindAll(_ca->getCallId());
delete [] _micData;
delete [] _micDataConverted;
@@ -243,22 +242,21 @@ namespace sfl {
delete [] _spkrDataConverted;
delete _time;
delete _converter;
- _debug ("AudioRtpSession instance deleted");
}
template <typename D>
void AudioRtpSession<D>::initBuffers()
{
- // Set sampling rate, main buffer choose the highest one
- // _audiolayer->getMainBuffer()->setInternalSamplingRate(_codecSampleRate);
+ // Set sampling rate, main buffer choose the highest one
+ // _audiolayer->getMainBuffer()->setInternalSamplingRate(_codecSampleRate);
_manager->getAudioDriver()->getMainBuffer()->setInternalSamplingRate(_codecSampleRate);
- // may be different than one already setted
- // converterSamplingRate = _audiolayer->getMainBuffer()->getInternalSamplingRate();
- _converterSamplingRate = _manager->getAudioDriver()->getMainBuffer()->getInternalSamplingRate();
+ // may be different than one already setted
+ // converterSamplingRate = _audiolayer->getMainBuffer()->getInternalSamplingRate();
+ _converterSamplingRate = _manager->getAudioDriver()->getMainBuffer()->getInternalSamplingRate();
- // initialize SampleRate converter using AudioLayer's sampling rate
- // (internal buffers initialized with maximal sampling rate and frame size)
+ // initialize SampleRate converter using AudioLayer's sampling rate
+ // (internal buffers initialized with maximal sampling rate and frame size)
_converter = new SamplerateConverter(_layerSampleRate, _layerFrameSize);
int nbSamplesMax = (int)(_codecSampleRate * _layerFrameSize /1000)*2;
@@ -268,7 +266,7 @@ namespace sfl {
_spkrDataConverted = new SFLDataFormat[nbSamplesMax];
_spkrDataDecoded = new SFLDataFormat[nbSamplesMax];
- _manager->addStream(_ca->getCallId());
+ _manager->addStream(_ca->getCallId());
}
template <typename D>
@@ -288,8 +286,8 @@ namespace sfl {
{
assert(_ca);
- AudioCodecType pl = (AudioCodecType)_ca->getLocalSDP()->get_session_media()->getPayload();
- _audiocodec = _manager->getCodecDescriptorMap().instantiateCodec(pl);
+ AudioCodecType pl = (AudioCodecType)_ca->getLocalSDP()->get_session_media()->getPayload();
+ _audiocodec = _manager->getCodecDescriptorMap().instantiateCodec(pl);
if (_audiocodec == NULL) {
_debug ("No audiocodec, can't init RTP media");
@@ -318,26 +316,29 @@ namespace sfl {
void AudioRtpSession<D>::setDestinationIpAddress(void)
{
if (_ca == NULL) {
- _debug ("Sipcall is gone.");
+ _warn ("Rtp: Sipcall is gone.");
throw AudioRtpSessionException();
}
- _debug ("Setting IP address for the RTP session");
+ _info ("RTP: Setting IP address for the RTP session");
- // Store remote ip in case we would need to forget current destination
+ // Store remote ip in case we would need to forget current destination
_remote_ip = ost::InetHostAddress(_ca->getLocalSDP()->get_remote_ip().c_str());
- _debug ("Init audio RTP session: remote ip %s", _ca->getLocalSDP()->get_remote_ip().data());
if (!_remote_ip) {
- _debug ("Target IP address [%s] is not correct!", _ca->getLocalSDP()->get_remote_ip().data());
+ _warn("Rtp: Target IP address (%s) is not correct!",
+ _ca->getLocalSDP()->get_remote_ip().data());
return;
}
- // Store remote port in case we would need to forget current destination
- _remote_port = (unsigned short) _ca->getLocalSDP()->get_remote_audio_port();
+ // Store remote port in case we would need to forget current destination
+ _remote_port = (unsigned short) _ca->getLocalSDP()->get_remote_audio_port();
+
+ _info("RTP: New remote address for session: %s:%d",
+ _ca->getLocalSDP()->get_remote_ip().data(), _remote_port);
if (! static_cast<D*>(this)->addDestination (_remote_ip, _remote_port)) {
- _debug ("Can't add destination to session!");
+ _warn("Rtp: Can't add new destination to session!");
return;
}
}
@@ -346,12 +347,13 @@ namespace sfl {
void AudioRtpSession<D>::updateDestinationIpAddress(void)
{
// Destination address are stored in a list in ccrtp
- // This method clear off this entry
- _debug("updateDestinationIpAddress: remove destination %s", _ca->getLocalSDP()->get_remote_ip().c_str());
- static_cast<D*>(this)->forgetDestination(_remote_ip, _remote_port);
+ // This method remove the current destination entry
+
+ if(!static_cast<D*>(this)->forgetDestination(_remote_ip, _remote_port, _remote_port+1))
+ _warn("Rtp: Could not remove previous destination");
- // new destination is stored in call
- // we just need to recall this method
+ // new destination is stored in call
+ // we just need to recall this method
setDestinationIpAddress();
}
@@ -534,7 +536,6 @@ namespace sfl {
{
setSessionTimeouts();
- setDestinationIpAddress();
setSessionMedia();
initBuffers();
diff --git a/sflphone-common/src/audio/audiortp/AudioZrtpSession.cpp b/sflphone-common/src/audio/audiortp/AudioZrtpSession.cpp
index 5b6b67101f740635b5eb7f9ebcc77343689eba93..cb82fe29e227fe0071d48c6aeda2cab3be5e9008 100644
--- a/sflphone-common/src/audio/audiortp/AudioZrtpSession.cpp
+++ b/sflphone-common/src/audio/audiortp/AudioZrtpSession.cpp
@@ -38,7 +38,6 @@ AudioZrtpSession::AudioZrtpSession (ManagerImpl * manager, SIPCall * sipcall, co
{
_debug ("AudioZrtpSession initialized");
initializeZid();
- // startZrtp();
}
void AudioZrtpSession::initializeZid (void)
diff --git a/sflphone-common/src/audio/audiortp/AudioZrtpSession.h b/sflphone-common/src/audio/audiortp/AudioZrtpSession.h
index 2e41e04b45b42086f903e551207563cdd9fdfba7..138bd2684733881494e2e26f8e1af01c85fe931c 100644
--- a/sflphone-common/src/audio/audiortp/AudioZrtpSession.h
+++ b/sflphone-common/src/audio/audiortp/AudioZrtpSession.h
@@ -38,8 +38,8 @@ namespace sfl {
class AudioZrtpSession : public ost::SymmetricZRTPSession, public AudioRtpSession<AudioZrtpSession>
{
public:
- AudioZrtpSession(ManagerImpl * manager, SIPCall * sipcall, const std::string& zidFilename);
-
+ AudioZrtpSession(ManagerImpl * manager, SIPCall * sipcall, const std::string& zidFilename);
+
private:
void initializeZid(void);
std::string _zidFilename;
diff --git a/sflphone-common/src/audio/audiortp/ZrtpSessionCallback.cpp b/sflphone-common/src/audio/audiortp/ZrtpSessionCallback.cpp
index f866b39e007a7b9ffd11fb4c5307a05c6b9f0547..145792e0bce09f97977772b4602a12559e2d4566 100644
--- a/sflphone-common/src/audio/audiortp/ZrtpSessionCallback.cpp
+++ b/sflphone-common/src/audio/audiortp/ZrtpSessionCallback.cpp
@@ -150,7 +150,6 @@ ZrtpSessionCallback::showMessage (GnuZrtpCodes::MessageSeverity sev, int32_t sub
msg = _infoMap[subCode];
if (msg != NULL) {
- _debug ("ZRTP Debug:");
}
}
@@ -158,7 +157,6 @@ ZrtpSessionCallback::showMessage (GnuZrtpCodes::MessageSeverity sev, int32_t sub
msg = _warningMap[subCode];
if (msg != NULL) {
- _debug ("ZRTP Debug:");
}
}
@@ -166,10 +164,11 @@ ZrtpSessionCallback::showMessage (GnuZrtpCodes::MessageSeverity sev, int32_t sub
msg = _severeMap[subCode];
if (msg != NULL) {
- _debug ("ZRTP Debug:");
}
}
+
+
if (sev == ZrtpError) {
if (subCode < 0) { // received an error packet from peer
subCode *= -1;
@@ -181,7 +180,7 @@ ZrtpSessionCallback::showMessage (GnuZrtpCodes::MessageSeverity sev, int32_t sub
msg = _zrtpMap[subCode];
if (msg != NULL) {
- _debug ("ZRTP Debug: %s", msg->c_str());
+
}
}
}
diff --git a/sflphone-common/src/sip/sdp.cpp b/sflphone-common/src/sip/sdp.cpp
index ba4e856b301a3009fd1a222f60b157bcbd67fa27..886dc1430f588511d7b3794f3cf77b96a6a31c83 100644
--- a/sflphone-common/src/sip/sdp.cpp
+++ b/sflphone-common/src/sip/sdp.cpp
@@ -489,9 +489,6 @@ void Sdp::set_negotiated_sdp (const pjmedia_sdp_session *sdp)
pjmedia_sdp_attr_to_rtpmap (_pool, attribute, &rtpmap);
- // _debug("================== set_negociated_offer ===================== %i", pj_strtoul(&rtpmap->pt));
- // _debug("================== set_negociated_offer ===================== %s", current->desc.fmt[j].ptr);
- // _debug("================== set_negociated_offer ===================== %i", atoi(current->desc.fmt[j].ptr));
iter = codecs_list.find ( (AudioCodecType) pj_strtoul (&rtpmap->pt));
if (iter==codecs_list.end())
@@ -627,7 +624,7 @@ void Sdp::set_remote_ip_from_sdp (const pjmedia_sdp_session *r_sdp)
{
std::string remote_ip (r_sdp->conn->addr.ptr, r_sdp->conn->addr.slen);
- _debug (" Remote IP from fetching SDP: %s", remote_ip.c_str());
+ _info ("SDP: Remote IP from fetching SDP: %s", remote_ip.c_str());
this->set_remote_ip (remote_ip);
}
@@ -637,21 +634,21 @@ void Sdp::set_remote_audio_port_from_sdp (pjmedia_sdp_media *r_media)
int remote_port;
remote_port = r_media->desc.port;
- _debug (" Remote Audio Port from fetching SDP: %d", remote_port);
+ _info ("SDP: Remote Audio Port from fetching SDP: %d", remote_port);
this->set_remote_audio_port (remote_port);
}
void Sdp::set_media_transport_info_from_remote_sdp (const pjmedia_sdp_session *remote_sdp)
{
- _debug ("Fetching media from sdp");
+ _info ("SDP: Fetching media from sdp");
pjmedia_sdp_media *r_media;
this->get_remote_sdp_media_from_offer (remote_sdp, &r_media);
if (r_media==NULL) {
- _debug ("SDP Failure: no remote sdp media found in the remote offer");
+ _warn ("SDP: Error: no remote sdp media found in the remote offer");
return;
}
diff --git a/sflphone-common/src/sip/sipvoiplink.cpp b/sflphone-common/src/sip/sipvoiplink.cpp
index 9e25418377d8708030acd33bb9c0a9adb8a38cd6..5c5bed594177da1b7dd388a7f38b1b50cb699d4b 100644
--- a/sflphone-common/src/sip/sipvoiplink.cpp
+++ b/sflphone-common/src/sip/sipvoiplink.cpp
@@ -1020,7 +1020,7 @@ int SIPVoIPLink::inv_session_reinvite (SIPCall *call, std::string direction)
local_sdp = call->getLocalSDP()->get_local_sdp_session();
if (local_sdp == NULL) {
- _debug ("! SIP Failure: unable to find local_sdp");
+ _debug ("SIP: Error: unable to find local sdp");
return !PJ_SUCCESS;
}
@@ -3142,7 +3142,7 @@ void call_on_state_changed (pjsip_inv_session *inv, pjsip_event *e)
// This callback is called after SDP offer/answer session has completed.
void call_on_media_update (pjsip_inv_session *inv, pj_status_t status)
{
- _debug ("UserAgent: call_on_media_update");
+ _debug ("UserAgent: Call on media update");
const pjmedia_sdp_session *local_sdp;
const pjmedia_sdp_session *remote_sdp;
@@ -3153,19 +3153,19 @@ void call_on_media_update (pjsip_inv_session *inv, pj_status_t status)
call = reinterpret_cast<SIPCall *> (inv->mod_data[getModId() ]);
if (!call) {
- _debug ("Call declined by peer, SDP negociation stopped");
+ _debug ("UserAgent: Call declined by peer, SDP negociation stopped");
return;
}
link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink (AccountNULL));
if (link == NULL) {
- _debug ("Failed to get sip link");
+ _warn ("UserAgent: Error: Failed to get sip link");
return;
}
if (status != PJ_SUCCESS) {
- _debug ("Error while negotiating the offer");
+ _warn ("UserAgent: Error: while negotiating the offer");
link->hangup (call->getCallId());
Manager::instance().callFailure (call->getCallId());
return;
@@ -3198,39 +3198,39 @@ void call_on_media_update (pjsip_inv_session *inv, pj_status_t status)
}
+
// Get the crypto attribute containing srtp's cryptographic context (keys, cipher)
CryptoOffer crypto_offer;
call->getLocalSDP()->get_remote_sdp_crypto_from_offer(remote_sdp, crypto_offer);
-
bool nego_success = false;
if(!crypto_offer.empty()) {
- _debug("Crypto attribute in SDP: init Srtp session");
+ _debug("Crypto attribute in SDP: init Srtp session");
- // init local cryptografic capabilities for negotiation
- std::vector<sfl::CryptoSuiteDefinition>localCapabilities;
- for(int i = 0; i < 3; i++) {
- localCapabilities.push_back(sfl::CryptoSuites[i]);
- }
+ // init local cryptografic capabilities for negotiation
+ std::vector<sfl::CryptoSuiteDefinition>localCapabilities;
+ for(int i = 0; i < 3; i++) {
+ localCapabilities.push_back(sfl::CryptoSuites[i]);
+ }
- sfl::SdesNegotiator sdesnego(localCapabilities, crypto_offer);
+ sfl::SdesNegotiator sdesnego(localCapabilities, crypto_offer);
- if(sdesnego.negotiate()) {
- _debug("SDES negociation successfull \n");
- nego_success = true;
+ if(sdesnego.negotiate()) {
+ _debug("SDES negociation successfull \n");
+ nego_success = true;
_debug("Set remote cryptographic context\n");
try {
- call->getAudioRtp()->setRemoteCryptoInfo(sdesnego);
+ call->getAudioRtp()->setRemoteCryptoInfo(sdesnego);
}
catch(...) {}
- DBusManager::instance().getCallManager()->secureSdesOn (call->getCallId());
- }
- else {
- DBusManager::instance().getCallManager()->secureSdesOff (call->getCallId());
- }
+ DBusManager::instance().getCallManager()->secureSdesOn (call->getCallId());
+ }
+ else {
+ DBusManager::instance().getCallManager()->secureSdesOff (call->getCallId());
+ }
}
// We did not found any crypto context for this media
@@ -3256,6 +3256,7 @@ void call_on_media_update (pjsip_inv_session *inv, pj_status_t status)
_debug("Sdes not initialized for this call\n");
}
+
try {
call->setAudioStart (true);
call->getAudioRtp()->start();