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();