Commit 2e9a361f authored by Rafaël Carré's avatar Rafaël Carré
Browse files

* #6269: Rename AudioRtpSession to AudioSymmetricRtpSession

parent 1d84bb4d
......@@ -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:
......
......@@ -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;
}
......
......@@ -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
......
......@@ -46,7 +46,7 @@ namespace sfl
{
AudioSrtpSession::AudioSrtpSession (SIPCall * sipcall) :
AudioRtpSession (sipcall),
AudioSymmetricRtpSession (sipcall),
_remoteCryptoCtx (NULL),
_localCryptoCtx (NULL),
_localCryptoSuite (0),
......
......@@ -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:
......
......@@ -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;
......
......@@ -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;
};
......
......@@ -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));
}
}
......
......@@ -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
......
......@@ -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");
}
......
......@@ -41,7 +41,7 @@
#include <time.h>
#include "rtptest.h"
#include "audio/audiortp/AudioRtpSession.h"
#include "audio/audiortp/AudioSymmetricRtpSession.h"
#include <unistd.h>
......
......@@ -61,7 +61,7 @@
using namespace sfl;
class AudioRtpSession;
class AudioSymmetricRtpSession;
//class AudioRtpFactory;
class SIPVoIPLink;
......
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