diff --git a/sflphone-common/src/audio/audiortp/AudioRtpFactory.cpp b/sflphone-common/src/audio/audiortp/AudioRtpFactory.cpp index 65516192f168f229d0a1eb64d64d4a6369d28480..70f21a0626f1480e768fd4ea491271dc6978636e 100644 --- a/sflphone-common/src/audio/audiortp/AudioRtpFactory.cpp +++ b/sflphone-common/src/audio/audiortp/AudioRtpFactory.cpp @@ -33,7 +33,7 @@ #include "AudioRtpFactory.h" #include "AudioZrtpSession.h" #include "AudioSrtpSession.h" -#include "AudioRtpSession.h" +#include "AudioSymmetricRtpSession.h" #include "manager.h" #include "sip/sdp.h" #include "sip/sipcall.h" @@ -102,7 +102,7 @@ void AudioRtpFactory::registerAccount(SIPAccount *sipaccount, const std::string& _helloHashEnabled = sipaccount->getZrtpHelloHash(); } -void AudioRtpFactory::initAudioRtpSession (SIPCall * ca) +void AudioRtpFactory::initAudioSymmetricRtpSession (SIPCall * ca) { ost::MutexLock m (_audioRtpThreadMutex); @@ -138,7 +138,7 @@ void AudioRtpFactory::initAudioRtpSession (SIPCall * ca) } } else { _rtpSessionType = Symmetric; - _rtpSession = new AudioRtpSession (ca); + _rtpSession = new AudioSymmetricRtpSession (ca); _debug ("AudioRtpFactory: Starting a symmetric unencrypted rtp session"); } } @@ -166,8 +166,8 @@ void AudioRtpFactory::start (AudioCodec* audiocodec) case Symmetric: _debug ("Starting symmetric rtp thread"); - if (static_cast<AudioRtpSession *> (_rtpSession)->startRtpThread (audiocodec) != 0) { - throw AudioRtpFactoryException ("AudioRtpFactory: Error: Failed to start AudioRtpSession thread"); + if (static_cast<AudioSymmetricRtpSession *> (_rtpSession)->startRtpThread (audiocodec) != 0) { + throw AudioRtpFactoryException ("AudioRtpFactory: Error: Failed to start AudioSymmetricRtpSession thread"); } break; @@ -202,7 +202,7 @@ void AudioRtpFactory::stop (void) break; case Symmetric: - static_cast<AudioRtpSession *> (_rtpSession)->stopRtpThread(); + static_cast<AudioSymmetricRtpSession *> (_rtpSession)->stopRtpThread(); break; case Zrtp: @@ -232,7 +232,7 @@ int AudioRtpFactory::getSessionMedia() payloadType = static_cast<AudioSrtpSession *> (_rtpSession)->getCodecPayloadType(); break; case Symmetric: - payloadType = static_cast<AudioRtpSession *> (_rtpSession)->getCodecPayloadType(); + payloadType = static_cast<AudioSymmetricRtpSession *> (_rtpSession)->getCodecPayloadType(); break; case Zrtp: payloadType = static_cast<AudioZrtpSession *> (_rtpSession)->getCodecPayloadType(); @@ -255,7 +255,7 @@ void AudioRtpFactory::updateSessionMedia (AudioCodec *audiocodec) static_cast<AudioSrtpSession *> (_rtpSession)->updateSessionMedia (audiocodec); break; case Symmetric: - static_cast<AudioRtpSession *> (_rtpSession)->updateSessionMedia (audiocodec); + static_cast<AudioSymmetricRtpSession *> (_rtpSession)->updateSessionMedia (audiocodec); break; case Zrtp: static_cast<AudioZrtpSession *> (_rtpSession)->updateSessionMedia (audiocodec); @@ -281,7 +281,7 @@ void AudioRtpFactory::updateDestinationIpAddress (void) break; case Symmetric: - static_cast<AudioRtpSession *> (_rtpSession)->updateDestinationIpAddress(); + static_cast<AudioSymmetricRtpSession *> (_rtpSession)->updateDestinationIpAddress(); break; case Zrtp: diff --git a/sflphone-common/src/audio/audiortp/AudioRtpFactory.h b/sflphone-common/src/audio/audiortp/AudioRtpFactory.h index 939eabccbb5f9ba79c0020c2e0ab318c9efd240d..c4840f0ac8dc752b49d6e87da6fe30eae379a6ca 100644 --- a/sflphone-common/src/audio/audiortp/AudioRtpFactory.h +++ b/sflphone-common/src/audio/audiortp/AudioRtpFactory.h @@ -82,20 +82,20 @@ class AudioRtpFactory * 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 + * @return A new AudioSymmetricRtpSession object */ - void initAudioRtpSession (SIPCall *ca); + void initAudioSymmetricRtpSession (SIPCall *ca); /** * Start the audio rtp thread of the type specified in the configuration - * file. initAudioRtpSession must have been called prior to that. + * file. initAudioSymmetricRtpSession must have been called prior to that. * @param None */ void start (AudioCodec*); /** * Stop the audio rtp thread of the type specified in the configuration - * file. initAudioRtpSession must have been called prior to that. + * file. initAudioSymmetricRtpSession must have been called prior to that. * @param None */ void stop(); @@ -119,9 +119,9 @@ class AudioRtpFactory /** * @param None * @return The internal audio rtp thread of the type specified in the configuration - * file. initAudioRtpSession must have been called prior to that. + * file. initAudioSymmetricRtpSession must have been called prior to that. */ - void * getAudioRtpSession (void) const { + void * getAudioSymmetricRtpSession (void) const { return _rtpSession; } diff --git a/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp b/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp index 1427eb68d0585ad3669f067773a7f0ee544d8bf9..b44ab585d0f559ccabdd0df72769c370a364e197 100644 --- a/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp +++ b/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp @@ -227,7 +227,7 @@ void AudioRtpRecordHandler::updateNoiseSuppress() _audioRtpRecord._noiseSuppress = NULL; - _debug ("AudioRtpSession: Update noise suppressor with sampling rate %d and frame size %d", getCodecSampleRate(), getCodecFrameSize()); + _debug ("AudioSymmetricRtpSession: Update noise suppressor with sampling rate %d and frame size %d", getCodecSampleRate(), getCodecFrameSize()); NoiseSuppress *noiseSuppress = new NoiseSuppress (getCodecFrameSize(), getCodecSampleRate()); AudioProcessing *processing = new AudioProcessing (noiseSuppress); @@ -249,7 +249,7 @@ void AudioRtpRecordHandler::putDtmfEvent (int digit) dtmf->newevent = true; dtmf->length = 1000; getEventQueue()->push_back (dtmf); - _debug ("AudioRtpSession: Put Dtmf Event %d", digit); + _debug ("AudioSymmetricRtpSession: Put Dtmf Event %d", digit); } #ifdef DUMP_PROCESS_DATA_ENCODE diff --git a/sflphone-common/src/audio/audiortp/AudioSrtpSession.cpp b/sflphone-common/src/audio/audiortp/AudioSrtpSession.cpp index c7f3269bf696cf4386b43c5c2004156611318410..6eee958ebfd1fd2183d4b855361b34fb50fe9d6d 100644 --- a/sflphone-common/src/audio/audiortp/AudioSrtpSession.cpp +++ b/sflphone-common/src/audio/audiortp/AudioSrtpSession.cpp @@ -46,7 +46,7 @@ namespace sfl { AudioSrtpSession::AudioSrtpSession (SIPCall * sipcall) : - AudioRtpSession (sipcall), + AudioSymmetricRtpSession (sipcall), _remoteCryptoCtx (NULL), _localCryptoCtx (NULL), _localCryptoSuite (0), diff --git a/sflphone-common/src/audio/audiortp/AudioSrtpSession.h b/sflphone-common/src/audio/audiortp/AudioSrtpSession.h index 86c2718ca82328f1c402448711d5ac50cb7357d4..73e8cd78ef6f89af50d172350ed61e4948239ba0 100644 --- a/sflphone-common/src/audio/audiortp/AudioSrtpSession.h +++ b/sflphone-common/src/audio/audiortp/AudioSrtpSession.h @@ -30,7 +30,7 @@ #ifndef __SFL_AUDIO_SRTP_SESSION_H__ #define __SFL_AUDIO_SRTP_SESSION_H__ -#include "AudioRtpSession.h" +#include "AudioSymmetricRtpSession.h" #include "sip/SdesNegotiator.h" #include <ccrtp/CryptoContext.h> @@ -66,7 +66,7 @@ class SIPCall; namespace sfl { -class AudioSrtpSession : public AudioRtpSession +class AudioSrtpSession : public AudioSymmetricRtpSession { public: diff --git a/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp b/sflphone-common/src/audio/audiortp/AudioSymmetricRtpSession.cpp similarity index 71% rename from sflphone-common/src/audio/audiortp/AudioRtpSession.cpp rename to sflphone-common/src/audio/audiortp/AudioSymmetricRtpSession.cpp index 59d96332e58f06f6aa670900449ee66654bd7709..cd98dedf5c19d2e3e1abb9e77a85fb52adae6177 100644 --- a/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp +++ b/sflphone-common/src/audio/audiortp/AudioSymmetricRtpSession.cpp @@ -32,14 +32,14 @@ * as that of the covered work. */ -#include "AudioRtpSession.h" +#include "AudioSymmetricRtpSession.h" #include "sip/sdp.h" #include "audio/audiolayer.h" namespace sfl { -AudioRtpSession::AudioRtpSession (SIPCall * sipcall) : +AudioSymmetricRtpSession::AudioSymmetricRtpSession (SIPCall * sipcall) : AudioRtpRecordHandler (sipcall) , ost::SymmetricRTPSession (ost::InetHostAddress (sipcall->getLocalIp().c_str()), sipcall->getLocalAudioPort()) , _mainloopSemaphore (0) @@ -53,37 +53,37 @@ AudioRtpSession::AudioRtpSession (SIPCall * sipcall) : { assert (_ca); - _info ("AudioRtpSession: Setting new RTP session with destination %s:%d", _ca->getLocalIp().c_str(), _ca->getLocalAudioPort()); + _info ("AudioSymmetricRtpSession: Setting new RTP session with destination %s:%d", _ca->getLocalIp().c_str(), _ca->getLocalAudioPort()); _audioRtpRecord._callId = _ca->getCallId(); setTypeOfService (tosEnhanced); } -AudioRtpSession::~AudioRtpSession() +AudioSymmetricRtpSession::~AudioSymmetricRtpSession() { - _info ("AudioRtpSession: Delete AudioRtpSession instance"); + _info ("AudioSymmetricRtpSession: Delete AudioSymmetricRtpSession instance"); } -void AudioRtpSession::final() +void AudioSymmetricRtpSession::final() { delete _rtpThread; - delete static_cast<AudioRtpSession *> (this); + delete static_cast<AudioSymmetricRtpSession *> (this); } -void AudioRtpSession::setSessionTimeouts (void) +void AudioSymmetricRtpSession::setSessionTimeouts (void) { - _debug ("AudioRtpSession: Set session scheduling timeout (%d) and expireTimeout (%d)", sfl::schedulingTimeout, sfl::expireTimeout); + _debug ("AudioSymmetricRtpSession: Set session scheduling timeout (%d) and expireTimeout (%d)", sfl::schedulingTimeout, sfl::expireTimeout); setSchedulingTimeout (sfl::schedulingTimeout); setExpireTimeout (sfl::expireTimeout); } -void AudioRtpSession::setSessionMedia (AudioCodec *audioCodec) +void AudioSymmetricRtpSession::setSessionMedia (AudioCodec *audioCodec) { - _debug ("AudioRtpSession: Set session media"); + _debug ("AudioSymmetricRtpSession: Set session media"); // set internal codec info for this session setRtpMedia (audioCodec); @@ -101,19 +101,19 @@ void AudioRtpSession::setSessionMedia (AudioCodec *audioCodec) _timestampIncrement = frameSize; _debug ("AudioRptSession: Codec payload: %d", payloadType); - _debug ("AudioRtpSession: Codec sampling rate: %d", smplRate); - _debug ("AudioRtpSession: Codec frame size: %d", frameSize); - _debug ("AudioRtpSession: RTP timestamp increment: %d", _timestampIncrement); + _debug ("AudioSymmetricRtpSession: Codec sampling rate: %d", smplRate); + _debug ("AudioSymmetricRtpSession: Codec frame size: %d", frameSize); + _debug ("AudioSymmetricRtpSession: RTP timestamp increment: %d", _timestampIncrement); if (payloadType == g722PayloadType) { - _debug ("AudioRtpSession: Setting G722 payload format"); + _debug ("AudioSymmetricRtpSession: Setting G722 payload format"); setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) payloadType, g722RtpClockRate)); } else { if (dynamic) { - _debug ("AudioRtpSession: Setting dynamic payload format"); + _debug ("AudioSymmetricRtpSession: Setting dynamic payload format"); setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) payloadType, smplRate)); } else { - _debug ("AudioRtpSession: Setting static payload format"); + _debug ("AudioSymmetricRtpSession: Setting static payload format"); setPayloadFormat (ost::StaticPayloadFormat ( (ost::StaticPayloadType) payloadType)); } } @@ -121,9 +121,9 @@ void AudioRtpSession::setSessionMedia (AudioCodec *audioCodec) _ca->setRecordingSmplRate (getCodecSampleRate()); } -void AudioRtpSession::updateSessionMedia (AudioCodec *audioCodec) +void AudioSymmetricRtpSession::updateSessionMedia (AudioCodec *audioCodec) { - _debug ("AudioRtpSession: Update session media"); + _debug ("AudioSymmetricRtpSession: Update session media"); // Update internal codec for this session updateRtpMedia (audioCodec); @@ -140,19 +140,19 @@ void AudioRtpSession::updateSessionMedia (AudioCodec *audioCodec) _timestampIncrement = frameSize; _debug ("AudioRptSession: Codec payload: %d", payloadType); - _debug ("AudioRtpSession: Codec sampling rate: %d", smplRate); - _debug ("AudioRtpSession: Codec frame size: %d", frameSize); - _debug ("AudioRtpSession: RTP timestamp increment: %d", _timestampIncrement); + _debug ("AudioSymmetricRtpSession: Codec sampling rate: %d", smplRate); + _debug ("AudioSymmetricRtpSession: Codec frame size: %d", frameSize); + _debug ("AudioSymmetricRtpSession: RTP timestamp increment: %d", _timestampIncrement); if (payloadType == g722PayloadType) { - _debug ("AudioRtpSession: Setting G722 payload format"); + _debug ("AudioSymmetricRtpSession: Setting G722 payload format"); setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) payloadType, g722RtpClockRate)); } else { if (dynamic) { - _debug ("AudioRtpSession: Setting dynamic payload format"); + _debug ("AudioSymmetricRtpSession: Setting dynamic payload format"); setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) payloadType, smplRate)); } else { - _debug ("AudioRtpSession: Setting static payload format"); + _debug ("AudioSymmetricRtpSession: Setting static payload format"); setPayloadFormat (ost::StaticPayloadFormat ( (ost::StaticPayloadType) payloadType)); } } @@ -164,14 +164,14 @@ void AudioRtpSession::updateSessionMedia (AudioCodec *audioCodec) } -void AudioRtpSession::setDestinationIpAddress (void) +void AudioSymmetricRtpSession::setDestinationIpAddress (void) { - _info ("AudioRtpSession: Setting IP address for the RTP session"); + _info ("AudioSymmetricRtpSession: Setting IP address for the RTP session"); // Store remote ip in case we would need to forget current destination _remote_ip = ost::InetHostAddress (_ca->getLocalSDP()->getRemoteIP().c_str()); if (!_remote_ip) { - _warn ("AudioRtpSession: Target IP address (%s) is not correct!", + _warn ("AudioSymmetricRtpSession: Target IP address (%s) is not correct!", _ca->getLocalSDP()->getRemoteIP().data()); return; } @@ -179,24 +179,24 @@ void AudioRtpSession::setDestinationIpAddress (void) // Store remote port in case we would need to forget current destination _remote_port = (unsigned short) _ca->getLocalSDP()->getRemoteAudioPort(); - _info ("AudioRtpSession: New remote address for session: %s:%d", + _info ("AudioSymmetricRtpSession: New remote address for session: %s:%d", _ca->getLocalSDP()->getRemoteIP().data(), _remote_port); if (!addDestination (_remote_ip, _remote_port)) { - _warn ("AudioRtpSession: Can't add new destination to session!"); + _warn ("AudioSymmetricRtpSession: Can't add new destination to session!"); return; } } -void AudioRtpSession::updateDestinationIpAddress (void) +void AudioSymmetricRtpSession::updateDestinationIpAddress (void) { - _debug ("AudioRtpSession: Update destination ip address"); + _debug ("AudioSymmetricRtpSession: Update destination ip address"); // Destination address are stored in a list in ccrtp // This method remove the current destination entry if (!forgetDestination (_remote_ip, _remote_port, _remote_port+1)) { - _warn ("AudioRtpSession: Could not remove previous destination: %s:%d", + _warn ("AudioSymmetricRtpSession: Could not remove previous destination: %s:%d", inet_ntoa(_remote_ip.getAddress()), _remote_port); } @@ -205,9 +205,9 @@ void AudioRtpSession::updateDestinationIpAddress (void) setDestinationIpAddress(); } -void AudioRtpSession::sendDtmfEvent (sfl::DtmfEvent *dtmf) +void AudioSymmetricRtpSession::sendDtmfEvent (sfl::DtmfEvent *dtmf) { - _debug ("AudioRtpSession: Send Dtmf"); + _debug ("AudioSymmetricRtpSession: Send Dtmf"); _timestamp += _timestampIncrement; dtmf->factor++; @@ -250,7 +250,7 @@ void AudioRtpSession::sendDtmfEvent (sfl::DtmfEvent *dtmf) } } -bool AudioRtpSession::onRTPPacketRecv (ost::IncomingRTPPkt&) +bool AudioSymmetricRtpSession::onRTPPacketRecv (ost::IncomingRTPPkt&) { receiveSpeakerData (); @@ -259,7 +259,7 @@ bool AudioRtpSession::onRTPPacketRecv (ost::IncomingRTPPkt&) -void AudioRtpSession::sendMicData() +void AudioSymmetricRtpSession::sendMicData() { int compSize = processDataEncode(); @@ -277,7 +277,7 @@ void AudioRtpSession::sendMicData() } -void AudioRtpSession::receiveSpeakerData () +void AudioSymmetricRtpSession::receiveSpeakerData () { const ost::AppDataUnit* adu = NULL; @@ -304,12 +304,12 @@ void AudioRtpSession::receiveSpeakerData () delete adu; } -int AudioRtpSession::startRtpThread (AudioCodec* audiocodec) +int AudioSymmetricRtpSession::startRtpThread (AudioCodec* audiocodec) { if (_isStarted) return 0; - _debug ("AudioRtpSession: Starting main thread"); + _debug ("AudioSymmetricRtpSession: Starting main thread"); _isStarted = true; setSessionTimeouts(); @@ -323,26 +323,26 @@ int AudioRtpSession::startRtpThread (AudioCodec* audiocodec) return 0; } -void AudioRtpSession::stopRtpThread () +void AudioSymmetricRtpSession::stopRtpThread () { - _debug ("AudioRtpSession: Stoping main thread"); + _debug ("AudioSymmetricRtpSession: Stoping main thread"); _rtpThread->stopRtpThread(); disableStack(); } -AudioRtpSession::AudioRtpThread::AudioRtpThread (AudioRtpSession *session) : rtpSession (session), running (true) +AudioSymmetricRtpSession::AudioRtpThread::AudioRtpThread (AudioSymmetricRtpSession *session) : rtpSession (session), running (true) { - _debug ("AudioRtpSession: Create new rtp thread"); + _debug ("AudioSymmetricRtpSession: Create new rtp thread"); } -AudioRtpSession::AudioRtpThread::~AudioRtpThread() +AudioSymmetricRtpSession::AudioRtpThread::~AudioRtpThread() { - _debug ("AudioRtpSession: Delete rtp thread"); + _debug ("AudioSymmetricRtpSession: Delete rtp thread"); } -void AudioRtpSession::AudioRtpThread::run() +void AudioSymmetricRtpSession::AudioRtpThread::run() { int threadSleep = 20; diff --git a/sflphone-common/src/audio/audiortp/AudioRtpSession.h b/sflphone-common/src/audio/audiortp/AudioSymmetricRtpSession.h similarity index 91% rename from sflphone-common/src/audio/audiortp/AudioRtpSession.h rename to sflphone-common/src/audio/audiortp/AudioSymmetricRtpSession.h index 4acd2c77a6bed5089b352ecb8995edd20b2a188c..8ac08c60dd14d881c1720c22a21687054105c827 100644 --- a/sflphone-common/src/audio/audiortp/AudioRtpSession.h +++ b/sflphone-common/src/audio/audiortp/AudioSymmetricRtpSession.h @@ -54,17 +54,17 @@ using std::ptrdiff_t; namespace sfl { -// class AudioRtpSession : protected ost::Thread, public ost::TimerPort, public AudioRtpRecordHandler, public ost::TRTPSessionBase<ost::DualRTPUDPIPv4Channel,ost::DualRTPUDPIPv4Channel,ost::AVPQueue> -class AudioRtpSession : public ost::TimerPort, public AudioRtpRecordHandler, public ost::SymmetricRTPSession +// class AudioSymmetricRtpSession : protected ost::Thread, public ost::TimerPort, public AudioRtpRecordHandler, public ost::TRTPSessionBase<ost::DualRTPUDPIPv4Channel,ost::DualRTPUDPIPv4Channel,ost::AVPQueue> +class AudioSymmetricRtpSession : public ost::TimerPort, public AudioRtpRecordHandler, public ost::SymmetricRTPSession { public: /** * Constructor * @param sipcall The pointer on the SIP call */ - AudioRtpSession (SIPCall* sipcall); + AudioSymmetricRtpSession (SIPCall* sipcall); - ~AudioRtpSession(); + ~AudioSymmetricRtpSession(); virtual void final (); @@ -111,7 +111,7 @@ class AudioRtpSession : public ost::TimerPort, public AudioRtpRecordHandler, pub class AudioRtpThread : public ost::Thread, public ost::TimerPort { public: - AudioRtpThread (AudioRtpSession *session); + AudioRtpThread (AudioSymmetricRtpSession *session); ~AudioRtpThread(); void stopRtpThread (void) { @@ -121,7 +121,7 @@ class AudioRtpSession : public ost::TimerPort, public AudioRtpRecordHandler, pub virtual void run(); private: - AudioRtpSession *rtpSession; + AudioSymmetricRtpSession *rtpSession; bool running; }; diff --git a/sflphone-common/src/audio/audiortp/AudioZrtpSession.cpp b/sflphone-common/src/audio/audiortp/AudioZrtpSession.cpp index 351e5303e6c3a44c7a630cd9a5cb00f62a258167..c673323cae578b53b63e9f2bd2c8219191ed3e8d 100644 --- a/sflphone-common/src/audio/audiortp/AudioZrtpSession.cpp +++ b/sflphone-common/src/audio/audiortp/AudioZrtpSession.cpp @@ -77,7 +77,7 @@ AudioZrtpSession::AudioZrtpSession (SIPCall * sipcall, const std::string& zidFil AudioZrtpSession::~AudioZrtpSession() { - _debug ("AudioZrtpSession: Delete AudioRtpSession instance"); + _debug ("AudioZrtpSession: Delete AudioSymmetricRtpSession instance"); try { terminate(); @@ -152,7 +152,7 @@ void AudioZrtpSession::setSessionTimeouts (void) void AudioZrtpSession::setSessionMedia (AudioCodec* audioCodec) { - _debug ("AudioRtpSession: Set session media"); + _debug ("AudioSymmetricRtpSession: Set session media"); // set internal codec info for this session setRtpMedia (audioCodec); @@ -176,10 +176,10 @@ void AudioZrtpSession::setSessionMedia (AudioCodec* audioCodec) // Even if specified as a 16 kHz codec, G722 requires rtp sending rate to be 8 kHz if (dynamic) { - _debug ("AudioRtpSession: Setting dynamic payload format"); + _debug ("AudioSymmetricRtpSession: Setting dynamic payload format"); setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) payloadType, smplRate)); } else { - _debug ("AudioRtpSession: Setting static payload format"); + _debug ("AudioSymmetricRtpSession: Setting static payload format"); setPayloadFormat (ost::StaticPayloadFormat ( (ost::StaticPayloadType) payloadType)); } @@ -187,7 +187,7 @@ void AudioZrtpSession::setSessionMedia (AudioCodec* audioCodec) void AudioZrtpSession::updateSessionMedia (AudioCodec *audioCodec) { - _debug ("AudioRtpSession: Update session media"); + _debug ("AudioSymmetricRtpSession: Update session media"); // updateRtpMedia (audioCodec); @@ -204,16 +204,16 @@ void AudioZrtpSession::updateSessionMedia (AudioCodec *audioCodec) _timestampIncrement = frameSize; _debug ("AudioRptSession: Codec payload: %d", payloadType); - _debug ("AudioRtpSession: Codec sampling rate: %d", smplRate); - _debug ("AudioRtpSession: Codec frame size: %d", frameSize); - _debug ("AudioRtpSession: RTP timestamp increment: %d", _timestampIncrement); + _debug ("AudioSymmetricRtpSession: Codec sampling rate: %d", smplRate); + _debug ("AudioSymmetricRtpSession: Codec frame size: %d", frameSize); + _debug ("AudioSymmetricRtpSession: RTP timestamp increment: %d", _timestampIncrement); // Even if specified as a 16 kHz codec, G722 requires rtp sending rate to be 8 kHz if (dynamic) { - _debug ("AudioRtpSession: Setting dynamic payload format"); + _debug ("AudioSymmetricRtpSession: Setting dynamic payload format"); setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) payloadType, smplRate)); } else { - _debug ("AudioRtpSession: Setting static payload format"); + _debug ("AudioSymmetricRtpSession: Setting static payload format"); setPayloadFormat (ost::StaticPayloadFormat ( (ost::StaticPayloadType) payloadType)); } } diff --git a/sflphone-common/src/audio/audiortp/Makefile.am b/sflphone-common/src/audio/audiortp/Makefile.am index fd485065caa2cd157d1fce197e83faa612827ae9..d31503edb899bcfe9487af72f5927ee216f2c063 100644 --- a/sflphone-common/src/audio/audiortp/Makefile.am +++ b/sflphone-common/src/audio/audiortp/Makefile.am @@ -3,7 +3,7 @@ include $(top_srcdir)/globals.mak noinst_LTLIBRARIES = libaudiortp.la libaudiortp_la_SOURCES = \ - AudioRtpSession.cpp \ + AudioSymmetricRtpSession.cpp \ AudioRtpRecordHandler.cpp \ AudioRtpFactory.cpp \ AudioZrtpSession.cpp \ @@ -13,7 +13,7 @@ libaudiortp_la_SOURCES = \ noinst_HEADERS = \ AudioRtpRecordHandler.h \ AudioRtpFactory.h \ - AudioRtpSession.h \ + AudioSymmetricRtpSession.h \ AudioZrtpSession.h \ ZrtpSessionCallback.h \ AudioSrtpSession.h diff --git a/sflphone-common/src/sip/sipvoiplink.cpp b/sflphone-common/src/sip/sipvoiplink.cpp index 30709631100b7d71ecf682898ce1a14876c106ba..76ab7bfe02adaa82d6c73c7a4891a1f39cece9b6 100644 --- a/sflphone-common/src/sip/sipvoiplink.cpp +++ b/sflphone-common/src/sip/sipvoiplink.cpp @@ -716,7 +716,7 @@ Call *SIPVoIPLink::newOutgoingCall (const CallID& id, const std::string& toUrl) try { _info ("UserAgent: Creating new rtp session"); call->getAudioRtp()->initAudioRtpConfig (call); - call->getAudioRtp()->initAudioRtpSession (call); + call->getAudioRtp()->initAudioSymmetricRtpSession (call); call->getAudioRtp()->initLocalCryptoInfo (call); _info ("UserAgent: Start audio rtp session"); call->getAudioRtp()->start (static_cast<sfl::AudioCodec *>(audiocodec)); @@ -1020,7 +1020,7 @@ SIPVoIPLink::offhold (const CallID& id) throw (VoipLinkException) } call->getAudioRtp()->initAudioRtpConfig (call); - call->getAudioRtp()->initAudioRtpSession (call); + call->getAudioRtp()->initAudioSymmetricRtpSession (call); call->getAudioRtp()->start (static_cast<sfl::AudioCodec *>(audiocodec)); } @@ -1790,7 +1790,7 @@ bool SIPVoIPLink::SIPNewIpToIpCall (const CallID& id, const std::string& to) // since SDES require crypto attribute. try { call->getAudioRtp()->initAudioRtpConfig (call); - call->getAudioRtp()->initAudioRtpSession (call); + call->getAudioRtp()->initAudioSymmetricRtpSession (call); call->getAudioRtp()->initLocalCryptoInfo (call); call->getAudioRtp()->start (static_cast<sfl::AudioCodec *>(audiocodec)); } catch (...) { @@ -3495,7 +3495,7 @@ void sdp_media_update_cb (pjsip_inv_session *inv, pj_status_t status) SIPAccount *account = (SIPAccount *) Manager::instance().getAccount (accountID); if (account->getSrtpFallback()) - call->getAudioRtp()->initAudioRtpSession (call); + call->getAudioRtp()->initAudioSymmetricRtpSession (call); } if (!sdpSession) @@ -3972,7 +3972,7 @@ transaction_request_cb (pjsip_rx_data *rdata) try { _debug ("UserAgent: Create RTP session for this call"); call->getAudioRtp()->initAudioRtpConfig (call); - call->getAudioRtp()->initAudioRtpSession (call); + call->getAudioRtp()->initAudioSymmetricRtpSession (call); } catch (...) { _warn ("UserAgent: Error: Failed to create rtp thread from answer"); } diff --git a/sflphone-common/test/rtptest.cpp b/sflphone-common/test/rtptest.cpp index 56afefc3b8a899907b21dc64978312d00cb7596a..446aaef714b49d481577f971f2dfe97d0607e4d6 100644 --- a/sflphone-common/test/rtptest.cpp +++ b/sflphone-common/test/rtptest.cpp @@ -41,7 +41,7 @@ #include <time.h> #include "rtptest.h" -#include "audio/audiortp/AudioRtpSession.h" +#include "audio/audiortp/AudioSymmetricRtpSession.h" #include <unistd.h> diff --git a/sflphone-common/test/rtptest.h b/sflphone-common/test/rtptest.h index 146d79e4c0995eb5a1b19c6f222c84797cfb9aba..7a5faa3b419a9852f1b171a496d4d020f5e8af52 100644 --- a/sflphone-common/test/rtptest.h +++ b/sflphone-common/test/rtptest.h @@ -61,7 +61,7 @@ using namespace sfl; -class AudioRtpSession; +class AudioSymmetricRtpSession; //class AudioRtpFactory; class SIPVoIPLink;