From b39a205e2060030265f621fc57236f3a39c43cbb Mon Sep 17 00:00:00 2001
From: yanmorin <yanmorin>
Date: Fri, 4 Nov 2005 22:22:09 +0000
Subject: [PATCH] Remove debug message Add changes to project files Fix a
nonat-nonat bug (reinvite)
---
AUTHORS | 1 +
CHANGES | 9 +-
DEPS | 2 +
src/audio/audiolayer.cpp | 56 +++---
src/audio/audiolayer.h | 19 +-
src/audio/audiortp.cpp | 318 ++++++++++++++++---------------
src/audio/ringbuffer.cpp | 2 -
src/audio/tonegenerator.cpp | 5 +-
src/global.h | 16 +-
src/gui/server/tcpsessionio.cpp | 4 -
src/gui/server/tcpstreampool.cpp | 7 -
src/managerimpl.cpp | 74 ++++---
src/sipcall.cpp | 56 +++---
src/sipvoiplink.cpp | 81 ++++----
src/zeroconf/DNSService.cpp | 9 +-
15 files changed, 331 insertions(+), 328 deletions(-)
diff --git a/AUTHORS b/AUTHORS
index 301f9be8ac..3e3ab1355d 100644
--- a/AUTHORS
+++ b/AUTHORS
@@ -16,6 +16,7 @@ Yan Morin
(yan dot morin at savoifairelinux dot com)
- zeroconf integration
- sflphoned deamon
+ - add and improve sip core feature
- tests and debugging
diff --git a/CHANGES b/CHANGES
index e1644f7854..6f1eb920e7 100644
--- a/CHANGES
+++ b/CHANGES
@@ -1,4 +1,11 @@
-SFLphoned/SFLphone (0.5) / 2005-10-17
+SFLphoned/SFLphone (0.6) / 2005-11-04
+ * improve sip protocol
+ * can now receive text message
+ * accept reinvite call
+ * cleaning headers
+ * improving bash launcher script
+
+SFLphoned/SFLphone (0.5a) / 2005-10-17
* sflphoned
* bug fixes
* sflphone-qt
diff --git a/DEPS b/DEPS
index 3308b72d8e..94e54d0478 100644
--- a/DEPS
+++ b/DEPS
@@ -4,3 +4,5 @@ ccRTP 1.3.5: http://sourceforge.net/projects/cplusplus/
libosip 2.2.1: http://savannah.gnu.org/projects/osip/
libeXosip2-1.9.1-pre17: http://www.antisip.com/download/
mDNSResponder87: http://developer.apple.com/darwin/projects/bonjour/
+
+See tools/config.sh
diff --git a/src/audio/audiolayer.cpp b/src/audio/audiolayer.cpp
index 933043f903..f24cb1ef58 100644
--- a/src/audio/audiolayer.cpp
+++ b/src/audio/audiolayer.cpp
@@ -2,17 +2,17 @@
* Copyright (C) 2005 Savoir-Faire Linux inc.
* Author: Yan Morin <yan.morin@savoirfairelinux.com>
* Author: Jerome Oufella <jerome.oufella@savoirfairelinux.com>
- *
+ *
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
- *
+ *
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
- *
+ *
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
@@ -41,7 +41,11 @@ AudioLayer::AudioLayer()
// Destructor
AudioLayer::~AudioLayer (void)
{
- portaudio::System::terminate();
+ try {
+ portaudio::System::terminate();
+ } catch (const portaudio::PaException &e) {
+ _debug("Catch an exception when portaudio tried to terminate\n");
+ }
closeStream();
}
@@ -55,23 +59,12 @@ AudioLayer::closeStream (void)
}
}
-void
-AudioLayer::listDevices()
-{
- ost::MutexLock guard(_mutex);
- portaudio::System::DeviceIterator pos = portaudio::System::instance().devicesBegin();
- while(pos != portaudio::System::instance().devicesEnd()) {
- _debug("AudioLayer: Device (%d) %s\n", pos->index(), pos->name());
- pos++;
- }
-
-}
-
void
AudioLayer::openDevice (int index)
{
closeStream();
+ try {
// Set up the parameters required to open a (Callback)Stream:
portaudio::DirectionSpecificStreamParameters
outParams(portaudio::System::instance().deviceByIndex(index),
@@ -84,7 +77,6 @@ AudioLayer::openDevice (int index)
2, portaudio::INT16, true,
portaudio::System::instance().deviceByIndex(index).defaultLowInputLatency(),
NULL);
-
// we could put paFramesPerBufferUnspecified instead of FRAME_PER_BUFFER to be variable
portaudio::StreamParameters const params(inParams, outParams,
@@ -95,32 +87,46 @@ AudioLayer::openDevice (int index)
_stream = new portaudio::MemFunCallbackStream<AudioLayer>(params,
*this,
&AudioLayer::audioCallback);
+ } catch(...) {
+ throw;
+ }
}
void
AudioLayer::startStream(void)
{
- ost::MutexLock guard(_mutex);
- if (_stream && !_stream->isActive()) {
- _debug("starting stream...\n");
- _stream->start();
+ try {
+ ost::MutexLock guard(_mutex);
+ if (_stream && !_stream->isActive()) {
+ //_debug("Starting sound stream\n");
+ _stream->start();
+ }
+ } catch (const portaudio::PaException &e) {
+ _debugException("Portaudio error: error on starting audiolayer stream");
+ throw;
+ } catch(...) {
+ _debugException("stream start error");
+ throw;
}
}
void
AudioLayer::stopStream(void)
{
- ost::MutexLock guard(_mutex);
try {
+ ost::MutexLock guard(_mutex);
if (_stream && !_stream->isStopped()) {
- _debug("stopping stream...\n");
_stream->stop();
_mainSndRingBuffer.flush();
_urgentRingBuffer.flush();
_micRingBuffer.flush();
}
- } catch (...) {
- _debug("Portaudio error: error when stoping audiolayer stream\n");
+ } catch (const portaudio::PaException &e) {
+ _debugException("Portaudio error: error on stoping audiolayer stream");
+ throw;
+ } catch(...) {
+ _debugException("stream stop error");
+ throw;
}
}
diff --git a/src/audio/audiolayer.h b/src/audio/audiolayer.h
index b444ed8370..8ba2d34370 100644
--- a/src/audio/audiolayer.h
+++ b/src/audio/audiolayer.h
@@ -40,16 +40,15 @@ class RingBuffer;
class AudioLayer {
public:
- AudioLayer();
- ~AudioLayer (void);
-
- void listDevices();
- void openDevice (int);
- void startStream (void);
- void stopStream (void);
- void sleep (int);
- bool isStreamActive (void);
- bool isStreamStopped (void);
+ AudioLayer();
+ ~AudioLayer(void);
+
+ void openDevice(int);
+ void startStream(void);
+ void stopStream(void);
+ void sleep(int);
+ bool isStreamActive(void);
+ bool isStreamStopped(void);
void flushMain();
int putMain(void* buffer, int toCopy);
diff --git a/src/audio/audiortp.cpp b/src/audio/audiortp.cpp
index bba9342fb7..992342192f 100644
--- a/src/audio/audiortp.cpp
+++ b/src/audio/audiortp.cpp
@@ -37,12 +37,13 @@
////////////////////////////////////////////////////////////////////////////////
// AudioRtp
////////////////////////////////////////////////////////////////////////////////
-AudioRtp::AudioRtp () {
- _RTXThread = 0;
+AudioRtp::AudioRtp ()
+{
+ _RTXThread = 0;
}
AudioRtp::~AudioRtp (void) {
- delete _RTXThread; _RTXThread = 0;
+ delete _RTXThread; _RTXThread = 0;
}
int
@@ -51,7 +52,7 @@ AudioRtp::createNewSession (SipCall *ca) {
// something should stop the thread before...
if ( _RTXThread != 0 ) {
- _debug("AudioRTP error: try to create a new audio rtp thread...\n");
+ _debug("Try to create a new audio rtp thread...\n");
return -1;
}
@@ -59,14 +60,13 @@ AudioRtp::createNewSession (SipCall *ca) {
_symmetric = Manager::instance().getConfigInt(SIGNALISATION,SYMMETRIC) ? true : false;
_RTXThread = new AudioRtpRTX (ca, Manager::instance().getAudioDriver(), _symmetric);
- _debug("AudioRtp::createNewSession: starting RTX thread\n");
try {
if (_RTXThread->start() != 0) {
-// if (_RTXThread->start() != 0) {
return -1;
}
} catch(...) {
- _debug("exception on start?");
+ _debugException("AudioRTP try to start a thread");
+ throw;
}
return 0;
}
@@ -74,10 +74,15 @@ AudioRtp::createNewSession (SipCall *ca) {
void
AudioRtp::closeRtpSession () {
+ ost::MutexLock m(_threadMutex);
// This will make RTP threads finish.
- _debug("waiting start signal...\n");
- _debug("receive start signal...");
- delete _RTXThread; _RTXThread = 0;
+ // _debug("Stopping AudioRTP\n");
+ try {
+ delete _RTXThread; _RTXThread = 0;
+ } catch(...) {
+ _debugException("Exception when stopping audiortp\n");
+ throw;
+ }
}
////////////////////////////////////////////////////////////////////////////////
@@ -113,7 +118,7 @@ AudioRtpRTX::~AudioRtpRTX () {
try {
this->terminate();
} catch(...) {
- _debug("audiortprtx dtor catches an exception\n");
+ _debugException("AudioRTP Thread destructor didn't terminate correctly");
throw;
}
//_debug("terminate audiortprtx ended...\n");
@@ -133,58 +138,52 @@ void
AudioRtpRTX::initAudioRtpSession (void)
{
try {
- //_debug("Init audio RTP session\n");
- 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);
- }
-
- if (!_sym) {
- if ( !_sessionRecv->addDestination(remote_ip, (unsigned short) _ca->getRemoteSdpAudioPort()) ) {
- _debug("RTX recv: could not connect to port %d\n", _ca->getLocalAudioPort());
+ //_debug("Init audio RTP session\n");
+ ost::InetHostAddress remote_ip(_ca->getRemoteSdpAudioIp());
+ if (!remote_ip) {
+ _debug("AudioRTP Thread Error: Target IP address [%s] is not correct!\n", _ca->getRemoteSdpAudioIp());
return;
}
- if (!_sessionSend->addDestination (remote_ip, (unsigned short) _ca->getRemoteSdpAudioPort())) {
- _debug("RTX send: could not connect to port %d\n", _ca->getRemoteSdpAudioPort());
- return;
+
+ // Initialization
+ if (!_sym) {
+ _sessionRecv->setSchedulingTimeout (10000);
+ _sessionRecv->setExpireTimeout(1000000);
+
+ _sessionSend->setSchedulingTimeout(10000);
+ _sessionSend->setExpireTimeout(1000000);
+ } else {
+ _session->setSchedulingTimeout(10000);
+ _session->setExpireTimeout(1000000);
}
- _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));
+ if (!_sym) {
+ if ( !_sessionRecv->addDestination(remote_ip, (unsigned short) _ca->getRemoteSdpAudioPort()) ) {
+ _debug("AudioRTP Thread Error: could not connect to port %d\n", _ca->getLocalAudioPort());
+ return;
+ }
+ if (!_sessionSend->addDestination (remote_ip, (unsigned short) _ca->getRemoteSdpAudioPort())) {
+ _debug("AudioRTP Thread Error: could not connect to port %d\n", _ca->getRemoteSdpAudioPort());
+ return;
+ }
+ //_debug("AudioRTP Thread: Added sessionSend destination %s:%d\n", remote_ip.getHostname(), (unsigned short) _ca->getRemoteSdpAudioPort());
- _sessionSend->setMark(true);
- //setCancel(cancelImmediate);
- } else {
+ _sessionRecv->setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) _ca->payload));
+ _sessionSend->setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) _ca->payload));
- _debug("RTP(Send): Added session destination %s:%d\n",
- remote_ip.getHostname(), (unsigned short) _ca->getRemoteSdpAudioPort());
+ _sessionSend->setMark(true);
+ } else {
- if (!_session->addDestination (remote_ip, (unsigned short) _ca->getRemoteSdpAudioPort())) {
- return;
- }
+ //_debug("AudioRTP Thread: Added session destination %s:%d\n", remote_ip.getHostname(), (unsigned short) _ca->getRemoteSdpAudioPort());
- _session->setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) _ca->payload));
- //setCancel(cancelImmediate);
- }
- _debug("== AudioRtpRTX::initAudioRtpSession end == \n");
+ if (!_session->addDestination (remote_ip, (unsigned short) _ca->getRemoteSdpAudioPort())) {
+ return;
+ }
+
+ _session->setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) _ca->payload));
+ }
} catch(...) {
- _debug("initAudioRtpSession catch an exception\n");
+ _debugException("AudioRTP initialisation failed");
throw;
}
}
@@ -192,108 +191,115 @@ AudioRtpRTX::initAudioRtpSession (void)
void
AudioRtpRTX::sendSessionFromMic (unsigned char* data_to_send, int16* data_from_mic_stereo, int16* data_from_mic_mono, int timestamp)
{
- int availBytesFromMic = Manager::instance().getAudioDriver()->canGetMic();
- int maxBytesToGet = RTP_FRAMES2SEND * 2 * 2; // * channels * int16/byte
- int bytesAvail;
-
- // take the lower
- if (availBytesFromMic < maxBytesToGet) {
- bytesAvail = availBytesFromMic;
- } else {
- bytesAvail = maxBytesToGet;
- }
-
- // Get bytes from micRingBuffer to data_from_mic
- Manager::instance().getAudioDriver()->getMic(data_from_mic_stereo, bytesAvail);
- // control volume and stereo->mono
- // 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_mono[j] = (int16)(0.5f*(data_from_mic_stereo[k] + data_from_mic_stereo[k+1]));
- }
- if ( bytesAvail != maxBytesToGet ) {
- // fill end with 0...
- //_debug("Padding mic: %d bytes\n", (maxBytesToGet-bytesAvail)/2);
- bzero(data_from_mic_mono + (bytesAvail/4), (maxBytesToGet-bytesAvail)/2);
- }
-
- if ( _ca != NULL ) {
- // Encode acquired audio sample
- AudioCodec* ac = _ca->getAudioCodec();
- if ( ac != NULL ) {
- // 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
- //fprintf(stderr, "m");
- if (!_sym) {
- _sessionSend->putData(timestamp, data_to_send, compSize);
- } else {
- _session->putData(timestamp, data_to_send, compSize);
+ try {
+ int availBytesFromMic = Manager::instance().getAudioDriver()->canGetMic();
+ int maxBytesToGet = RTP_FRAMES2SEND * 2 * 2; // * channels * int16/byte
+ int bytesAvail;
+
+ // take the lower
+ if (availBytesFromMic < maxBytesToGet) {
+ bytesAvail = availBytesFromMic;
+ } else {
+ bytesAvail = maxBytesToGet;
+ }
+
+ // Get bytes from micRingBuffer to data_from_mic
+ Manager::instance().getAudioDriver()->getMic(data_from_mic_stereo, bytesAvail);
+ // control volume and stereo->mono
+ // 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_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 ) {
+ // 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);
+ } else {
+ _session->putData(timestamp, data_to_send, compSize);
+ }
+ } else {
+ _debug("Exception: No AudioCodec for the mic\n");
}
- } else {
- _debug("No AudioCodec for the mic\n");
}
+ } catch(...) {
+ _debugException("AudioRTP sending failed");
+ throw;
}
}
void
AudioRtpRTX::receiveSessionForSpkr (int16* data_for_speakers_stereo, int16* data_for_speakers_recv, int& countTime)
{
- 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) {
- return;
- }
-
- int payload = adu->getType();
- unsigned char* data = (unsigned char*)adu->getData();
- unsigned int size = adu->getSize();
-
- // Decode data with relevant codec
- int expandedSize = 0;
- AudioCodec* ac = _codecBuilder.alloc(payload, "");
- if (ac != NULL) {
- // 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
- // 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_stereo[k] = data_for_speakers_stereo[k+1] = data_for_speakers_recv[j];
+ try {
+ 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) {
+ return;
+ }
+
+ int payload = adu->getType();
+ unsigned char* data = (unsigned char*)adu->getData();
+ unsigned int size = adu->getSize();
+
+ // Decode data with relevant codec
+ int expandedSize = 0;
+ AudioCodec* ac = _codecBuilder.alloc(payload, "");
+ if (ac != NULL) {
+ // 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
+ // 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_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
+ // 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);
+ //Manager::instance().getAudioDriver()->startStream();
+
+ // Notify (with a beep) an incoming call when there is already a call
+ countTime += time->getSecond();
+ if (Manager::instance().incomingCallWaiting() > 0) {
+ countTime = countTime % 500; // more often...
+ if (countTime == 0) {
+ Manager::instance().notificationIncomingCall();
+ }
+ }
+
+ delete adu; adu = NULL;
+ } catch(...) {
+ _debugException("AudioRTP receiving failed");
+ throw;
}
-
- // If the current call is the call which is answered
- // Set decoded data to sound device
- // expandedSize is in mono/bytes, since we double in stereo, we send two time more
- //fprintf(stderr, "r");
- Manager::instance().getAudioDriver()->putMain(data_for_speakers_stereo, expandedSize*2);
- Manager::instance().getAudioDriver()->startStream();
-
- // Notify (with a beep) an incoming call when there is already a call
- countTime += time->getSecond();
- if (Manager::instance().incomingCallWaiting() > 0) {
- countTime = countTime % 500; // more often...
- if (countTime == 0) {
- Manager::instance().notificationIncomingCall();
- }
- }
-
- delete adu; adu = NULL;
}
void
@@ -310,33 +316,33 @@ AudioRtpRTX::run () {
int16 *data_for_speakers_stereo = new int16[RTP_FRAMES2SEND*2];
try {
- _debug("Audio RtpRTX is running\n");
+ //_debug("AudioRTP Thread is running\n");
// Init the session
initAudioRtpSession();
- _start.post();
// flush stream:
AudioLayer *audiolayer = Manager::instance().getAudioDriver();
// start running the packet queue scheduler.
- _debug("Thread: start session of AudioRtpRTX\n");
+ //_debug("AudioRTP Thread started\n");
if (!_sym) {
_sessionRecv->startRunning();
_sessionSend->startRunning();
} else {
_session->startRunning();
- _debug("Session is now: %d active?\n", _session->isActive());
+ //_debug("Session is now: %d active\n", _session->isActive());
}
-
+
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();
audiolayer->startStream();
+ _start.post();
while (!testCancel()) {
////////////////////////////
// Send session
@@ -356,10 +362,12 @@ AudioRtpRTX::run () {
//_debug("stop stream for audiortp loop\n");
audiolayer->stopStream();
} catch(std::exception &e) {
- _debug("AudioRTP Thread, run function, has caught an exception here: %s\n", e.what());
+ _start.post();
+ _debug("AudioRTP Thread, run: %s\n", e.what());
throw;
} catch(...) {
- _debug("AudioRTP Thread, run function, has caught an unknown exception here\n");
+ _start.post();
+ _debugException("AudioRTP Thread, run()");
throw;
}
delete [] data_for_speakers_stereo; data_for_speakers_stereo = 0;
diff --git a/src/audio/ringbuffer.cpp b/src/audio/ringbuffer.cpp
index 68893aad3c..fe8f49b9b1 100644
--- a/src/audio/ringbuffer.cpp
+++ b/src/audio/ringbuffer.cpp
@@ -143,8 +143,6 @@ RingBuffer::Get(void *buffer, int toCopy, unsigned short volume) {
}
// bcopy(src, dest, len)
bcopy (mBuffer + mStart, dest, block);
- //fprintf(stderr, "has %d get %d\t", len, block);
- //_debug("get %d chars at address %ld, mBufferSize=%d, toCopy=%d\n", block, mBuffer+mStart, mBufferSize, toCopy);
dest += block;
mStart = (mStart + block) % mBufferSize;
toCopy -= block;
diff --git a/src/audio/tonegenerator.cpp b/src/audio/tonegenerator.cpp
index 4b3b4057b9..22c6458004 100644
--- a/src/audio/tonegenerator.cpp
+++ b/src/audio/tonegenerator.cpp
@@ -47,7 +47,7 @@ ToneThread::~ToneThread (void) {
try {
terminate();
} catch (...) {
- _debug("ToneThread: try to terminate, but catch an exception...\n");
+ _debugException("ToneThread: didn't terminate correctly");
}
delete[] buf_ctrl_vol; buf_ctrl_vol=NULL;
}
@@ -79,7 +79,6 @@ ToneThread::run (void) {
// int16 are the buf_ctrl_vol
// unsigned char are the sample_ptr inside ringbuffer
int size_in_char = _size * 2 * (sizeof(int16)/sizeof(unsigned char));
- //_debug(" size : %d\t size_in_char : %d\n", _size, size_in_char);
while (!testCancel()) {
manager.getAudioDriver()->putMain(buf_ctrl_vol, size_in_char);
@@ -331,8 +330,6 @@ ToneGenerator::toneHandle (unsigned int idr, const std::string& zone) {
void
ToneGenerator::stopTone() {
- //_currentTone = ZT_TONE_NULL;
-
//_debug("Thread: delete tonethread\n");
delete tonethread; tonethread = NULL;
// we end the last thread
diff --git a/src/global.h b/src/global.h
index 22b59e440c..9ec66d35e5 100644
--- a/src/global.h
+++ b/src/global.h
@@ -27,22 +27,20 @@
typedef float float32;
typedef short int16;
-#define DEBUG
-#define DEBUG_LEVEL
+//#define DEBUG
#ifdef DEBUG
- #define _debug(...) fprintf(stderr, "[sflphoned] " __VA_ARGS__)
+ #define _debug(...) fprintf(stderr, "[sflphoned] " __VA_ARGS__)
+ #define _debugException(...) fprintf(stderr, "[sflphoned-exception] " __VA_ARGS__ "\n")
+ #define _debugInit(...) fprintf(stderr, "[sflphoned-init] " __VA_ARGS__ "\n")
#else
#define _debug(...)
-#endif
-#ifdef DEBUG_LEVEL
- #define _debugInit(...) fprintf(stderr, "[sflphoned-init] " __VA_ARGS__ "\n")
-#else
+ #define _debugException(...)
#define _debugInit(...)
#endif
-#define SFLPHONED_VERSION "0.5"
-#define SFLPHONED_VERSIONNUM 0x000500
+#define SFLPHONED_VERSION "0.6"
+#define SFLPHONED_VERSIONNUM 0x000600
#define PROGNAME "sflphoned"
#define PROGNAME_GLOBAL "sflphone"
diff --git a/src/gui/server/tcpsessionio.cpp b/src/gui/server/tcpsessionio.cpp
index 8d7559e9cf..dbf0563a89 100644
--- a/src/gui/server/tcpsessionio.cpp
+++ b/src/gui/server/tcpsessionio.cpp
@@ -17,7 +17,6 @@
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#include "tcpsessionio.h"
-#include <cstdio>
const int TCPSessionIO::PORT = 3999;
const char * const TCPSessionIO::IP = "127.0.0.1";
@@ -38,11 +37,8 @@ TCPSessionIO::TCPSessionIO() : SessionIO()
TCPSessionIO::~TCPSessionIO()
{
- fprintf(stderr, "TCPSessionIO: delete clientStream\n");
delete _clientStream; _clientStream = NULL;
- fprintf(stderr, "TCPSessionIO: delete serverSocket\n");
delete _serverSocket; _serverSocket = NULL;
- fprintf(stderr, "TCPSessionIO: end\n");
}
bool
diff --git a/src/gui/server/tcpstreampool.cpp b/src/gui/server/tcpstreampool.cpp
index 89670835f3..7afd51e131 100644
--- a/src/gui/server/tcpstreampool.cpp
+++ b/src/gui/server/tcpstreampool.cpp
@@ -17,15 +17,12 @@
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#include "tcpstreampool.h"
-#include "../../global.h"
#define WAITING_TIME 2UL
TCPStreamPool::~TCPStreamPool()
{
- _debug("TCPStreamPool terminate\n");
terminate();
- _debug("terminate done\n");
}
void
@@ -61,8 +58,6 @@ void
TCPStreamPool::sendLast() {
std::string output;
while (good() && _outputPool.pop(output, WAITING_TIME)) {
- _debug("TCPStreamPool send last %s\n", output.c_str());
- //_debug("sending last message...\n");
*this << output << std::endl;
}
}
@@ -72,5 +67,3 @@ TCPStreamPool::receive(std::string& request)
{
return _inputPool.pop(request, WAITING_TIME);
}
-
-
diff --git a/src/managerimpl.cpp b/src/managerimpl.cpp
index 52e9839ef0..4e6615e667 100644
--- a/src/managerimpl.cpp
+++ b/src/managerimpl.cpp
@@ -101,8 +101,6 @@ ManagerImpl::~ManagerImpl (void)
delete _DNSService; _DNSService = NULL;
#endif
- //delete _error; _error = NULL;
-
_debug("%s stop correctly.\n", PROGNAME);
}
@@ -203,7 +201,7 @@ ManagerImpl::pushBackNewCall (CALLID id, enum CallType type)
Call*
ManagerImpl::getCall (CALLID id)
{
- _debug("%10d: Getting call\n", id);
+ //_debug("%10d: Getting call\n", id);
Call* call = NULL;
unsigned int size = _callVector.size();
for (unsigned int i = 0; i < size; i++) {
@@ -223,7 +221,7 @@ ManagerImpl::getCall (CALLID id)
void
ManagerImpl::deleteCall (CALLID id)
{
- _debug("%10d: Deleting call\n", id);
+ //_debug("%10d: Deleting call\n", id);
CallVector::iterator iter = _callVector.begin();
while(iter!=_callVector.end()) {
Call *call = *iter;
@@ -243,7 +241,7 @@ ManagerImpl::deleteCall (CALLID id)
void
ManagerImpl::setCurrentCallId(CALLID id)
{
- _debug("%10d: Setting current callid, old one was: %d\n", id, _currentCallId);
+ //_debug("%10d: Setting current callid, old one was: %d\n", id, _currentCallId);
_currentCallId = id;
}
@@ -251,7 +249,7 @@ void
ManagerImpl::removeCallFromCurrent(CALLID id)
{
if ( _currentCallId == id ) {
- _debug("%10d: Setting current callid, old one was: %d\n", 0, _currentCallId);
+ //_debug("%10d: Setting current callid, old one was: %d\n", 0, _currentCallId);
_currentCallId = 0;
}
}
@@ -265,8 +263,9 @@ ManagerImpl::removeCallFromCurrent(CALLID id)
*/
int
ManagerImpl::outgoingCall (const std::string& to)
-{
+{
CALLID id = generateNewCallId();
+ _debug("%10d: Outgoing Call\n", id);
Call *call = pushBackNewCall(id, Outgoing);
ost::MutexLock m(_mutex);
call->setState(Call::Progressing);
@@ -287,22 +286,21 @@ ManagerImpl::hangupCall (CALLID id)
{
_debug("%10d: Hangup Call\n", id);
ost::MutexLock m(_mutex);
- if (id == _currentCallId || _currentCallId == 0) {
- stopTone(); // stop tone, like a 700 error: number not found Not Found
- }
Call* call = getCall(id);
if (call == NULL) {
+ stopTone();
return -1;
}
int result = -1;
if (call->getState() != Call::Error) {
result = call->hangup();
+ } else {
+ stopTone();
}
deleteCall(id);
// current call id or no line selected
if (id == _currentCallId || _currentCallId == 0) {
removeCallFromCurrent(id);
- stopTone(); // stop tone, like a 700 error: number not found Not Found
}
return result;
}
@@ -394,7 +392,11 @@ ManagerImpl::offHoldCall (CALLID id)
int returnValue = call->offHold();
// start audio if it's ok
if (returnValue != -1) {
- getAudioDriver()->startStream();
+ try {
+ getAudioDriver()->startStream();
+ } catch(...) {
+ _debugException("Off hold could not start audio stream");
+ }
}
return returnValue;
}
@@ -483,7 +485,7 @@ bool
ManagerImpl::initRegisterVoIPLink()
{
int returnValue = true;
- _debug("Initiate VoIP Link Registration\n");
+ _debugInit("Initiate VoIP Link Registration\n");
if (_hasTriedToRegister == false) {
if ( _voIPLinkVector.at(DFT_VOIP_LINK)->init() ) {
// we call here, because it's long...
@@ -603,14 +605,15 @@ ManagerImpl::playDtmf(char code)
int nbInt16InChar = sizeof(int16)/sizeof(char);
audiolayer->putUrgent(buf_ctrl_vol, size * CHANNELS * nbInt16InChar);
- // We activate the stream if it's not active yet.
- if (!audiolayer->isStreamActive()) {
- audiolayer->startStream();
- //TODO: Is this really what we want?
- //audiolayer->sleep(pulselen);
- //audiolayer->stopStream();
- } else {
- audiolayer->sleep(pulselen); // in milliseconds
+ try {
+ // We activate the stream if it's not active yet.
+ if (!audiolayer->isStreamActive()) {
+ audiolayer->startStream();
+ } else {
+ audiolayer->sleep(pulselen); // in milliseconds
+ }
+ } catch(...) {
+ _debugException("Portaudio exception when playing a dtmf");
}
delete[] buf_ctrl_vol; buf_ctrl_vol = 0;
returnValue = true;
@@ -635,14 +638,14 @@ void
ManagerImpl::incWaitingCall() {
ost::MutexLock m(_incomingCallMutex);
_nbIncomingWaitingCall++;
- _debug("incWaitingCall: %d\n", _nbIncomingWaitingCall);
+ //_debug("incWaitingCall: %d\n", _nbIncomingWaitingCall);
}
void
ManagerImpl::decWaitingCall() {
ost::MutexLock m(_incomingCallMutex);
_nbIncomingWaitingCall--;
- _debug("decWaitingCall: %d\n", _nbIncomingWaitingCall);
+ //_debug("decWaitingCall: %d\n", _nbIncomingWaitingCall);
}
@@ -968,7 +971,11 @@ ManagerImpl::playATone(Tone::TONEID toneId) {
_telephoneTone->setCurrentTone(toneId);
_toneMutex.leaveMutex();
- getAudioDriver()->startStream();
+ try {
+ getAudioDriver()->startStream();
+ } catch(...) {
+ _debugException("Off hold could not start audio stream");
+ }
return true;
}
@@ -977,7 +984,11 @@ ManagerImpl::playATone(Tone::TONEID toneId) {
*/
void
ManagerImpl::stopTone() {
- getAudioDriver()->stopStream();
+ try {
+ getAudioDriver()->stopStream();
+ } catch(...) {
+ _debugException("Stop tone and stop stream");
+ }
_toneMutex.enterMutex();
_telephoneTone->setCurrentTone(Tone::TONE_NULL);
@@ -1033,7 +1044,11 @@ ManagerImpl::ringtone()
_toneMutex.enterMutex();
_audiofile.start();
_toneMutex.leaveMutex();
- getAudioDriver()->startStream();
+ try {
+ getAudioDriver()->startStream();
+ } catch(...) {
+ _debugException("Audio file couldn't start audio stream");
+ }
} else {
ringback();
}
@@ -1104,15 +1119,14 @@ ManagerImpl::getStunInfo (StunAddress4& stunSvrAddr)
if (ok) {
closesocket(fd3);
closesocket(fd4);
- _debug("Got port pair at %d\n", mappedAddr.port);
_firewallPort = mappedAddr.port;
// Convert ipv4 address to host byte ordering
in.s_addr = ntohl (mappedAddr.addr);
addr = inet_ntoa(in);
_firewallAddr = std::string(addr);
- _debug("address firewall = %s\n",_firewallAddr.data());
+ _debug("STUN Firewall: [%s:%d]\n", _firewallAddr.data(), _firewallPort);
} else {
- _debug("Opened a stun socket pair FAILED\n");
+ _debug("Opening a stun socket pair failed\n");
}
}
@@ -1263,7 +1277,7 @@ ManagerImpl::selectAudioDriver (void)
if (nbDevice == 0) {
throw std::runtime_error("Portaudio detect no sound card.");
} else if (noDevice >= nbDevice) {
- _debug("Portaudio auto-select device #0 because device #%d is not found\n", noDevice);
+ _debug(" Portaudio auto-select device #0 because device #%d is not found\n", noDevice);
_setupLoaded = false;
noDevice = 0;
}
diff --git a/src/sipcall.cpp b/src/sipcall.cpp
index c4c8aea721..5909e4d429 100644
--- a/src/sipcall.cpp
+++ b/src/sipcall.cpp
@@ -163,7 +163,7 @@ SipCall::newIncomingCall (eXosip_event_t *event) {
if (event->response != NULL) {
_status_code = event->response->status_code;
snprintf (_reason_phrase, 49, "%s", event->response->reason_phrase);
- _debug(" Status: %d %s\n", _status_code, _reason_phrase);
+ _debug(" Status: %d %s\n", _status_code, _reason_phrase);
}
strcpy(_remote_uri, "");
@@ -192,9 +192,9 @@ SipCall::newIncomingCall (eXosip_event_t *event) {
osip_from_free(from);
}
}
- _debug(" Name: %s\n", _name.c_str());
- _debug(" Number: %s\n", _number.c_str());
- _debug(" Remote URI: %s\n", _remote_uri);
+ _debug(" Name: %s\n", _name.c_str());
+ _debug(" Number: %s\n", _number.c_str());
+ _debug(" Remote URI: %s\n", _remote_uri);
/* negotiate payloads */
sdp_message_t *remote_sdp = NULL;
@@ -218,7 +218,7 @@ SipCall::newIncomingCall (eXosip_event_t *event) {
eXosip_unlock();
if (conn != NULL && conn->c_addr != NULL) {
snprintf (_remote_sdp_audio_ip, 49, "%s", conn->c_addr);
- _debug(" Remote Audio IP: %s\n", _remote_sdp_audio_ip);
+ _debug(" Remote Audio IP: %s\n", _remote_sdp_audio_ip);
}
// Remote Media Port
@@ -236,7 +236,7 @@ SipCall::newIncomingCall (eXosip_event_t *event) {
return -1;
}
_remote_sdp_audio_port = atoi(remote_med->m_port);
- _debug(" Remote Audio Port: %d\n", _remote_sdp_audio_port);
+ _debug(" Remote Audio Port: %d\n", _remote_sdp_audio_port);
// Remote Payload
char *tmp = NULL;
@@ -253,7 +253,7 @@ SipCall::newIncomingCall (eXosip_event_t *event) {
}
if (tmp != NULL) {
payload = atoi (tmp);
- _debug(" Payload: %d\n", payload);
+ _debug(" Payload: %d\n", payload);
setAudioCodec(_cdv->at(0)->alloc(payload, "")); // codec builder for the mic
} else {
_debug("< Sending 415 Unsupported media type\n");
@@ -284,16 +284,16 @@ SipCall::newIncomingCall (eXosip_event_t *event) {
/* search if stream is sendonly or recvonly */
_remote_sendrecv = sdp_analyse_attribute (remote_sdp, remote_med);
_local_sendrecv = sdp_analyse_attribute (local_sdp, local_med);
- _debug(" Remote SendRecv: %d\n", _remote_sendrecv);
- _debug(" Local SendRecv: %d\n", _local_sendrecv);
+ _debug(" Remote SendRecv: %d\n", _remote_sendrecv);
+ _debug(" Local SendRecv: %d\n", _local_sendrecv);
if (_local_sendrecv == _SENDRECV) {
if (_remote_sendrecv == _SENDONLY) { _local_sendrecv = _RECVONLY; }
else if (_remote_sendrecv == _RECVONLY) { _local_sendrecv = _SENDONLY; }
}
- _debug(" Final Local SendRecv: %d\n", _local_sendrecv);
+ _debug(" Final Local SendRecv: %d\n", _local_sendrecv);
sdp_message_free (local_sdp);
}
- _debug(" < Sending answer 183\n");
+ _debug("< Sending answer 183\n");
if (0 != eXosip_call_send_answer (_tid, 183, answer)) {
_debug("SipCall::newIncomingCall: cannot send 183 progress?\n");
}
@@ -323,7 +323,7 @@ SipCall::newReinviteCall (eXosip_event_t *event) {
if (event->response != NULL) {
_status_code = event->response->status_code;
snprintf (_reason_phrase, 49, "%s", event->response->reason_phrase);
- _debug(" Status: %d %s\n", _status_code, _reason_phrase);
+ _debug(" Status: %d %s\n", _status_code, _reason_phrase);
}
strcpy(_remote_uri, "");
@@ -352,9 +352,9 @@ SipCall::newReinviteCall (eXosip_event_t *event) {
osip_from_free(from);
}
}
- _debug(" Name: %s\n", _name.c_str());
- _debug(" Number: %s\n", _number.c_str());
- _debug(" Remote URI: %s\n", _remote_uri);
+ _debug(" Name: %s\n", _name.c_str());
+ _debug(" Number: %s\n", _number.c_str());
+ _debug(" Remote URI: %s\n", _remote_uri);
/* negotiate payloads */
sdp_message_t *remote_sdp = NULL;
@@ -379,7 +379,7 @@ SipCall::newReinviteCall (eXosip_event_t *event) {
eXosip_unlock();
if (conn != NULL && conn->c_addr != NULL) {
snprintf (_remote_sdp_audio_ip, 49, "%s", conn->c_addr);
- _debug(" Remote Audio IP: %s\n", _remote_sdp_audio_ip);
+ _debug(" Remote Audio IP: %s\n", _remote_sdp_audio_ip);
}
// Remote Media Port
@@ -397,7 +397,7 @@ SipCall::newReinviteCall (eXosip_event_t *event) {
return 0;
}
_remote_sdp_audio_port = atoi(remote_med->m_port);
- _debug(" Remote Audio Port: %d\n", _remote_sdp_audio_port);
+ _debug(" Remote Audio Port: %d\n", _remote_sdp_audio_port);
// Remote Payload
char *tmp = NULL;
@@ -414,7 +414,7 @@ SipCall::newReinviteCall (eXosip_event_t *event) {
}
if (tmp != NULL) {
payload = atoi (tmp);
- _debug(" Payload: %d\n", payload);
+ _debug(" Payload: %d\n", payload);
setAudioCodec(_cdv->at(0)->alloc(payload, "")); // codec builder for the mic
} else {
_debug("< Sending 415 Unsupported media type\n");
@@ -445,13 +445,13 @@ SipCall::newReinviteCall (eXosip_event_t *event) {
/* search if stream is sendonly or recvonly */
_remote_sendrecv = sdp_analyse_attribute (remote_sdp, remote_med);
_local_sendrecv = sdp_analyse_attribute (local_sdp, local_med);
- _debug(" Remote SendRecv: %d\n", _remote_sendrecv);
- _debug(" Local SendRecv: %d\n", _local_sendrecv);
+ _debug(" Remote SendRecv: %d\n", _remote_sendrecv);
+ _debug(" Local SendRecv: %d\n", _local_sendrecv);
if (_local_sendrecv == _SENDRECV) {
if (_remote_sendrecv == _SENDONLY) { _local_sendrecv = _RECVONLY; }
else if (_remote_sendrecv == _RECVONLY) { _local_sendrecv = _SENDONLY; }
}
- _debug(" Final Local SendRecv: %d\n", _local_sendrecv);
+ _debug(" Final Local SendRecv: %d\n", _local_sendrecv);
sdp_message_free (local_sdp);
}
_debug("< Sending answer 200\n");
@@ -528,8 +528,8 @@ SipCall::answeredCall(eXosip_event_t *event) {
osip_free (tmp);
}
}
- _debug(" Status: %d %s\n", _status_code, _reason_phrase);
- _debug(" From URI: %s\n", _remote_uri);
+ _debug(" Status: %d %s\n", _status_code, _reason_phrase);
+ _debug(" From URI: %s\n", _remote_uri);
eXosip_lock ();
{
@@ -606,7 +606,7 @@ SipCall::answeredCall_without_hold (eXosip_event_t *event)
_remote_sdp_audio_port = atoi (remote_med->m_port);
}
eXosip_unlock();
- _debug(" Remote Audio: %s:%d\n", _remote_sdp_audio_ip, _remote_sdp_audio_port);
+ _debug(" Remote Audio: %s:%d\n", _remote_sdp_audio_ip, _remote_sdp_audio_port);
if (_remote_sdp_audio_port > 0 && _remote_sdp_audio_ip[0] != '\0' &&
remote_med != NULL) {
@@ -618,7 +618,7 @@ SipCall::answeredCall_without_hold (eXosip_event_t *event)
setAudioCodec(_cdv->at(0)->alloc(payload, ""));
}
}
- _debug(" Remote Payload: %d\n", payload);
+ _debug(" Remote Payload: %d\n", payload);
if (local_sdp == NULL) {
_debug("SipCall::answeredCall_without_hold: SDP body was probably in the ACK (TODO)\n");
@@ -632,7 +632,7 @@ SipCall::answeredCall_without_hold (eXosip_event_t *event)
if (local_med != NULL && local_med->m_port != NULL) {
audio_port = atoi (local_med->m_port);
}
- _debug(" Local Audio port: %d\n", audio_port);
+ _debug(" Local Audio port: %d\n", audio_port);
if (tmp != NULL && audio_port > 0
&& _remote_sdp_audio_port > 0
@@ -648,8 +648,8 @@ SipCall::answeredCall_without_hold (eXosip_event_t *event)
_local_sendrecv = _SENDONLY;
}
}
- _debug(" Remote Sendrecv: %d\n", _remote_sendrecv);
- _debug(" Local Sendrecv: %d\n", _local_sendrecv);
+ _debug(" Remote Sendrecv: %d\n", _remote_sendrecv);
+ _debug(" Local Sendrecv: %d\n", _local_sendrecv);
}
sdp_message_free (local_sdp);
sdp_message_free (remote_sdp);
diff --git a/src/sipvoiplink.cpp b/src/sipvoiplink.cpp
index ea3b8a647c..14bfc14ba6 100644
--- a/src/sipvoiplink.cpp
+++ b/src/sipvoiplink.cpp
@@ -256,7 +256,6 @@ SipVoIPLink::setUnregister (void)
eXosip_lock();
if (_reg_id > 0) {
- _debug("UNREGISTER\n");
i = eXosip_register_build_register (_reg_id, 0, ®);
}
eXosip_unlock();
@@ -264,20 +263,16 @@ SipVoIPLink::setUnregister (void)
return -1;
}
- if (setAuthentication() == -1) {
- _debug("No authentication\n");
- return -1;
- }
-
eXosip_lock();
+ _debug("< Sending REGISTER (expire=0)\n");
i = eXosip_register_send_register (_reg_id, reg);
if (i == -2) {
- _debug("(unregister) Cannot build registration, check the setup\n");
+ _debug(" Cannot build registration (unregister), check the setup\n");
eXosip_unlock();
return -1;
}
if (i == -1) {
- _debug("(unregister) Registration Failed\n");
+ _debug(" Registration (unregister) Failed\n");
}
eXosip_unlock();
_reg_id = -1;
@@ -314,8 +309,8 @@ SipVoIPLink::outgoingInvite (CALLID id, const std::string& to_url)
}
}
- _debug("From: %s\n", from.data());
- _debug("To: %s\n", to.data());
+ _debug(" From: %s\n", from.data());
+ _debug(" To: %s\n", to.data());
if (manager.getConfigString(SIGNALISATION, PROXY).empty()) {
// If no SIP proxy setting for direct call with only IP address
@@ -361,7 +356,7 @@ SipVoIPLink::answer (CALLID id)
_debug("%10d: Answer call [cid = %d, did = %d]\n", id, getSipCall(id)->getCid(), getSipCall(id)->getDid());
port = getSipCall(id)->getLocalAudioPort();
- _debug("Local audio port: %d\n", port);
+ _debug(" Local audio port: %d\n", port);
osip_message_t *answerMessage = NULL;
SipCall* ca = getSipCall(id);
@@ -393,7 +388,7 @@ SipVoIPLink::answer (CALLID id)
eXosip_unlock();
// Incoming call is answered, start the sound channel.
- _debug("Starting AudioRTP\n");
+ _debug(" Starting AudioRTP\n");
if (_audiortp.createNewSession (getSipCall(id)) < 0) {
_debug("FATAL: Unable to start sound (%s:%d)\n", __FILE__, __LINE__);
i = -1;
@@ -411,7 +406,7 @@ SipVoIPLink::hangup (CALLID id)
int i = 0;
SipCall* sipcall = getSipCall(id);
if (sipcall == NULL) { return -1; }
- _debug("Hang up call [id = %d, cid = %d, did = %d]\n",
+ _debug("%10d: Hang up call [cid = %d, did = %d]\n",
id, sipcall->getCid(), sipcall->getDid());
// Release SIP stack.
eXosip_lock();
@@ -419,7 +414,6 @@ SipVoIPLink::hangup (CALLID id)
eXosip_unlock();
// Release RTP channels
- _debug("Stopping AudioRTP\n");
if (id == Manager::instance().getCurrentCallId()) {
_audiortp.closeRtpSession();
}
@@ -433,7 +427,7 @@ SipVoIPLink::cancel (CALLID id)
{
int i = 0;
SipCall* sipcall = getSipCall(id);
- _debug("Cancel call [id = %d, cid = %d]\n", id, sipcall->getCid());
+ _debug("%10d: Cancel call [cid = %d]\n", id, sipcall->getCid());
// Release SIP stack.
eXosip_lock();
i = eXosip_call_terminate (sipcall->getCid(), -1);
@@ -502,7 +496,6 @@ SipVoIPLink::onhold (CALLID id)
}
// Send request
- _debug("Stopping AudioRTP\n");
_audiortp.closeRtpSession();
eXosip_lock ();
@@ -573,7 +566,7 @@ SipVoIPLink::offhold (CALLID id)
eXosip_unlock ();
// Enable audio
- _debug("Starting AudioRTP\n");
+ _debug(" Starting AudioRTP\n");
if (_audiortp.createNewSession (getSipCall(id)) < 0) {
_debug("FATAL: Unable to start sound (%s:%d)\n", __FILE__, __LINE__);
i = -1;
@@ -665,24 +658,24 @@ SipVoIPLink::getEvent (void)
// Generate id
id = Manager::instance().generateNewCallId();
Manager::instance().pushBackNewCall(id, Incoming);
- _debug(" ID: %d [cid = %d, did = %d]\n",id, event->cid, event->did);
+ _debug("%10d: [cid = %d, did = %d]\n", id, event->cid, event->did);
// Associate an audio port with a call
sipcall = getSipCall(id);
sipcall->setLocalAudioPort(_localPort);
sipcall->setLocalIp(getLocalIpAddress());
- _debug(" Local listening port: %d\n", _localPort);
- _debug(" Local listening IP: %s\n", getLocalIpAddress().c_str());
+ _debug(" Local listening port: %d\n", _localPort);
+ _debug(" Local listening IP: %s\n", getLocalIpAddress().c_str());
if (sipcall->newIncomingCall(event) == 0 ) {
if (Manager::instance().incomingCall(id, sipcall->getName(), sipcall->getNumber()) == -1) {
- Manager::instance().displayError(" Incoming Call Failed");
+ Manager::instance().displayError(" Incoming Call Failed");
deleteSipCall(id);
}
} else {
Manager::instance().peerHungupCall(id);
deleteSipCall(id);
- Manager::instance().displayError(" Incoming Call Failed");
+ Manager::instance().displayError(" Incoming Call Failed");
}
break;
@@ -695,10 +688,10 @@ SipVoIPLink::getEvent (void)
if (id != 0) {
sipcall = getSipCall(id);
if (sipcall != 0) {
- _debug(" Info [id = %d, cid = %d, did = %d], localport=%d\n", id, event->cid, event->did,sipcall->getLocalAudioPort());
+ _debug("%10d: Receive Reinvite [cid = %d, did = %d], localport=%d\n", id, event->cid, event->did,sipcall->getLocalAudioPort());
if ( id == Manager::instance().getCurrentCallId() ) {
- _debug("Stopping AudioRTP\n");
+ Manager::instance().stopTone();
_audiortp.closeRtpSession();
}
sipcall->newReinviteCall(event);
@@ -718,7 +711,7 @@ SipVoIPLink::getEvent (void)
case EXOSIP_CALL_RINGING: // 9 peer call is ringing
id = findCallIdInitial(event);
- _debug("> Receive Call Ringing [id = %d, cid = %d, did = %d]\n", id, event->cid, event->did);
+ _debug("%10d: Receive Call Ringing [cid = %d, did = %d]\n", id, event->cid, event->did);
if (id != 0) {
getSipCall(id)->ringingCall(event);
Manager::instance().peerRingingCall(id);
@@ -734,7 +727,7 @@ SipVoIPLink::getEvent (void)
if ( id != 0) {
sipcall = getSipCall(id);
if ( sipcall != 0 ) {
- _debug("> Receive Call Answer [id = %d, cid = %d, did = %d], localport=%d\n", id, event->cid, event->did, sipcall->getLocalAudioPort());
+ _debug("%10d: Receive Call Answer [cid = %d, did = %d], localport=%d\n", id, event->cid, event->did, sipcall->getLocalAudioPort());
// Answer
if (Manager::instance().callCanBeAnswered(id)) {
@@ -745,9 +738,9 @@ SipVoIPLink::getEvent (void)
if(!Manager::instance().callIsOnHold(id) && Manager::instance().getCurrentCallId()==id) {
// Outgoing call is answered, start the sound channel.
- _debug("Starting AudioRTP\n");
+ _debug(" Starting AudioRTP\n");
if (_audiortp.createNewSession(sipcall) < 0) {
- _debug("FATAL: Unable to start sound (%s:%d)\n",
+ _debug(" FATAL: Unable to start sound (%s:%d)\n",
__FILE__, __LINE__);
returnValue = -1;
}
@@ -755,7 +748,7 @@ SipVoIPLink::getEvent (void)
}
} else {
// Answer to on/off hold to send ACK
- _debug("Answering call\n");
+ _debug(" Answering call\n");
sipcall->answeredCall(event);
}
}
@@ -769,7 +762,7 @@ SipVoIPLink::getEvent (void)
case EXOSIP_CALL_ACK: // 15
id = findCallId(event);
- _debug("> Receive ACK [id = %d, cid = %d, did = %d]\n", id, event->cid, event->did);
+ _debug("%10d: Receive ACK [cid = %d, did = %d]\n", id, event->cid, event->did);
if (id != 0 ) {
sipcall = getSipCall(id);
if(sipcall != 0 ) {
@@ -777,8 +770,10 @@ SipVoIPLink::getEvent (void)
if (sipcall->isReinvite()) {
sipcall->endReinvite();
if(!Manager::instance().callIsOnHold(id) && Manager::instance().getCurrentCallId()==id) {
- _debug("Starting AudioRTP\n");
+ _debug(" Starting AudioRTP\n");
_audiortp.createNewSession(sipcall);
+ } else {
+ _debug(" Didn't start RTP because it's on hold or it's not the current call id\n");
}
}
}
@@ -790,11 +785,10 @@ SipVoIPLink::getEvent (void)
// The peer-user closed the phone call(we received BYE).
case EXOSIP_CALL_CLOSED: // 25
id = findCallId(event);
- _debug("> BYE [id = %d, cid = %d, did = %d]\n", id, event->cid, event->did);
+ _debug("%10d: Receive BYE [cid = %d, did = %d]\n", id, event->cid, event->did);
if (id != 0) {
if (Manager::instance().callCanBeClosed(id)) {
sipcall = getSipCall(id);
- _debug("Stopping AudioRTP\n");
_audiortp.closeRtpSession();
}
Manager::instance().peerHungupCall(id);
@@ -809,17 +803,11 @@ SipVoIPLink::getEvent (void)
id = findCallId(event);
if (id!=0) {
// not supposed to be execute on a current call...
- _debug("send a call failure...");
Manager::instance().callFailure(id);
deleteSipCall(id);
}
}
- //id = findCallId(event);
- //if (id!=0) {
- //Manager::instance().peerHungupCall(id);
- //deleteSipCall(id);
- //}
break;
case EXOSIP_CALL_REQUESTFAILURE:
id = findCallId(event);
@@ -944,7 +932,7 @@ SipVoIPLink::getEvent (void)
// Get the message body
ii = osip_message_get_body(event->request, 0, &body);
if (ii != 0) {
- _debug("Cannot get body in a new EXOSIP_MESSAGE_NEW event\n");
+ _debug(" Cannot get body in a new EXOSIP_MESSAGE_NEW event\n");
returnValue = -1;
break;
}
@@ -1019,11 +1007,9 @@ SipVoIPLink::getEvent (void)
break;
default:
- //Manager::instance().displayErrorText(event->type, "getEvent:default");
returnValue = -1;
break;
}
- //_debug(" : end event : %d / %d\n", event->type, returnValue);
eXosip_event_free(event);
return returnValue;
@@ -1113,7 +1099,6 @@ SipVoIPLink::endSipCalls()
eXosip_unlock();
// Release RTP channels
- _debug("Stopping AudioRTP\n");
_audiortp.closeRtpSession();
delete *iter; *iter = NULL;
}
@@ -1453,12 +1438,12 @@ SipVoIPLink::startCall (CALLID id, const std::string& from, const std::string& t
if (!Manager::instance().useStun()) {
// Set random port for outgoing call if no firewall
setLocalPort(RANDOM_LOCAL_PORT);
- _debug(" Setting local port to random: %d\n",_localPort);
+ _debug(" Setting local port to random: %d\n",_localPort);
} else {
// If use Stun server
if (behindNat() != 0) {
setLocalPort(Manager::instance().getFirewallPort());
- _debug(" Setting local port to firewall port: %d\n", _localPort);
+ _debug(" Setting local port to firewall port: %d\n", _localPort);
} else {
return -1;
}
@@ -1529,10 +1514,10 @@ SipVoIPLink::startCall (CALLID id, const std::string& from, const std::string& t
}
// this is the cid (call id from exosip)
- _debug("< INVITE (%d)\n", id);
+ _debug("%10d: Receive INVITE\n", id);
int cid = eXosip_call_send_initial_invite (invite);
- _debug(" Local IP:port: %s:%d\n", getLocalIpAddress().c_str(), getLocalPort());
- _debug(" Payload: %s\n", media_audio);
+ _debug(" Local IP:port: %s:%d\n", getLocalIpAddress().c_str(), getLocalPort());
+ _debug(" Payload: %s\n", media_audio);
// Keep the cid in case of cancelling
sipcall->setCid(cid);
diff --git a/src/zeroconf/DNSService.cpp b/src/zeroconf/DNSService.cpp
index 03f1a0d96e..f70e79efa6 100644
--- a/src/zeroconf/DNSService.cpp
+++ b/src/zeroconf/DNSService.cpp
@@ -24,7 +24,6 @@
#include "DNSService.h"
#include "DNSServiceTXTRecord.h"
#include "DNSQueryThread.h"
-#include "../global.h" // for _debug()
#include <cc++/thread.h>
@@ -200,10 +199,10 @@ DNSServiceAddServicesCallback(DNSServiceRef,
std::string tempService;
tempService = std::string(serviceName) + "." + std::string(replyType) + std::string(replyDomain);
if (flags&kDNSServiceFlagsAdd) {
- _debug("DNSServiceAddServicesCallback call addService\n");
+// _debug("DNSServiceAddServicesCallback call addService\n");
service->addService(tempService);
} else {
- _debug("DNSServiceAddServicesCallback call removeService\n");
+// _debug("DNSServiceAddServicesCallback call removeService\n");
service->removeService(tempService);
}
}
@@ -228,10 +227,10 @@ DNSServiceQueryRecordCallback(
{
if (errorCode==kDNSServiceErr_NoError) {
if (flags&kDNSServiceFlagsAdd) {
- _debug("DNSServiceQueryRecordCallback call addTXTRecord\n");
+// _debug("DNSServiceQueryRecordCallback call addTXTRecord\n");
((DNSService *)context)->addTXTRecord(fullname, rdlen, rdata);
} else {
- _debug("DNSServiceQueryRecordCallback call removeService\n");
+// _debug("DNSServiceQueryRecordCallback call removeService\n");
((DNSService *)context)->removeService(fullname);
}
}
--
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