From 59659d29c72a6293c58ccd0dc8793bcbced1dfa9 Mon Sep 17 00:00:00 2001
From: yanmorin <yanmorin>
Date: Thu, 27 Oct 2005 18:37:26 +0000
Subject: [PATCH] Remove some comment Correcting some sound issue Using fix
FRAME_PER_BUFFER is better than 0
---
src/audio/audiolayer.cpp | 18 ++-
src/audio/audiortp.cpp | 235 ++++++++++++++++++---------------------
src/audio/audiortp.h | 1 -
src/audio/ringbuffer.cpp | 8 --
src/audio/ringbuffer.h | 1 -
src/audio/tone.h | 1 +
src/audio/tonelist.cpp | 3 +
src/managerimpl.cpp | 11 +-
src/sipvoiplink.cpp | 16 +--
9 files changed, 133 insertions(+), 161 deletions(-)
diff --git a/src/audio/audiolayer.cpp b/src/audio/audiolayer.cpp
index 8818db811d..bee238e64c 100644
--- a/src/audio/audiolayer.cpp
+++ b/src/audio/audiolayer.cpp
@@ -90,7 +90,7 @@ AudioLayer::openDevice (int index)
// we could put paFramesPerBufferUnspecified instead of FRAME_PER_BUFFER to be variable
portaudio::StreamParameters const params(inParams, outParams,
- SAMPLING_RATE, paFramesPerBufferUnspecified, paNoFlag /*paPrimeOutputBuffersUsingStreamCallback | paNeverDropInput*/);
+ SAMPLING_RATE, FRAME_PER_BUFFER /*paFramesPerBufferUnspecified*/, paNoFlag /*paPrimeOutputBuffersUsingStreamCallback | paNeverDropInput*/);
// Create (and open) a new Stream, using the AudioLayer::audioCallback
_stream = new portaudio::MemFunCallbackStream<AudioLayer>(params,
@@ -103,7 +103,6 @@ AudioLayer::startStream(void)
{
ost::MutexLock guard(_mutex);
if (_stream && !_stream->isActive()) {
- _debug("Thread: start audiolayer stream\n");
_stream->start();
}
}
@@ -113,9 +112,10 @@ AudioLayer::stopStream(void)
{
ost::MutexLock guard(_mutex);
if (_stream && !_stream->isStopped()) {
- _debug("Thread: stop audiolayer stream\n");
- _stream->stop();
+ _stream->stop();
_mainSndRingBuffer.flush();
+ _urgentRingBuffer.flush();
+ _micRingBuffer.flush();
}
}
@@ -225,13 +225,11 @@ AudioLayer::audioCallback (const void *inputBuffer, void *outputBuffer,
toGet = (normalAvail < (int)framesPerBuffer * NBCHARFORTWOINT16) ? normalAvail : framesPerBuffer * NBCHARFORTWOINT16;
if (toGet) {
- _mainSndRingBuffer.Get(out, toGet, spkrVolume);
- } else {
- toGet = framesPerBuffer;
- _mainSndRingBuffer.PutZero(toGet);
- _mainSndRingBuffer.Get(out, toGet, 100);
- }
+ _mainSndRingBuffer.Get(out, toGet, spkrVolume);
+ } else {
+ bzero(out, framesPerBuffer * NBCHARFORTWOINT16);
}
+ }
}
// Additionally handle the mic's audio stream
diff --git a/src/audio/audiortp.cpp b/src/audio/audiortp.cpp
index ef7c1cb793..e44b6a7453 100644
--- a/src/audio/audiortp.cpp
+++ b/src/audio/audiortp.cpp
@@ -47,27 +47,16 @@ AudioRtp::~AudioRtp (void) {
int
AudioRtp::createNewSession (SipCall *ca) {
- // Start RTP Send/Receive threads
- ca->enable_audio = 1;
- if (Manager::instance().getConfigInt(SIGNALISATION,SYMMETRIC)) {
- _symmetric = true;
- } else {
- _symmetric = false;
- }
+ // Start RTP Send/Receive threads
+ ca->enable_audio = 1;
+ _symmetric = Manager::instance().getConfigInt(SIGNALISATION,SYMMETRIC) ? true : false;
+ _RTXThread = new AudioRtpRTX (ca, Manager::instance().getAudioDriver(), _symmetric);
- _RTXThread = new AudioRtpRTX (ca, Manager::instance().getAudioDriver(),
- _symmetric);
-
- // Start PortAudio
- //Manager::instance().getAudioDriver()->flushMic();
- //Manager::instance().getAudioDriver()->startStream();
-
//_debug("AudioRtp::createNewSession: starting RTX thread\n");
- if (_RTXThread->start() != 0) {
- return -1;
- }
-
- return 0;
+ if (_RTXThread->start() != 0) {
+ return -1;
+ }
+ return 0;
}
@@ -91,12 +80,13 @@ AudioRtpRTX::AudioRtpRTX (SipCall *sipcall, AudioLayer* driver, bool sym) : _cod
std::string localipConfig = _ca->getLocalIp();
ost::InetHostAddress local_ip(localipConfig.c_str());
+ _debug("AudioRtpRTX ctor : Local IP:port %s:%d\tsymmetric:%d\n", local_ip.getHostname(), _ca->getLocalAudioPort(), _sym);
+
if (!_sym) {
_sessionRecv = new ost::RTPSession (local_ip, _ca->getLocalAudioPort());
_sessionSend = new ost::RTPSession (local_ip);
_session = NULL;
} else {
- _debug("Symmetric RTP Session on local: %s:%d\n", localipConfig.c_str(), _ca->getLocalAudioPort());
_session = new ost::SymmetricRTPSession (local_ip, _ca->getLocalAudioPort());
_sessionRecv = NULL;
_sessionSend = NULL;
@@ -124,42 +114,38 @@ AudioRtpRTX::~AudioRtpRTX () {
void
AudioRtpRTX::initAudioRtpSession (void)
{
- ost::InetHostAddress remote_ip(_ca->getRemoteSdpAudioIp());
-
- if (!remote_ip) {
- _debug("RTP: Target IP address [%s] is not correct!\n", _ca->getRemoteSdpAudioIp());
- return;
- }
-
- // Initialization
- if (!_sym) {
- //_sessionRecv->setSchedulingTimeout (10000);
- _sessionRecv->setExpireTimeout(1000000);
-
- _sessionSend->setSchedulingTimeout(10000);
- _sessionSend->setExpireTimeout(1000000);
- } else {
- _session->setSchedulingTimeout(10000);
- _session->setExpireTimeout(1000000);
- }
+ ost::InetHostAddress remote_ip(_ca->getRemoteSdpAudioIp());
+ if (!remote_ip) {
+ _debug("RTP: Target IP address [%s] is not correct!\n", _ca->getRemoteSdpAudioIp());
+ return;
+ }
- if (!_sym) {
- if (!_sessionSend->addDestination (remote_ip,
- (unsigned short) _ca->getRemoteSdpAudioPort())) {
- _debug("RTX send: could not connect to port %d\n",
- _ca->getRemoteSdpAudioPort());
- return;
- }
- _debug("RTP(Send): Added sessionSend destination %s:%d\n",
- remote_ip.getHostname(), (unsigned short) _ca->getRemoteSdpAudioPort());
+ // Initialization
+ if (!_sym) {
+ //_sessionRecv->setSchedulingTimeout (10000);
+ _sessionRecv->setExpireTimeout(1000000);
+
+ _sessionSend->setSchedulingTimeout(10000);
+ _sessionSend->setExpireTimeout(1000000);
+ } else {
+ _session->setSchedulingTimeout(10000);
+ _session->setExpireTimeout(1000000);
+ }
+
+ if (!_sym) {
+ if (!_sessionSend->addDestination (remote_ip, (unsigned short) _ca->getRemoteSdpAudioPort())) {
+ _debug("RTX send: could not connect to port %d\n", _ca->getRemoteSdpAudioPort());
+ return;
+ }
+ _debug("RTP(Send): Added sessionSend destination %s:%d\n", remote_ip.getHostname(), (unsigned short) _ca->getRemoteSdpAudioPort());
//setPayloadFormat(StaticPayloadFormat(sptPCMU));
//_debug("Payload Format: %d\n", _ca->payload);
- _sessionRecv->setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) _ca->payload));
- _sessionSend->setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) _ca->payload));
+ _sessionRecv->setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) _ca->payload));
+ _sessionSend->setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) _ca->payload));
- setCancel(cancelImmediate);
_sessionSend->setMark(true);
+ setCancel(cancelImmediate);
} else {
@@ -173,42 +159,46 @@ AudioRtpRTX::initAudioRtpSession (void)
_session->setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) _ca->payload));
setCancel(cancelImmediate);
}
-
- Manager::instance().getAudioDriver()->flushMic();
- Manager::instance().getAudioDriver()->flushMain();
_debug("== AudioRtpRTX::initAudioRtpSession end == \n");
}
void
-// AudioRtpRTX::sendSessionFromMic (unsigned char* data_to_send, int16* data_from_mic, int16* data_from_mic_tmp, int timestamp, int micVolume)
-AudioRtpRTX::sendSessionFromMic (unsigned char* data_to_send, int16* data_from_mic, int16* data_from_mic_tmp, int timestamp)
+AudioRtpRTX::sendSessionFromMic (unsigned char* data_to_send, int16* data_from_mic_stereo, int16* data_from_mic_mono, int timestamp)
{
- int k;
- int compSize;
-
- // Control volume for micro
- int availFromMic = Manager::instance().getAudioDriver()->micRingBuffer().AvailForGet();
+ int availBytesFromMic = Manager::instance().getAudioDriver()->micRingBuffer().AvailForGet();
+ int maxBytesToGet = RTP_FRAMES2SEND * 2 * 2; // * channels * int16/byte
int bytesAvail;
- if (availFromMic < (int)RTP_FRAMES2SEND) {
- bytesAvail = availFromMic;
+
+ // take the lower
+ if (availBytesFromMic < maxBytesToGet) {
+ bytesAvail = availBytesFromMic;
} else {
- bytesAvail = (int)RTP_FRAMES2SEND;
+ bytesAvail = maxBytesToGet;
}
// Get bytes from micRingBuffer to data_from_mic
- Manager::instance().getAudioDriver()->micRingBuffer().Get(data_from_mic, SAMPLES_SIZE(bytesAvail), 100);
+ Manager::instance().getAudioDriver()->startStream();
+ Manager::instance().getAudioDriver()->micRingBuffer().Get(data_from_mic_stereo, bytesAvail, 100);
// control volume and stereo->mono
- for (int j = 0; j < RTP_FRAMES2SEND; j++) {
+ // the j is in int16 RTP_FRAMES2SEND
+ // data_from_mic_mono = 0 to RTP_FRAME2SEND [in int16]
+ for (int j = 0, k=0; j < bytesAvail/4; j++) {
k = j<<1;
- data_from_mic_tmp[j] = (int16)(0.5f*(data_from_mic[k] + data_from_mic[k+1]));
- //micVolume/100);
+ data_from_mic_mono[j] = (int16)(0.5f*(data_from_mic_stereo[k] + data_from_mic_stereo[k+1]));
+ }
+ if ( bytesAvail != maxBytesToGet ) {
+ // fill end with 0...
+ bzero(data_from_mic_mono + (bytesAvail/4), (maxBytesToGet-bytesAvail)/2);
}
if ( _ca != NULL ) {
// Encode acquired audio sample
AudioCodec* ac = _ca->getAudioCodec();
if ( ac != NULL ) {
- compSize = ac->codecEncode (data_to_send, data_from_mic_tmp, RTP_FRAMES2SEND*2);
+ // for the mono: range = 0 to RTP_FRAME2SEND * sizeof(int16)
+ // codecEncode(char *dest, int16* src, size in bytes of the src)
+ int compSize = ac->codecEncode (data_to_send, data_from_mic_mono, RTP_FRAMES2SEND*2);
+ // encode divise by two
// Send encoded audio sample over the network
if (!_sym) {
_sessionSend->putData(timestamp, data_to_send, compSize);
@@ -220,21 +210,18 @@ AudioRtpRTX::sendSessionFromMic (unsigned char* data_to_send, int16* data_from_m
}
void
-AudioRtpRTX::receiveSessionForSpkr (int16* data_for_speakers,
-// int16* data_for_speakers_tmp, int spkrVolume, int& countTime)
- int16* data_for_speakers_tmp, int& countTime)
+AudioRtpRTX::receiveSessionForSpkr (int16* data_for_speakers_stereo, int16* data_for_speakers_recv, int& countTime)
{
- int k;
- const ost::AppDataUnit* adu = NULL;
-
+ const ost::AppDataUnit* adu = NULL;
// Get audio data stream
+
if (!_sym) {
adu = _sessionRecv->getData(_sessionRecv->getFirstTimestamp());
} else {
adu = _session->getData(_session->getFirstTimestamp());
}
if (adu == NULL) {
- Manager::instance().getAudioDriver()->flushMain();
+ //Manager::instance().getAudioDriver()->flushMain();
return;
}
@@ -246,20 +233,25 @@ AudioRtpRTX::receiveSessionForSpkr (int16* data_for_speakers,
int expandedSize = 0;
AudioCodec* ac = _codecBuilder.alloc(payload, "");
if (ac != NULL) {
- expandedSize = ac->codecDecode (data_for_speakers, data, size);
- }
+ // codecDecode(int16 *dest, char* src, size in bytes of the src)
+ // decode multiply by two
+ // size shall be RTP_FRAME2SEND or lower
+ expandedSize = ac->codecDecode(data_for_speakers_recv, data, size);
+ }
ac = NULL;
// control volume for speakers and mono->stereo
- for (int j = 0; j < expandedSize; j++) {
+ // expandedSize is in bytes for data_for_speakers_recv
+ // data_for_speakers_recv are in int16
+ for (int j = 0, k=0; j < expandedSize/2; j++) {
k = j<<1; // fast multiply by two
- data_for_speakers_tmp[k] = data_for_speakers_tmp[k+1] = data_for_speakers[j];
- // * spkrVolume/100;
+ data_for_speakers_stereo[k] = data_for_speakers_stereo[k+1] = data_for_speakers_recv[j];
}
// If the current call is the call which is answered
// Set decoded data to sound device
- Manager::instance().getAudioDriver()->putMain(data_for_speakers_tmp, SAMPLES_SIZE(RTP_FRAMES2SEND));
+ // expandedSize is in mono/bytes, since we double in stereo, we send two time more
+ Manager::instance().getAudioDriver()->putMain(data_for_speakers_stereo, expandedSize*2);
//}
// Notify (with a beep) an incoming call when there is already a call
@@ -270,80 +262,69 @@ AudioRtpRTX::receiveSessionForSpkr (int16* data_for_speakers,
Manager::instance().notificationIncomingCall();
}
}
- Manager::instance().getAudioDriver()->startStream();
-
+
delete adu; adu = NULL;
}
void
AudioRtpRTX::run (void) {
-// int micVolume;
-// int spkrVolume;
- unsigned char *data_to_send;
- int16 *data_from_mic;
- int16 *data_from_mic_tmp;
- int timestamp;
- int16 *data_for_speakers = NULL;
- int16 *data_for_speakers_tmp = NULL;
- int countTime = 0;
-
- data_from_mic = new int16[SIZEDATA];
- data_from_mic_tmp = new int16[SIZEDATA];
- data_to_send = new unsigned char[SIZEDATA];
- data_for_speakers = new int16[SIZEDATA];
- data_for_speakers_tmp = new int16[SIZEDATA*2];
-
- // Init the session
- initAudioRtpSession();
-
- timestamp = 0;
+ //mic, we receive from soundcard in stereo, and we send encoded
+ //encoding before sending
+ int16 *data_from_mic_stereo = new int16[RTP_FRAMES2SEND*2];
+ int16 *data_from_mic_mono = new int16[RTP_FRAMES2SEND];
+ unsigned char *char_to_send = new unsigned char[RTP_FRAMES2SEND]; // two time more for codec
- // TODO: get frameSize from user config
- int frameSize = 20; // 20ms frames
- TimerPort::setTimer(frameSize);
+ //spkr, we receive from rtp in mono and we send in stereo
+ //decoding after receiving
+ int16 *data_for_speakers_recv = new int16[RTP_FRAMES2SEND];
+ int16 *data_for_speakers_stereo = new int16[RTP_FRAMES2SEND*2];
+
+ // Init the session
+ initAudioRtpSession();
// flush stream:
ManagerImpl& manager = Manager::instance();
AudioLayer *audiolayer = manager.getAudioDriver();
- audiolayer->urgentRingBuffer().flush();
- // start running the packet queue scheduler.
+ // start running the packet queue scheduler.
//_debug("Thread: start session of AudioRtpRTX\n");
- if (!_sym) {
- _sessionRecv->startRunning();
- _sessionSend->startRunning();
- } else {
- _session->startRunning();
- }
+ if (!_sym) {
+ _sessionRecv->startRunning();
+ _sessionSend->startRunning();
+ } else {
+ _session->startRunning();
+ _debug("Session is now: %d active?\n", _session->isActive());
+ }
- while (!testCancel() && _ca != NULL && _ca->enable_audio != -1) {
- //micVolume = manager.getMicVolume();
- //spkrVolume = manager.getSpkrVolume();
+ int timestamp = 0; // for mic
+ int countTime = 0; // for receive
+ // TODO: get frameSize from user config
+ int frameSize = 20; // 20ms frames
+ TimerPort::setTimer(frameSize);
+ audiolayer->flushMic();
+ while (!testCancel() && _ca != NULL && _ca->enable_audio != -1) {
////////////////////////////
// Send session
////////////////////////////
- //sendSessionFromMic(data_to_send, data_from_mic, data_from_mic_tmp, timestamp, micVolume);
- sendSessionFromMic(data_to_send, data_from_mic, data_from_mic_tmp, timestamp);
-
+ sendSessionFromMic(char_to_send, data_from_mic_stereo, data_from_mic_mono, timestamp);
timestamp += RTP_FRAMES2SEND;
////////////////////////////
// Recv session
////////////////////////////
- //receiveSessionForSpkr(data_for_speakers, data_for_speakers_tmp, spkrVolume, countTime);
- receiveSessionForSpkr(data_for_speakers, data_for_speakers_tmp, countTime);
+ receiveSessionForSpkr(data_for_speakers_stereo, data_for_speakers_recv, countTime);
// Let's wait for the next transmit cycle
Thread::sleep(TimerPort::getTimer());
TimerPort::incTimer(frameSize); // 'frameSize' ms
}
- delete [] data_for_speakers_tmp; data_for_speakers_tmp = 0;
- delete [] data_for_speakers; data_for_speakers = 0;
- delete [] data_to_send; data_to_send = 0;
- delete [] data_from_mic_tmp; data_from_mic_tmp = 0;
- delete [] data_from_mic; data_from_mic = 0;
+ delete [] data_for_speakers_stereo; data_for_speakers_stereo = 0;
+ delete [] data_for_speakers_recv; data_for_speakers_recv = 0;
+ delete [] char_to_send; char_to_send = 0;
+ delete [] data_from_mic_mono; data_from_mic_mono = 0;
+ delete [] data_from_mic_stereo; data_from_mic_stereo = 0;
audiolayer->stopStream();
}
diff --git a/src/audio/audiortp.h b/src/audio/audiortp.h
index 716296f5eb..6158438811 100644
--- a/src/audio/audiortp.h
+++ b/src/audio/audiortp.h
@@ -29,7 +29,6 @@
#define RTP_FRAMES2SEND 160
-#define SIZEDATA SAMPLES_SIZE(RTP_FRAMES2SEND)
class AudioLayer;
class SipCall;
diff --git a/src/audio/ringbuffer.cpp b/src/audio/ringbuffer.cpp
index 0a18386eb9..68893aad3c 100644
--- a/src/audio/ringbuffer.cpp
+++ b/src/audio/ringbuffer.cpp
@@ -62,14 +62,6 @@ RingBuffer::AvailForPut() const {
return (mBufferSize-4) - Len();
}
-void
-RingBuffer::PutZero(int toZero)
-{
- unsigned char p[toZero];
- bzero(p, toZero);
- Put(p, toZero);
-}
-
// This one puts some data inside the ring buffer.
// Change the volume if it's not 100
int
diff --git a/src/audio/ringbuffer.h b/src/audio/ringbuffer.h
index d474c56a4b..c0ba8027a9 100644
--- a/src/audio/ringbuffer.h
+++ b/src/audio/ringbuffer.h
@@ -44,7 +44,6 @@ class RingBuffer {
//
int AvailForPut (void) const;
int Put (void*, int, unsigned short volume = 100);
- void PutZero(int);
//
// For the reader only:
diff --git a/src/audio/tone.h b/src/audio/tone.h
index 7335844579..9f8c26da10 100644
--- a/src/audio/tone.h
+++ b/src/audio/tone.h
@@ -49,6 +49,7 @@ public:
* @return the number of int16 sent (nb*2)
*/
int getNext(int16* output, int nb, short volume=100);
+ void reset() { _pos = 0; }
private:
/**
diff --git a/src/audio/tonelist.cpp b/src/audio/tonelist.cpp
index 40087c6ca9..2c24f345f9 100644
--- a/src/audio/tonelist.cpp
+++ b/src/audio/tonelist.cpp
@@ -123,6 +123,9 @@ void
TelephoneTone::setCurrentTone(Tone::TONEID toneId)
{
_currentTone = toneId;
+ if ( _currentTone != Tone::TONE_NULL ) {
+ _tone[_currentTone]->reset();
+ }
}
Tone*
diff --git a/src/managerimpl.cpp b/src/managerimpl.cpp
index 6ac5dd3ab2..88c5fc3178 100644
--- a/src/managerimpl.cpp
+++ b/src/managerimpl.cpp
@@ -572,16 +572,14 @@ ManagerImpl::playDtmf(char code)
}
AudioLayer *audiolayer = getAudioDriver();
-
_toneMutex.enterMutex();
- audiolayer->urgentRingBuffer().flush();
// Put buffer to urgentRingBuffer
// put the size in bytes...
// so size * CHANNELS * 2 (bytes for the int16)
int nbInt16InChar = sizeof(int16)/sizeof(char);
int toSend = audiolayer->urgentRingBuffer().AvailForPut();
- if (toSend > size * CHANNELS * nbInt16InChar ) {
+ if (toSend > (size * CHANNELS * nbInt16InChar)) {
toSend = size * CHANNELS * nbInt16InChar;
}
audiolayer->urgentRingBuffer().Put(buf_ctrl_vol, toSend);
@@ -589,14 +587,13 @@ ManagerImpl::playDtmf(char code)
// We activate the stream if it's not active yet.
if (!audiolayer->isStreamActive()) {
audiolayer->startStream();
- audiolayer->sleep(pulselen);
- audiolayer->urgentRingBuffer().flush();
- audiolayer->stopStream();
+ //TODO: Is this really what we want?
+ //audiolayer->sleep(pulselen);
+ //audiolayer->stopStream();
} else {
audiolayer->sleep(pulselen); // in milliseconds
}
_toneMutex.leaveMutex();
- //setZonetone(false);
delete[] buf_ctrl_vol; buf_ctrl_vol = 0;
returnValue = true;
}
diff --git a/src/sipvoiplink.cpp b/src/sipvoiplink.cpp
index 56f384ba9e..ef942fe897 100644
--- a/src/sipvoiplink.cpp
+++ b/src/sipvoiplink.cpp
@@ -487,12 +487,13 @@ SipVoIPLink::onhold (CALLID id)
eXosip_lock ();
// Send request
+ _audiortp.closeRtpSession();
+
i = eXosip_call_send_request (did, invite);
eXosip_unlock ();
// Disable audio
sipcall->enable_audio = false;
- _audiortp.closeRtpSession();
return i;
}
@@ -609,7 +610,8 @@ SipVoIPLink::refuse (CALLID id)
int
SipVoIPLink::getEvent (void)
{
- eXosip_event_t* event = eXosip_event_wait (0, 1);
+ // wait for 0 s, 50 ms
+ eXosip_event_t* event = eXosip_event_wait (0, 50);
eXosip_lock();
eXosip_automatic_action();
eXosip_unlock();
@@ -786,11 +788,11 @@ SipVoIPLink::getEvent (void)
}
break;
case EXOSIP_CALL_RELEASED:
- id = findCallId(event);
- if (id!=0) {
- Manager::instance().peerHungupCall(id);
- deleteSipCall(id);
- }
+ //id = findCallId(event);
+ //if (id!=0) {
+ //Manager::instance().peerHungupCall(id);
+ //deleteSipCall(id);
+ //}
break;
case EXOSIP_CALL_REQUESTFAILURE:
id = findCallId(event);
--
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