diff --git a/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.h b/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.h
index 28ce2ed0882fcb68e665e6f3d5f091462a616d87..aff3f17409828235a79829a865de53f1a404ad83 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.h
+++ b/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.h
@@ -50,6 +50,13 @@ namespace sfl
static const int schedulingTimeout = 10000;
static const int expireTimeout = 100000;
+// G.722 VoIP is typically carried in RTP payload type 9.[2] Note that IANA records the clock rate for type 9 G.722 as 8 kHz
+// (instead of 16 kHz), RFC3551[3] clarifies that this is due to a historical error and is retained in order to maintain backward
+// compatibility. Consequently correct implementations represent the value 8,000 where required but encode and decode audio at 16 kHz.
+static const int g722PayloadType = 9;
+static const int g722RtpClockRate = 8000;
+static const int g722RtpTimeincrement = 160;
+
class AudioRtpSessionException: public std::exception
{
virtual const char* what() const throw() {
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp b/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp
index 37459267195228a32bec33ffd6a93276a6e18bbc..cb2f716bdee008c296596133ff65f81508573623 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp
+++ b/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp
@@ -72,7 +72,7 @@ AudioRtpSession::AudioRtpSession (ManagerImpl * manager, SIPCall * sipcall) :
// static_cast<ost::DualRTPUDPIPv4Channel>(dso)->sendSocket->setTypeOfService(ost::Socket::tosLowDelay);
// static_cast<ost::DualRTPChannel<ost::DualRTPUDPIPv4Channel> >(dso)->sendSocket->setTypeOfService(ost::Socket::tosLowDelay);
- // setTypeOfService(tosEnhanced);
+ setTypeOfService(tosEnhanced);
}
AudioRtpSession::~AudioRtpSession()
@@ -80,7 +80,7 @@ AudioRtpSession::~AudioRtpSession()
_info ("AudioRtpSession: Delete AudioRtpSession instance");
try {
- terminate();
+ terminate();
} catch (...) {
_debugException ("AudioRtpSession: Thread destructor didn't terminate correctly");
throw;
@@ -133,7 +133,7 @@ void AudioRtpSession::setSessionMedia (AudioCodec* audioCodec)
} else if (dynamic) {
_debug ("AudioRtpSession: Setting dynamic payload format");
setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) payloadType, smplRate));
- } else if (dynamic && payloadType != 9) {
+ } else if (dynamic && payloadType != g722PayloadType) {
_debug ("AudioRtpSession: Setting static payload format");
setPayloadFormat (ost::StaticPayloadFormat ( (ost::StaticPayloadType) payloadType));
}
@@ -229,8 +229,6 @@ void AudioRtpSession::sendDtmfEvent (sfl::DtmfEvent *dtmf)
bool AudioRtpSession::onRTPPacketRecv (ost::IncomingRTPPkt&)
{
- // _debug ("AudioRtpSession: onRTPPacketRecv");
-
receiveSpeakerData ();
return true;
@@ -257,7 +255,6 @@ void AudioRtpSession::sendMicData()
// putData put the data on RTP queue, sendImmediate bypass this queue
putData (_timestamp, getMicDataEncoded(), compSize);
- // sendData()
}
@@ -322,6 +319,7 @@ void AudioRtpSession::run ()
// Timestamp must be initialized randomly, already done when instantiating outgoing queue
_timestamp = getCurrentTimestamp();
+ /**
int threadSleep = 0;
if (getCodecSampleRate() != 0) {
@@ -332,6 +330,7 @@ void AudioRtpSession::run ()
}
TimerPort::setTimer (threadSleep);
+ */
// Set recording sampling rate
_ca->setRecordingSmplRate (getCodecSampleRate());
@@ -347,8 +346,6 @@ void AudioRtpSession::run ()
timeout = getSchedulingTimeout();
}
- // timeout = ;
-
_manager->getAudioLayerMutex()->enter();
// Send session
@@ -389,7 +386,6 @@ void AudioRtpSession::run ()
timeout = 0;
}
- // receiveSpeakerData();
}
_debug ("AudioRtpSession: Left main loop for call %s", _ca->getCallId().c_str());
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpSession.h b/sflphone-common/src/audio/audiortp/AudioRtpSession.h
index b948e5cf5a5ab4cd2b868d298e779e743751980a..46e8bc70ab8549eae3adea76273466c3434ce6a0 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpSession.h
+++ b/sflphone-common/src/audio/audiortp/AudioRtpSession.h
@@ -57,13 +57,6 @@
namespace sfl
{
-// G.722 VoIP is typically carried in RTP payload type 9.[2] Note that IANA records the clock rate for type 9 G.722 as 8 kHz
-// (instead of 16 kHz), RFC3551[3] clarifies that this is due to a historical error and is retained in order to maintain backward
-// compatibility. Consequently correct implementations represent the value 8,000 where required but encode and decode audio at 16 kHz.
-static const int g722PayloadType = 9;
-static const int g722RtpClockRate = 8000;
-static const int g722RtpTimeincrement = 160;
-
// class AudioRtpSession : protected ost::Thread, public ost::TimerPort, public AudioRtpRecordHandler, public ost::TRTPSessionBase<ost::DualRTPUDPIPv4Channel,ost::DualRTPUDPIPv4Channel,ost::AVPQueue>
class AudioRtpSession : protected ost::Thread, public ost::TimerPort, public AudioRtpRecordHandler, public ost::TRTPSessionBase<ost::SymmetricRTPChannel, ost::SymmetricRTPChannel, ost::AVPQueue>
{
diff --git a/sflphone-common/src/audio/audiortp/AudioZrtpSession.cpp b/sflphone-common/src/audio/audiortp/AudioZrtpSession.cpp
index 7da895bfcc4d38567d1c7851fefcd0dece8cc6e7..14415cada4c40ab0de203912e04316b759ae6aa0 100644
--- a/sflphone-common/src/audio/audiortp/AudioZrtpSession.cpp
+++ b/sflphone-common/src/audio/audiortp/AudioZrtpSession.cpp
@@ -33,6 +33,7 @@
#include "sip/sipcall.h"
#include "sip/sdp.h"
#include "audio/audiolayer.h"
+#include "manager.h"
#include <libzrtpcpp/ZrtpQueue.h>
#include <libzrtpcpp/ZrtpUserCallback.h>
@@ -64,25 +65,27 @@ AudioZrtpSession::AudioZrtpSession (ManagerImpl * manager, SIPCall * sipcall, co
assert (_ca);
- _info ("AudioZrtpSession: Local audio port %i will be used", _ca->getLocalAudioPort());
+ _info ("AudioZrtpSession: Setting new RTP session with destination %s:%d", _ca->getLocalIp().c_str(), _ca->getLocalAudioPort());
+
+ setTypeOfService(tosEnhanced);
}
AudioZrtpSession::~AudioZrtpSession()
{
_debug ("AudioZrtpSession: Delete AudioRtpSession instance");
- try {
- terminate();
- } catch (...) {
- _debugException ("AudioZrtpSession: Thread destructor didn't terminate correctly");
- throw;
- }
+ try {
+ terminate();
+ } catch (...) {
+ _debugException ("AudioZrtpSession: Thread destructor didn't terminate correctly");
+ throw;
+ }
- _manager->getAudioDriver()->getMainBuffer()->unBindAll (_ca->getCallId());
+ _manager->getAudioDriver()->getMainBuffer()->unBindAll (_ca->getCallId());
- if(_time)
- delete _time;
- _time = NULL;
+ if(_time)
+ delete _time;
+ _time = NULL;
}
@@ -134,12 +137,16 @@ void AudioZrtpSession::initializeZid (void)
void AudioZrtpSession::setSessionTimeouts (void)
{
+ _debug("AudioZrtpSession: Set session scheduling timeout (%d) and expireTimeout (%d)", sfl::schedulingTimeout, sfl::expireTimeout);
+
setSchedulingTimeout (sfl::schedulingTimeout);
setExpireTimeout (sfl::expireTimeout);
}
void AudioZrtpSession::setSessionMedia (AudioCodec* audioCodec)
{
+ _debug("AudioRtpSession: Set session media");
+
// set internal codec info for this session
setRtpMedia(audioCodec);
@@ -150,24 +157,25 @@ void AudioZrtpSession::setSessionMedia (AudioCodec* audioCodec)
bool dynamic = getHasDynamicPayload();
// G722 requires timestamp to be incremented at 8 kHz
- if (payloadType == 9)
- _timestampIncrement = 160;
+ if (payloadType == g722PayloadType)
+ _timestampIncrement = g722RtpTimeincrement;
else
_timestampIncrement = frameSize;
- _debug ("AudioRtpSession: Codec sampling rate: %d", smplRate);
- _debug ("AudioRtpSession: Codec frame size: %d", frameSize);
- _debug ("AudioRtpSession: RTP timestamp increment: %d", _timestampIncrement);
+ _debug ("AudioZrtpSession: Codec payload: %d", payloadType);
+ _debug ("AudioZrtpSession: Codec sampling rate: %d", smplRate);
+ _debug ("AudioZrtpSession: Codec frame size: %d", frameSize);
+ _debug ("AudioZrtpSession: RTP timestamp increment: %d", _timestampIncrement);
// Even if specified as a 16 kHz codec, G722 requires rtp sending rate to be 8 kHz
- if (payloadType == 9) {
- _debug ("AudioRtpSession: Setting G722 payload format");
- setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) payloadType, smplRate));
+ if (payloadType == g722PayloadType) {
+ _debug ("AudioZrtpSession: Setting G722 payload format");
+ setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) payloadType, g722RtpClockRate));
} else if (dynamic) {
- _debug ("AudioRtpSession: Setting dynamic payload format");
+ _debug ("AudioZrtpSession: Setting dynamic payload format");
setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) payloadType, smplRate));
- } else if (dynamic && payloadType != 9) {
- _debug ("AudioRtpSession: Setting static payload format");
+ } else if (dynamic && payloadType != g722PayloadType) {
+ _debug ("AudioZrtpSession: Setting static payload format");
setPayloadFormat (ost::StaticPayloadFormat ( (ost::StaticPayloadType) payloadType));
}
@@ -175,18 +183,18 @@ void AudioZrtpSession::setSessionMedia (AudioCodec* audioCodec)
void AudioZrtpSession::setDestinationIpAddress (void)
{
+ _info ("AudioZrtpSession: Setting IP address for the RTP session");
+
if (_ca == NULL) {
- _error ("AudioRtpSession: Sipcall is gone.");
+ _error ("AudioZrtpSession: Sipcall is gone.");
throw AudioRtpSessionException();
}
- _info ("AudioRtpSession: 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()->get_remote_ip().c_str());
if (!_remote_ip) {
- _warn ("AudioRtpSession: Target IP address (%s) is not correct!",
+ _warn ("AudioZrtpSession: Target IP address (%s) is not correct!",
_ca->getLocalSDP()->get_remote_ip().data());
return;
}
@@ -194,22 +202,24 @@ void AudioZrtpSession::setDestinationIpAddress (void)
// Store remote port in case we would need to forget current destination
_remote_port = (unsigned short) _ca->getLocalSDP()->get_remote_audio_port();
- _info ("AudioRtpSession: New remote address for session: %s:%d",
+ _info ("AudioZrtpSession: New remote address for session: %s:%d",
_ca->getLocalSDP()->get_remote_ip().data(), _remote_port);
if (!addDestination (_remote_ip, _remote_port)) {
- _warn ("AudioRtpSession: Can't add new destination to session!");
+ _warn ("AudioZrtpSession: Can't add new destination to session!");
return;
}
}
void AudioZrtpSession::updateDestinationIpAddress (void)
{
+ _debug("AudioZrtpSession: 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");
+ _warn ("AudioZrtpSession: Could not remove previous destination");
// new destination is stored in call
// we just need to recall this method
@@ -218,7 +228,7 @@ void AudioZrtpSession::updateDestinationIpAddress (void)
void AudioZrtpSession::sendDtmfEvent (sfl::DtmfEvent *dtmf)
{
- _debug ("AudioRtpSession: Send Dtmf %d", getEventQueueSize());
+ _debug ("AudioZrtpSession: Send Dtmf %d", getEventQueueSize());
_timestamp += 160;
@@ -258,9 +268,9 @@ void AudioZrtpSession::sendDtmfEvent (sfl::DtmfEvent *dtmf)
}
}
-bool onRTPPacketRecv (ost::IncomingRTPPkt&)
+bool AudioZrtpSession::onRTPPacketRecv (ost::IncomingRTPPkt&)
{
- _debug ("AudioRtpSession: onRTPPacketRecv");
+ receiveSpeakerData();
return true;
}
@@ -268,10 +278,23 @@ bool onRTPPacketRecv (ost::IncomingRTPPkt&)
void AudioZrtpSession::sendMicData()
{
+ int compSize = processDataEncode();
+
+ // if no data return
+ if(!compSize)
+ return;
+
// Increment timestamp for outgoing packet
_timestamp += _timestampIncrement;
- int compSize = processDataEncode();
+ // Reset timestamp to make sure the timing information are up to date
+ if (_timestampCount > RTP_TIMESTAMP_RESET_FREQ) {
+ _timestamp = getCurrentTimestamp();
+ _timestampCount = 0;
+ }
+
+ // Increment timestamp for outgoing packet
+ _timestamp += _timestampIncrement;
// putData put the data on RTP queue, sendImmediate bypass this queue
putData (_timestamp, getMicDataEncoded(), compSize);
@@ -293,14 +316,8 @@ void AudioZrtpSession::receiveSpeakerData ()
unsigned char* spkrDataIn = NULL;
unsigned int size = 0;
- if (adu) {
-
- spkrDataIn = (unsigned char*) adu->getData(); // data in char
- size = adu->getSize(); // size in char
-
- } else {
- _debug ("AudioRtpSession: No RTP packet available");
- }
+ spkrDataIn = (unsigned char*) adu->getData(); // data in char
+ size = adu->getSize(); // size in char
// DTMF over RTP, size must be over 4 in order to process it as voice data
if (size > 4) {
@@ -310,14 +327,32 @@ void AudioZrtpSession::receiveSpeakerData ()
delete adu;
}
+void AudioZrtpSession::notifyIncomingCall()
+{
+ // Notify (with a beep) an incoming call when there is already a call
+ if (Manager::instance().incomingCallWaiting() > 0) {
+ _countNotificationTime += _time->getSecond();
+ int countTimeModulo = _countNotificationTime % 5000;
+
+ if ( (countTimeModulo - _countNotificationTime) < 0) {
+ Manager::instance().notificationIncomingCall();
+ }
+ _countNotificationTime = countTimeModulo;
+ }
+}
int AudioZrtpSession::startRtpThread (AudioCodec* audiocodec)
{
- _debug ("RTP: Starting main thread");
+ _debug ("AudioZrtpSession: Starting main thread");
setSessionTimeouts();
setSessionMedia (audiocodec);
initBuffers();
- return start (_mainloopSemaphore);
+ initNoiseSuppress();
+ enableStack();
+
+ int ret = start(_mainloopSemaphore);
+
+ return ret;
}
void AudioZrtpSession::run ()
@@ -335,8 +370,6 @@ void AudioZrtpSession::run ()
threadSleep = getAudioLayerFrameSize();
}
- initNoiseSuppress();
-
TimerPort::setTimer (threadSleep);
// Set recording sampling rate
@@ -344,9 +377,8 @@ void AudioZrtpSession::run ()
// Start audio stream (if not started) AND flush all buffers (main and urgent)
_manager->getAudioDriver()->startStream();
- startRunning();
- _debug ("AudioRtpSession: Entering mainloop for call %s",_ca->getCallId().c_str());
+ _debug ("AudioZrtpSession: Entering mainloop for call %s",_ca->getCallId().c_str());
while (!testCancel()) {
@@ -384,7 +416,7 @@ void AudioZrtpSession::run ()
TimerPort::incTimer (threadSleep);
}
- _debug ("AudioRtpSession: Left main loop for call %s", _ca->getCallId().c_str());
+ _debug ("AudioZrtpSession: Left main loop for call %s", _ca->getCallId().c_str());
}
}
diff --git a/sflphone-common/src/audio/audiortp/AudioZrtpSession.h b/sflphone-common/src/audio/audiortp/AudioZrtpSession.h
index 8cc55312239835eae3a0f475d8b5a562743e30cd..91c77019e7691a856e7d71e8e900f6cef0a85abc 100644
--- a/sflphone-common/src/audio/audiortp/AudioZrtpSession.h
+++ b/sflphone-common/src/audio/audiortp/AudioZrtpSession.h
@@ -57,6 +57,8 @@ class AudioZrtpSession : public ost::TimerPort, public ost::SymmetricZRTPSession
// Thread associated method
virtual void run ();
+ virtual bool onRTPPacketRecv (ost::IncomingRTPPkt&);
+
int startRtpThread (AudioCodec*);
/**
@@ -92,6 +94,8 @@ class AudioZrtpSession : public ost::TimerPort, public ost::SymmetricZRTPSession
void setDestinationIpAddress (void);
+ void notifyIncomingCall();
+
/**
* Send encoded data to peer
*/