diff --git a/sflphone-common/src/audio/audiortp/AudioRtpSession.h b/sflphone-common/src/audio/audiortp/AudioRtpSession.h
index cb0578beaa7a12d70b5efd781fd3fd29b07afcdb..7ecfe5ecb9cb81bac9e042c62f6e531bb4f28da5 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpSession.h
+++ b/sflphone-common/src/audio/audiortp/AudioRtpSession.h
@@ -249,7 +249,7 @@ namespace sfl {
template <typename D>
AudioRtpSession<D>::~AudioRtpSession()
{
- _debug ("Rtp: Delete AudioRtpSession instance");
+ _debug ("RTP: Delete AudioRtpSession instance");
try {
terminate();
@@ -273,11 +273,9 @@ namespace sfl {
void AudioRtpSession<D>::initBuffers()
{
// Set sampling rate, main buffer choose the highest one
- // _audiolayer->getMainBuffer()->setInternalSamplingRate(_codecSampleRate);
_manager->getAudioDriver()->getMainBuffer()->setInternalSamplingRate(_codecSampleRate);
// may be different than one already setted
- // converterSamplingRate = _audiolayer->getMainBuffer()->getInternalSamplingRate();
_converterSamplingRate = _manager->getAudioDriver()->getMainBuffer()->getInternalSamplingRate();
// initialize SampleRate converter using AudioLayer's sampling rate
@@ -311,28 +309,33 @@ namespace sfl {
{
assert(_ca);
+ _debug("RTP: Get audio codec for call %s", _ca->getCallId().c_str());
+
AudioCodecType pl = (AudioCodecType)_ca->getLocalSDP()->get_session_media()->getPayload();
_audiocodec = _manager->getCodecDescriptorMap().instantiateCodec(pl);
- if (_audiocodec == NULL) {
- _debug ("No audiocodec, can't init RTP media");
+ if (!_audiocodec) {
+ _error ("RTP: Error: No audiocodec, can't init RTP media");
throw AudioRtpSessionException();
}
- _debug ("Init audio RTP session: codec payload %i", _audiocodec->getPayload());
+ _debug ("RTP: Init codec payload %i", _audiocodec->getPayload());
_codecSampleRate = _audiocodec->getClockRate();
_codecFrameSize = _audiocodec->getFrameSize();
+ _debug("RTP: Codec sampling rate: %d", _codecSampleRate);
+ _debug("RTP: Codec frame size: %d", _codecFrameSize);
+
//TODO: figure out why this is necessary.
if (_audiocodec->getPayload() == 9) {
- _debug ("Setting payload format to G722");
+ _debug ("RTP: Setting payload format to G722");
static_cast<D*>(this)->setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) _audiocodec->getPayload(), _audiocodec->getClockRate()));
} else if (_audiocodec->hasDynamicPayload()) {
- _debug ("Setting a dynamic payload format");
+ _debug ("RTP: Setting a dynamic payload format");
static_cast<D*>(this)->setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) _audiocodec->getPayload(), _audiocodec->getClockRate()));
} else if (!_audiocodec->hasDynamicPayload() && _audiocodec->getPayload() != 9) {
- _debug ("Setting a static payload format");
+ _debug ("RTP: Setting a static payload format");
static_cast<D*>(this)->setPayloadFormat (ost::StaticPayloadFormat ( (ost::StaticPayloadType) _audiocodec->getPayload()));
}
}
@@ -341,7 +344,7 @@ namespace sfl {
void AudioRtpSession<D>::setDestinationIpAddress(void)
{
if (_ca == NULL) {
- _warn ("Rtp: Sipcall is gone.");
+ _error ("RTP: Sipcall is gone.");
throw AudioRtpSessionException();
}
@@ -351,7 +354,7 @@ namespace sfl {
_remote_ip = ost::InetHostAddress(_ca->getLocalSDP()->get_remote_ip().c_str());
if (!_remote_ip) {
- _warn("Rtp: Target IP address (%s) is not correct!",
+ _warn("RTP: Target IP address (%s) is not correct!",
_ca->getLocalSDP()->get_remote_ip().data());
return;
}
@@ -363,7 +366,7 @@ namespace sfl {
_ca->getLocalSDP()->get_remote_ip().data(), _remote_port);
if (! static_cast<D*>(this)->addDestination (_remote_ip, _remote_port)) {
- _warn("Rtp: Can't add new destination to session!");
+ _warn("RTP: Can't add new destination to session!");
return;
}
}
@@ -375,7 +378,7 @@ namespace sfl {
// This method remove the current destination entry
if(!static_cast<D*>(this)->forgetDestination(_remote_ip, _remote_port, _remote_port+1))
- _warn("Rtp: Could not remove previous destination");
+ _warn("RTP: Could not remove previous destination");
// new destination is stored in call
// we just need to recall this method
@@ -619,7 +622,7 @@ namespace sfl {
template <typename D>
int AudioRtpSession<D>::startRtpThread ()
{
- _debug("Starting main thread");
+ _debug("RTP: Starting main thread");
return start(_mainloopSemaphore);
}
@@ -630,48 +633,50 @@ namespace sfl {
setSessionTimeouts();
setSessionMedia();
- initBuffers();
+ initBuffers();
- // Timestamp must be initialized randomly
- _timestamp = static_cast<D*>(this)->getCurrentTimestamp();
+ // Timestamp must be initialized randomly
+ _timestamp = static_cast<D*>(this)->getCurrentTimestamp();
int sessionWaiting;
int threadSleep = 0;
- if (_codecSampleRate != 0)
- { threadSleep = (_codecFrameSize * 1000) / _codecSampleRate; }
- else
- { threadSleep = _layerFrameSize; }
+ if (_codecSampleRate != 0){
+ threadSleep = (_codecFrameSize * 1000) / _codecSampleRate;
+ }
+ else {
+ threadSleep = _layerFrameSize;
+ }
TimerPort::setTimer (threadSleep);
if (_audiolayer == NULL) {
- _debug("For some unknown reason, audiolayer is null, just as \
- we were about to start the audio stream");
+ _error("RTP: Error: Audiolayer is null, cannot start the audio stream");
throw AudioRtpSessionException();
}
- _ca->setRecordingSmplRate(_audiocodec->getClockRate());
+ _ca->setRecordingSmplRate(_audiocodec->getClockRate());
- // Start audio stream (if not started) AND flush all buffers (main and urgent)
- _manager->getAudioDriver()->startStream();
+ // Start audio stream (if not started) AND flush all buffers (main and urgent)
+ _manager->getAudioDriver()->startStream();
static_cast<D*>(this)->startRunning();
- _debug ("Entering RTP mainloop for callid %s",_ca->getCallId().c_str());
+ _debug ("RTP: Entering mainloop for call %s",_ca->getCallId().c_str());
while (!testCancel()) {
- // ost::MutexLock lock(*(_manager->getAudioLayerMutex()));
+ // ost::MutexLock lock(*(_manager->getAudioLayerMutex()));
- _manager->getAudioLayerMutex()->enter();
+ _manager->getAudioLayerMutex()->enter();
- // converterSamplingRate = _audiolayer->getMainBuffer()->getInternalSamplingRate();
- _manager->getAudioDriver()->getMainBuffer()->getInternalSamplingRate();
+ // converterSamplingRate = _audiolayer->getMainBuffer()->getInternalSamplingRate();
+ _converterSamplingRate = _manager->getAudioDriver()->getMainBuffer()->getInternalSamplingRate();
- // Send session
- sessionWaiting = static_cast<D*>(this)->isWaiting();
+ sessionWaiting = static_cast<D*>(this)->isWaiting();
+
+ // Send session
if(_eventQueue.size() > 0) {
sendDtmfEvent(_eventQueue.front());
}
@@ -698,7 +703,7 @@ namespace sfl {
TimerPort::incTimer (threadSleep);
}
- _debug ("Left RTP main loop for callid %s",_ca->getCallId().c_str());
+ _debug ("RTP: Left main loop for call%s", _ca->getCallId().c_str());
}
}
diff --git a/sflphone-common/src/audio/codecs/Makefile.am b/sflphone-common/src/audio/codecs/Makefile.am
index 230ee7bbc0dce19ecfd5f02f18236afefb84965d..e338dc3fb5f898a86039e174c6705370fc20b91a 100644
--- a/sflphone-common/src/audio/codecs/Makefile.am
+++ b/sflphone-common/src/audio/codecs/Makefile.am
@@ -49,7 +49,7 @@ CELT_LIB = libcodec_celt.so
libcodec_celt_so_SOURCES = celtcodec.cpp
libcodec_celt_so_CFLAGS = -fPIC -g -Wall
libcodec_celt_so_CXXFLAGS = -fPIC -g -Wall
-libcodec_celt_so_LDFLAGS = --shared -lc -lcelt $(CELT_NIMP)
+libcodec_celt_so_LDFLAGS = --shared -lc -lcelt0 $(CELT_NIMP)
INSTALL_CELT_RULE = install-libcodec_celt_so
endif
diff --git a/sflphone-common/src/audio/codecs/celtcodec.cpp b/sflphone-common/src/audio/codecs/celtcodec.cpp
index 8787bd6b6b3f7c8bfeec6fbf31464de24e31816c..f68fce20578cd20b526771baddd64d0f58b7271b 100644
--- a/sflphone-common/src/audio/codecs/celtcodec.cpp
+++ b/sflphone-common/src/audio/codecs/celtcodec.cpp
@@ -26,27 +26,59 @@ class Celt : public AudioCodec
{
public:
- Celt (int payload=0)
- : AudioCodec (payload, "celt") {
- _clockRate = 44100;
- _frameSize = 512; // fixed frameSize, TODO: support variable size from 64 to 512
+ Celt (int payload=0) : AudioCodec (payload, "celt") {
+
+ _clockRate = 32000;
+ _frameSize = 512; // fixed frameSize, TODO: support variable size from 64 to 512
_channel = 1;
_bitrate = 0;
_bandwidth = 0;
+
initCelt();
+
}
Celt (const Celt&);
Celt& operator= (const Celt&);
void initCelt() {
- printf ("init celt");
+ printf ("Celt: Init Celt codec");
+
+ int error = 0;
+
+ _mode = celt_mode_create (_clockRate, _frameSize, &error);
+
+ if(error != CELT_OK) {
+ switch(error) {
+ case CELT_BAD_ARG:
+ printf("Celt: Error: An (or more) invalid argument (e.g. out of range)\n");
+ break;
+ case CELT_INVALID_MODE:
+ printf("Celt: Error: The mode struct passed is invalid\n");
+ break;
+ case CELT_INTERNAL_ERROR:
+ printf("Celt: Error: An internal error was detected\n");
+ break;
+ case CELT_CORRUPTED_DATA:
+ printf("Celt: Error: The data passed (e.g. compressed data to decoder) is corrupted\n");
+ break;
+ case CELT_UNIMPLEMENTED:
+ printf("Celt: Error: Invalid/unsupported request numbe\n");
+ break;
+ case CELT_INVALID_STATE:
+ printf("Celt: Error: An encoder or decoder structure is invalid or already freed\n");
+ break;
+ case CELT_ALLOC_FAIL:
+ printf("Celt: Error: Memory allocation has failed\n");
+ break;
+ default:
+ printf("Celt: Error");
+ }
- mode = celt_mode_create (_clockRate, _channel, _frameSize, NULL);
- // celt_mode_info(mode, CELT_GET_LOOKAHEAD, &skip);
+ }
- if (mode == NULL) {
- printf ("failed to create a mode");
+ if (_mode == NULL) {
+ printf ("Celt: Error: Failed to create Celt mode");
}
// bytes_per_packet = 1024;
@@ -58,11 +90,12 @@ class Celt : public AudioCodec
// celt_mode_info(mode, CELT_GET_FRAME_SIZE, &frame_size);
// celt_mode_info(mode, CELT_GET_NB_CHANNELS, &_channel);
- enc = celt_encoder_create (mode);
+ _enc = celt_encoder_create (_mode, _channel, &error);
- dec = celt_decoder_create (mode);
+ _dec = celt_decoder_create (_mode, _channel, &error);
- celt_encoder_ctl (enc,CELT_SET_COMPLEXITY (10));
+ celt_encoder_ctl (_enc, CELT_SET_COMPLEXITY (10));
+ celt_decoder_ctl(_dec, CELT_SET_COMPLEXITY (10));
}
@@ -72,32 +105,34 @@ class Celt : public AudioCodec
void terminateCelt() {
- celt_encoder_destroy (enc);
- celt_decoder_destroy (dec);
+ celt_encoder_destroy (_enc);
+ celt_decoder_destroy (_dec);
+
+ celt_mode_destroy(_mode);
}
virtual int codecDecode (short *dst, unsigned char *src, unsigned int size) {
int err = 0;
- err = celt_decode (dec, src, size, (celt_int16_t*) dst);
- return _frameSize * sizeof (celt_int16_t);
+ err = celt_decode (_dec, src, size, (celt_int16*) dst);
+ return _frameSize * sizeof (celt_int16);
}
virtual int codecEncode (unsigned char *dst, short *src, unsigned int size) {
int len = 0;
- len = celt_encode (enc, (celt_int16_t *) src, (celt_int16_t *) src, dst, 512);
+ len = celt_encode (_enc, (celt_int16*) src, (celt_int16 *) src, dst, 512);
// returns the number of bytes writen
return len;
}
private:
- CELTMode *mode;
+ CELTMode *_mode;
- CELTEncoder *enc;
- CELTDecoder *dec;
+ CELTEncoder *_enc;
+ CELTDecoder *_dec;
- celt_int32_t _celt_frame_size;
- celt_int32_t skip;
+ celt_int32 _celt_frame_size;
+ celt_int32 skip;
};
diff --git a/sflphone-common/src/audio/codecs/speexcodec_nb.cpp b/sflphone-common/src/audio/codecs/speexcodec_nb.cpp
index a2603fc8271e1430ca664edce67d07e3205b3ad4..9099dca71fbc7a69d71982394794c59f1d95492e 100644
--- a/sflphone-common/src/audio/codecs/speexcodec_nb.cpp
+++ b/sflphone-common/src/audio/codecs/speexcodec_nb.cpp
@@ -39,7 +39,7 @@ class Speex : public AudioCodec
_clockRate = 8000;
_frameSize = 160; // samples, 20 ms at 8kHz
_channel = 1;
- _bitrate = 0;
+ _bitrate = 24;
_bandwidth = 0;
initSpeex();
}
diff --git a/sflphone-common/src/audio/codecs/speexcodec_wb.cpp b/sflphone-common/src/audio/codecs/speexcodec_wb.cpp
index 9d3b618674d016fdd939b08e30e833aa70d47bf0..28e099791f511dd838fdedc5c37dd4e4f20d4265 100644
--- a/sflphone-common/src/audio/codecs/speexcodec_wb.cpp
+++ b/sflphone-common/src/audio/codecs/speexcodec_wb.cpp
@@ -39,7 +39,7 @@ class Speex : public AudioCodec
_clockRate = 16000;
_frameSize = 320; // 20 ms at 16 kHz
_channel = 1;
- _bitrate = 0;
+ _bitrate = 42;
_bandwidth = 0;
initSpeex();
}
diff --git a/sflphone-common/src/dbus/callmanager.cpp b/sflphone-common/src/dbus/callmanager.cpp
index 1c30ef975528dab36b6ed6840dd2bbac18cd7af5..36b5aa87a958aacb1c29046f11dfb860fe278f6e 100644
--- a/sflphone-common/src/dbus/callmanager.cpp
+++ b/sflphone-common/src/dbus/callmanager.cpp
@@ -77,14 +77,14 @@ CallManager::placeCallFirstAccount (const std::string& callID,
void
CallManager::refuse (const std::string& callID)
{
- _debug ("CallManager::refuse received");
+ _debug ("CallManager: refuse received");
Manager::instance().refuseCall (callID);
}
void
CallManager::accept (const std::string& callID)
{
- _debug ("CallManager::accept received");
+ _debug ("CallManager: accept received");
Manager::instance().answerCall (callID);
}
diff --git a/sflphone-common/src/managerimpl.cpp b/sflphone-common/src/managerimpl.cpp
index 5fc21e09a700677d936f2996c0ddfda3288727f8..2bb011d6420cb011e9a9df4e92e7f8daf81a1b5c 100644
--- a/sflphone-common/src/managerimpl.cpp
+++ b/sflphone-common/src/managerimpl.cpp
@@ -376,7 +376,9 @@ bool ManagerImpl::answerCall (const CallID& call_id) {
//THREAD=Main
bool ManagerImpl::hangupCall (const CallID& call_id) {
- _debug ("ManagerImpl::hangupCall(%s)", call_id.c_str());
+
+ _info("Manager: Hangup call %s", call_id.c_str());
+
PulseLayer *pulselayer;
AccountID account_id;
bool returnValue = true;
@@ -387,7 +389,7 @@ bool ManagerImpl::hangupCall (const CallID& call_id) {
stopTone();
/* Broadcast a signal over DBus */
- _debug (" hangupCall: Send DBUS call state change (HUNGUP) for id %s", call_id.c_str());
+ _debug ("Manager: Send DBUS call state change (HUNGUP) for id %s", call_id.c_str());
if (_dbus)
_dbus->getCallManager()->callStateChanged(call_id, "HUNGUP");
@@ -419,7 +421,7 @@ bool ManagerImpl::hangupCall (const CallID& call_id) {
if (account_id == AccountNULL) {
- _debug ("! Manager Hangup Call: Call doesn't exists");
+ _error ("Manager: Error: account id is NULL in hangup");
returnValue = false;
} else {
@@ -434,7 +436,7 @@ bool ManagerImpl::hangupCall (const CallID& call_id) {
// stop streams
if (audiolayer && (nbCalls <= 0)) {
- _debug (" hangupCall: stop audio stream, ther is only %i call(s) remaining", nbCalls);
+ _debug ("Manager: stop audio stream, ther is only %i call(s) remaining", nbCalls);
audiolayer->stopStream();
}
@@ -446,7 +448,8 @@ bool ManagerImpl::hangupCall (const CallID& call_id) {
}
bool ManagerImpl::hangupConference (const ConfID& id) {
- _debug ("ManagerImpl::hangupConference()");
+
+ _debug ("Manager: Hangup conference %s", id.c_str());
Conference *conf;
ConferenceMap::iterator iter_conf = _conferencemap.find(id);
@@ -463,7 +466,7 @@ bool ManagerImpl::hangupConference (const ConfID& id) {
ParticipantSet::iterator iter_participant = participants.begin();
while (iter_participant != participants.end()) {
- _debug ("ManagerImpl::hangupConference participant %s", (*iter_participant).c_str());
+ _debug ("Manager: Hangup onference participant %s", (*iter_participant).c_str());
hangupCall(*iter_participant);
@@ -483,6 +486,8 @@ bool ManagerImpl::cancelCall (const CallID& id) {
AccountID accountid;
bool returnValue;
+ _debug("Manager: Cancel call");
+
stopTone();
/* Direct IP to IP call */
@@ -518,13 +523,12 @@ bool ManagerImpl::onHoldCall (const CallID& call_id) {
AccountID account_id;
bool returnValue;
- _debug ("ManagerImpl::onHoldCall(%s)", call_id.c_str());
+ _debug ("Manager: Put call %s on hold", call_id.c_str());
stopTone();
CallID current_call_id = getCurrentCallId();
- _debug (" onHoldCall: try to put call %s on hold", call_id.c_str());
/* Direct IP to IP call */
@@ -537,7 +541,7 @@ bool ManagerImpl::onHoldCall (const CallID& call_id) {
account_id = getAccountFromCall(call_id);
if (account_id == AccountNULL) {
- _debug (" onHoldCall: Account ID %s or callid %s doesn't exists", account_id.c_str(), call_id.c_str());
+ _debug ("Manager: Account ID %s or callid %s doesn't exists in call onHold", account_id.c_str(), call_id.c_str());
return false;
}
@@ -568,7 +572,7 @@ bool ManagerImpl::offHoldCall (const CallID& call_id) {
is_rec = false;
- _debug ("ManagerImpl::offHoldCall(%s)", call_id.c_str());
+ _debug ("Manager: Put call %s off hold", call_id.c_str());
stopTone();
@@ -580,11 +584,9 @@ bool ManagerImpl::offHoldCall (const CallID& call_id) {
// if this is not a conferenceand this and is not a conference participant
if (!isConference(current_call_id) && !participToConference(
current_call_id)) {
- _debug (" offHoldCall: put current call (%s) on hold", current_call_id.c_str());
onHoldCall(current_call_id);
} else if (isConference(current_call_id) && !participToConference(
call_id)) {
- _debug (" offHoldCall Put current conference (%s) on hold", current_call_id.c_str());
detachParticipant(default_id, current_call_id);
}
}
@@ -603,11 +605,11 @@ bool ManagerImpl::offHoldCall (const CallID& call_id) {
account_id = getAccountFromCall(call_id);
if (account_id == AccountNULL) {
- _debug ("Manager OffHold Call: Call doesn't exists");
+ _warn ("Manager: Error: Call doesn't exists in off hold");
return false;
}
- _debug ("Setting OFFHOLD, Account %s, callid %s", account_id.c_str(), call_id.c_str());
+ _debug ("Manager: Setting offhold, Account %s, callid %s", account_id.c_str(), call_id.c_str());
is_rec = getAccountLink(account_id)->getCall(call_id)->isRecording();
returnValue = getAccountLink(account_id)->offhold(call_id);
@@ -636,12 +638,6 @@ bool ManagerImpl::offHoldCall (const CallID& call_id) {
_audiodriver->flushMain();
}
- // codecName = getCurrentCodecName (call_id);
- // _debug("ManagerImpl::hangupCall(): broadcast codec name %s ",codecName.c_str());
-
- // if (_dbus) _dbus->getCallManager()->currentSelectedCodec (call_id,codecName.c_str());
-
-
return returnValue;
}
@@ -650,39 +646,18 @@ bool ManagerImpl::transferCall (const CallID& call_id, const std::string& to) {
AccountID accountid;
bool returnValue;
- _info("Manager: Transfer Call\n");
-
- stopTone();
+ _info("Manager: Transfer call %s\n", call_id.c_str());
CallID current_call_id = getCurrentCallId();
- if (participToConference(call_id)) {
-
- _info("Manager: Particip to a conference\n");
-
- Conference *conf = getConferenceFromCallID(call_id);
-
- if (conf != NULL) {
- // remove this participant
- removeParticipant(call_id);
-
- processRemainingParticipant(current_call_id, conf);
- }
- } else {
-
- // we are not participating to a conference, current call switched to ""
- if (!isConference(current_call_id))
- switchCall("");
- }
-
- /* Direct IP to IP call */
+ // Direct IP to IP call
if (getConfigFromCall(call_id) == Call::IPtoIP) {
- returnValue
- = SIPVoIPLink::instance(AccountNULL)-> transfer(call_id, to);
+ returnValue = SIPVoIPLink::instance(AccountNULL)-> transfer(call_id, to);
}
- /* Classic call, attached to an account */
+ // Classic call, attached to an account
else {
- accountid = getAccountFromCall(call_id);
+
+ accountid = getAccountFromCall(call_id);
if (accountid == AccountNULL) {
_warn ("Manager: Call doesn't exists");
@@ -691,9 +666,9 @@ bool ManagerImpl::transferCall (const CallID& call_id, const std::string& to) {
returnValue = getAccountLink(accountid)->transfer(call_id, to);
- removeCallAccount(call_id);
}
+ // remove waiting call in case we make transfer without even answer
removeWaitingCall(call_id);
return returnValue;
@@ -721,6 +696,8 @@ bool ManagerImpl::refuseCall (const CallID& id) {
AccountID accountid;
bool returnValue;
+ _debug("Manager: Refuse call %s", id.c_str());
+
CallID current_call_id = getCurrentCallId();
stopTone();
@@ -730,12 +707,14 @@ bool ManagerImpl::refuseCall (const CallID& id) {
// AudioLayer* audiolayer = getAudioDriver();
if (nbCalls <= 1) {
- _debug (" hangupCall: stop audio stream, ther is only %i call(s) remaining", nbCalls);
+ _debug ("Manager: Stop audio stream, ther is only %i call(s) remaining", nbCalls);
AudioLayer* audiolayer = getAudioDriver();
audiolayer->stopStream();
}
+ _debug("OK");
+
/* Direct IP to IP call */
if (getConfigFromCall(id) == Call::IPtoIP) {
@@ -747,7 +726,7 @@ bool ManagerImpl::refuseCall (const CallID& id) {
accountid = getAccountFromCall(id);
if (accountid == AccountNULL) {
- _debug ("! Manager OffHold Call: Call doesn't exists");
+ _warn ("Manager: Call doesn't exists");
return false;
}
@@ -755,6 +734,7 @@ bool ManagerImpl::refuseCall (const CallID& id) {
removeCallAccount(id);
}
+ _debug("OK");
// if the call was outgoing or established, we didn't refuse it
// so the method did nothing
@@ -763,9 +743,6 @@ bool ManagerImpl::refuseCall (const CallID& id) {
if (_dbus)
_dbus->getCallManager()->callStateChanged(id, "HUNGUP");
-
- // if(current_call_id.compare("") != 0)
- // switchCall ("");
}
return returnValue;
@@ -1545,18 +1522,24 @@ bool ManagerImpl::incomingCallWaiting () {
}
void ManagerImpl::addWaitingCall (const CallID& id) {
+
ost::MutexLock m(_waitingCallMutex);
_waitingCall.insert(id);
_nbIncomingWaitingCall++;
+
+ _info("Manager: Add waiting call %s (%d calls)", id.c_str(), _nbIncomingWaitingCall);
}
void ManagerImpl::removeWaitingCall (const CallID& id) {
+
ost::MutexLock m(_waitingCallMutex);
// should return more than 1 if it erase a call
if (_waitingCall.erase(id)) {
_nbIncomingWaitingCall--;
}
+
+ _info("Manager: Remove waiting call %s (%d calls)", id.c_str(), _nbIncomingWaitingCall);
}
bool ManagerImpl::isWaitingCall (const CallID& id) {
@@ -1574,29 +1557,29 @@ bool ManagerImpl::isWaitingCall (const CallID& id) {
////////////////////////////////////////////////////////////////////////////////
// SipEvent Thread
bool ManagerImpl::incomingCall (Call* call, const AccountID& accountId) {
- PulseLayer *pulselayer;
+
std::string from, number, display_name, display;
+ if(!call)
+ _error("Manager: Error: no call at this point");
+
stopTone();
- _debug ("Incoming call %s for account %s", call->getCallId().data(), accountId.c_str());
+ _debug ("Manager: Incoming call %s for account %s", call->getCallId().data(), accountId.c_str());
associateCallToAccount(call->getCallId(), accountId);
// If account is null it is an ip to ip call
-
if (accountId == AccountNULL) {
-
associateConfigToCall(call->getCallId(), Call::IPtoIP);
- } else {
+ }
+ else {
// strip sip: which is not required and bring confusion with ip to ip calls
// when placing new call from history (if call is IAX, do nothing)
std::string peerNumber = call->getPeerNumber();
int startIndex = peerNumber.find("sip:");
- // if "sip:" is found => it is not an IAX call
-
if (startIndex != (int) string::npos) {
std::string strippedPeerNumber = peerNumber.substr(startIndex + 4);
call->setPeerNumber(strippedPeerNumber);
@@ -1604,32 +1587,23 @@ bool ManagerImpl::incomingCall (Call* call, const AccountID& accountId) {
}
- _debug ("ManagerImpl::incomingCall :: hasCurrentCall() %i ", hasCurrentCall());
-
if (!hasCurrentCall()) {
+ _debug ("Manager: Has no current call");
call->setConnectionState(Call::Ringing);
ringtone();
- // switchCall (call->getCallId());
}
-
- /*
- else {
- addWaitingCall(call->getCallId());
- }
- */
+ else {
+ _debug ("Manager: has current call");
+ }
addWaitingCall(call->getCallId());
from = call->getPeerName();
-
number = call->getPeerNumber();
-
display_name = call->getDisplayName();
- // _debug( "incomingCall from: %s, number: %s, display_name: %s", from.c_str(), number.c_str(), display_name.c_str());
-
if (from != "" && number != "") {
from.append(" <");
from.append(number);
@@ -1640,33 +1614,15 @@ bool ManagerImpl::incomingCall (Call* call, const AccountID& accountId) {
from.append(">");
}
- /*
- CallIDSet::iterator iter = _waitingCall.begin();
- while (iter != _waitingCall.end()) {
- CallID ident = *iter;
- _debug("ManagerImpl::incomingCall :: CALL iteration: %s ",ident.c_str());
- ++iter;
- }
- */
-
/* Broadcast a signal over DBus */
- _debug ("From: %s, Number: %s, DisplayName: %s", from.c_str(), number.c_str(), display_name.c_str());
+ _debug ("Manager: From: %s, Number: %s, Display Name: %s", from.c_str(), number.c_str(), display_name.c_str());
display = display_name;
-
display.append(" ");
-
display.append(from);
if (_dbus)
- _dbus->getCallManager()->incomingCall(accountId, call->getCallId(),
- display.c_str());
-
- //if (_dbus) _dbus->getCallManager()->callStateChanged(call->getCallId(), "INCOMING");
-
- if (_audiodriver->getLayerType() == PULSEAUDIO) {
- pulselayer = dynamic_cast<PulseLayer *> (getAudioDriver());
- }
+ _dbus->getCallManager()->incomingCall(accountId, call->getCallId(), display.c_str());
return true;
}
@@ -1943,7 +1899,8 @@ void ManagerImpl::ringback () {
* Multi Thread
*/
void ManagerImpl::ringtone () {
- _debug ("ManagerImpl::ringtone");
+
+ _debug ("Manager: Start ringtone");
std::string ringchoice;
AudioLayer *audiolayer;
AudioCodec *codecForTone;
@@ -1952,7 +1909,7 @@ void ManagerImpl::ringtone () {
if (isRingtoneEnabled()) {
- _debug (" Tone is enabled");
+ _debug ("Manager: Tone is enabled");
//TODO Comment this because it makes the daemon crashes since the main thread
//synchronizes the ringtone thread.
@@ -1967,10 +1924,12 @@ void ManagerImpl::ringtone () {
audiolayer = getAudioDriver();
- layer = audiolayer->getLayerType();
-
- if (audiolayer == 0)
+ if (!audiolayer) {
+ _error("Manager: Error: no audio layer in ringtone");
return;
+ }
+
+ layer = audiolayer->getLayerType();
samplerate = audiolayer->getSampleRate();
@@ -3937,7 +3896,6 @@ bool ManagerImpl::removeCallAccount (const CallID& callID) {
if (_callAccountMap.erase(callID)) {
return true;
}
-
return false;
}
@@ -4096,18 +4054,19 @@ ManagerImpl::getAccount (const AccountID& accountID) {
AccountID ManagerImpl::getAccountIdFromNameAndServer (
const std::string& userName, const std::string& server) {
+
AccountMap::iterator iter;
SIPAccount *account;
- _debug ("getAccountIdFromNameAndServer : username = %s , server = %s", userName.c_str(), server.c_str());
+
+ _info ("Manager : username = %s , server = %s", userName.c_str(), server.c_str());
// Try to find the account id from username and server name by full match
for (iter = _accountMap.begin(); iter != _accountMap.end(); ++iter) {
- _debug ("for : account = %s", iter->first.c_str());
account = dynamic_cast<SIPAccount *> (iter->second);
if (account != NULL) {
if (account->fullMatch(userName, server)) {
- _debug ("Matching accountId in request is a fullmatch");
+ _debug ("Manager: Matching account id in request is a fullmatch %s@%s", userName.c_str(), server.c_str());
return iter->first;
}
}
@@ -4119,7 +4078,7 @@ AccountID ManagerImpl::getAccountIdFromNameAndServer (
if (account != NULL) {
if (account->hostnameMatch(server)) {
- _debug ("Matching accountId in request with hostname");
+ _debug ("Manager: Matching account id in request with hostname %s", server.c_str());
return iter->first;
}
}
@@ -4131,12 +4090,14 @@ AccountID ManagerImpl::getAccountIdFromNameAndServer (
if (account != NULL) {
if (account->userMatch(userName)) {
- _debug ("Matching accountId in request with username");
+ _debug ("Manager: Matching account id in request with username %s", userName.c_str());
return iter->first;
}
}
}
+ _debug ("Manager: Username %s or server %s doesn't match any account, using IP2IP", userName.c_str(), server.c_str());
+
// Failed again! return AccountNULL
return AccountNULL;
}
@@ -4260,7 +4221,7 @@ bool ManagerImpl::associateConfigToCall (const CallID& callID,
if (getConfigFromCall(callID) == CallConfigNULL) { // nothing with the same ID
_callConfigMap[callID] = config;
- _debug ("Associate Call %s with config %i", callID.data(), config);
+ _debug ("Manager: Associate call %s with config %i", callID.c_str(), config);
return true;
} else {
return false;
diff --git a/sflphone-common/src/sip/sipcall.cpp b/sflphone-common/src/sip/sipcall.cpp
index d1e22660c64e97650684a27048114b875de382e2..9ed7f7a90e42b3651fdd67fddde9972fbd9a7565 100644
--- a/sflphone-common/src/sip/sipcall.cpp
+++ b/sflphone-common/src/sip/sipcall.cpp
@@ -33,17 +33,19 @@ SIPCall::SIPCall (const CallID& id, Call::CallType type, pj_pool_t *pool) : Call
, _invSession (NULL)
, _local_sdp (0)
{
+ _debug ("SIPCall: Create new call %s", id.c_str());
+
_local_sdp = new Sdp (pool);
- _debug ("SIPCALL::Constructor for this class is called ");
}
SIPCall::~SIPCall()
{
+ _debug ("SIPCall: Delete call");
+
delete _audiortp;
_audiortp = 0;
delete _local_sdp;
_local_sdp = 0;
- _debug ("SIPCALL::Destructor for this class is called ");
}
diff --git a/sflphone-common/src/sip/sipvoiplink.cpp b/sflphone-common/src/sip/sipvoiplink.cpp
index 2309c86923288825ba59aed8ce1b1ab8a04c50ff..36901c1e779c2b67730b6dcbd4f4cd965a8f3658 100644
--- a/sflphone-common/src/sip/sipvoiplink.cpp
+++ b/sflphone-common/src/sip/sipvoiplink.cpp
@@ -352,6 +352,7 @@ SIPVoIPLink::terminateSIPCall()
void
SIPVoIPLink::terminateOneCall (const CallID& id)
{
+ _debug("UserAgent: Terminate call %s", id.c_str());
SIPCall *call = getSIPCall (id);
@@ -396,7 +397,7 @@ std::string SIPVoIPLink::getInterfaceAddrFromName(std::string ifaceName) {
struct in_addr *addr_in;
if((fd = socket (AF_INET, SOCK_DGRAM,0)) < 0)
- _debug("UserAgent: getInterfaceAddrFromName error could not open socket\n");
+ _error("UserAgent: Error: could not open socket");
memset (&ifr, 0, sizeof (struct ifreq));
@@ -404,7 +405,7 @@ std::string SIPVoIPLink::getInterfaceAddrFromName(std::string ifaceName) {
ifr.ifr_addr.sa_family = AF_INET;
if((err = ioctl(fd, SIOCGIFADDR, &ifr)) < 0)
- _debug("UserAgent: getInterfaceAddrFromName use default interface (0.0.0.0)\n");
+ _debug("UserAgent: Use default interface (0.0.0.0)");
saddr_in = (struct sockaddr_in *)&ifr.ifr_addr;
addr_in = &(saddr_in->sin_addr);
@@ -974,15 +975,15 @@ SIPVoIPLink::peerHungup (const CallID& id)
bool
SIPVoIPLink::cancel (const CallID& id)
{
+ _info ("UserAgent: Cancel call %s", id.c_str());
+
SIPCall* call = getSIPCall (id);
- if (call==0) {
- _debug ("! SIP Error: Call doesn't exist");
+ if (!call) {
+ _warn("UserAgent: Error: Call doesn't exist");
return false;
}
- _debug ("- SIP Action: Cancel call %s [cid: %3d]", id.data(), call->getCid());
-
terminateOneCall (id);
removeCall (id);
@@ -1203,17 +1204,18 @@ SIPVoIPLink::refuse (const CallID& id)
pj_status_t status;
pjsip_tx_data *tdata;
+ _debug("UserAgent: Refuse call %s", id.c_str());
call = getSIPCall (id);
if (call==0) {
- _debug ("Call doesn't exist");
+ _error ("UserAgent: Error: Call doesn't exist");
return false;
}
// can't refuse outgoing call or connected
if (!call->isIncoming() || call->getConnectionState() == Call::Connected) {
- _debug ("It's not an incoming call, or it's already answered");
+ _debug ("UserAgent: Call %s is not in state incoming, or is already answered");
return false;
}
@@ -1232,6 +1234,8 @@ SIPVoIPLink::refuse (const CallID& id)
terminateOneCall (id);
+ _debug("UserAgent: Refuse call completed");
+
return true;
}
@@ -1395,25 +1399,18 @@ SIPVoIPLink::SIPStartCall (SIPCall* call, const std::string& subject UNUSED)
// Creates URI
std::string fromUri;
-
std::string toUri;
-
std::string contactUri;
fromUri = account->getFromUri();
-
toUri = call->getPeerNumber(); // expecting a fully well formed sip uri
std::string address = findLocalAddressFromUri (toUri, account->getAccountTransport ());
-
int port = findLocalPortFromUri (toUri, account->getAccountTransport ());
std::stringstream ss;
-
std::string portStr;
-
ss << port;
-
ss >> portStr;
contactUri = account->getContactHeader (address, portStr);
@@ -1424,15 +1421,12 @@ SIPVoIPLink::SIPStartCall (SIPCall* call, const std::string& subject UNUSED)
contactUri.c_str());
pj_str_t pjFrom;
-
pj_cstr (&pjFrom, fromUri.c_str());
pj_str_t pjContact;
-
pj_cstr (&pjContact, contactUri.c_str());
pj_str_t pjTo;
-
pj_cstr (&pjTo, toUri.c_str());
// Create the dialog (UAC)
@@ -1476,7 +1470,6 @@ SIPVoIPLink::SIPStartCall (SIPCall* call, const std::string& subject UNUSED)
// decrement transport's ref count
// pjsip_transport_dec_ref(account->getAccountTransport());
-
status = pjsip_inv_send_msg (inv, tdata);
if (status != PJ_SUCCESS) {
@@ -1499,7 +1492,7 @@ void
SIPVoIPLink::SIPCallServerFailure (SIPCall *call)
{
if (call != 0) {
- _debug ("Server error!");
+ _error ("UserAgent: Error: Server error!");
CallID id = call->getCallId();
Manager::instance().callFailure (id);
terminateOneCall (id);
@@ -2987,7 +2980,7 @@ void set_voicemail_info (AccountID account, pjsip_msg_body *body)
void SIPVoIPLink::handle_reinvite (SIPCall *call)
{
- _debug ("UserAgent: handle_reinvite");
+ _debug ("UserAgent: Handle reinvite");
/*
// Close the previous RTP session
call->getAudioRtp()->stop ();
@@ -3010,7 +3003,7 @@ void SIPVoIPLink::handle_reinvite (SIPCall *call)
// This callback is called when the invite session state has changed
void call_on_state_changed (pjsip_inv_session *inv, pjsip_event *e)
{
- _debug ("call_on_state_changed to state %s", invitationStateMap[inv->state]);
+ _debug ("UserAgent: Call state changed to %s", invitationStateMap[inv->state]);
pjsip_rx_data *rdata;
pj_status_t status;
@@ -3055,7 +3048,6 @@ void call_on_state_changed (pjsip_inv_session *inv, pjsip_event *e)
pjsip_evsub_state ev_state = PJSIP_EVSUB_STATE_ACTIVE;
switch (call->getInvSession()->state) {
- // switch (inv->state) {
case PJSIP_INV_STATE_NULL:
@@ -3094,9 +3086,7 @@ void call_on_state_changed (pjsip_inv_session *inv, pjsip_event *e)
pjsip_tx_data *tdata;
pj_status_t status;
- status = pjsip_xfer_notify (call->getXferSub(),
- ev_state, st_code,
- NULL, &tdata);
+ status = pjsip_xfer_notify (call->getXferSub(), ev_state, st_code, NULL, &tdata);
if (status != PJ_SUCCESS) {
_debug ("UserAgent: Unable to create NOTIFY -- %d", status);
@@ -3155,45 +3145,35 @@ void call_on_state_changed (pjsip_inv_session *inv, pjsip_event *e)
_debug ("State: %s. Cause: %.*s", invitationStateMap[inv->state], (int) inv->cause_text.slen, inv->cause_text.ptr);
+ accId = Manager::instance().getAccountFromCall (call->getCallId());
+ link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink (accId));
+
+ // Make sure link is valid
+ assert(link);
+
switch (inv->cause) {
- /* The call terminates normally - BYE / CANCEL */
+ // The call terminates normally - BYE / CANCEL
case PJSIP_SC_OK:
-
case PJSIP_SC_REQUEST_TERMINATED:
- accId = Manager::instance().getAccountFromCall (call->getCallId());
- link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink (accId));
-
- if (link) {
- link->SIPCallClosed (call);
- }
-
+ link->SIPCallClosed (call);
break;
- case PJSIP_SC_NOT_FOUND: /* peer not found */
-
- case PJSIP_SC_DECLINE: /* We have been ignored */
+ case PJSIP_SC_DECLINE:
+ _debug("UserAgent: Call %s is declined", call->getCallId().c_str());
+ if (inv->role == PJSIP_ROLE_UAC)
+ link->SIPCallServerFailure (call);
+ break;
+ case PJSIP_SC_NOT_FOUND: /* peer not found */
case PJSIP_SC_REQUEST_TIMEOUT: /* request timeout */
-
case PJSIP_SC_NOT_ACCEPTABLE_HERE: /* no compatible codecs */
-
case PJSIP_SC_NOT_ACCEPTABLE_ANYWHERE:
-
case PJSIP_SC_UNSUPPORTED_MEDIA_TYPE:
-
case PJSIP_SC_UNAUTHORIZED:
-
case PJSIP_SC_FORBIDDEN:
-
case PJSIP_SC_REQUEST_PENDING:
- accId = Manager::instance().getAccountFromCall (call->getCallId());
- link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink (accId));
-
- if (link) {
- link->SIPCallServerFailure (call);
- }
-
+ link->SIPCallServerFailure (call);
break;
default:
@@ -3207,7 +3187,7 @@ void call_on_state_changed (pjsip_inv_session *inv, pjsip_event *e)
// This callback is called after SDP offer/answer session has completed.
void call_on_media_update (pjsip_inv_session *inv, pj_status_t status)
{
- _debug ("UserAgent: Call on media update");
+ _debug ("UserAgent: Call media update");
const pjmedia_sdp_session *local_sdp;
const pjmedia_sdp_session *remote_sdp;
@@ -3337,8 +3317,9 @@ void call_on_forked (pjsip_inv_session *inv, pjsip_event *e)
void call_on_tsx_changed (pjsip_inv_session *inv, pjsip_transaction *tsx, pjsip_event *e)
{
- _debug("UserAgent: Transaction changed to state %s", transactionStateMap[tsx->state]);
+ assert(tsx);
+ _debug("UserAgent: Transaction changed to state %s", transactionStateMap[tsx->state]);
if (tsx->role==PJSIP_ROLE_UAS && tsx->state==PJSIP_TSX_STATE_TRYING &&
pjsip_method_cmp (&tsx->method, &pjsip_refer_method) ==0) {
@@ -3348,10 +3329,14 @@ void call_on_tsx_changed (pjsip_inv_session *inv, pjsip_transaction *tsx, pjsip_
if (e && e->body.rx_msg.rdata) {
+
+ _debug("Event");
pjsip_tx_data* t_data;
pjsip_rx_data* r_data = e->body.rx_msg.rdata;
- if (r_data->msg_info.msg->line.req.method.id == PJSIP_OTHER_METHOD) {
+ if (r_data && r_data->msg_info.msg->line.req.method.id == PJSIP_OTHER_METHOD) {
+
+ _debug("R_data");
std::string method_info = "INFO";
std::string method_notify = "NOTIFY";
@@ -3373,6 +3358,8 @@ void call_on_tsx_changed (pjsip_inv_session *inv, pjsip_transaction *tsx, pjsip_
}
}
}
+
+ _debug("OK: ransaction changed to state");
}
void regc_cb (struct pjsip_regc_cbparam *param)
@@ -3502,18 +3489,18 @@ mod_on_rx_request (pjsip_rx_data *rdata)
std::string method_name;
std::string request;
+
+ _info("UserAgent: Receiving REQUEST using transport: %s %s (refcnt=%d)",
+ rdata->tp_info.transport->obj_name,
+ rdata->tp_info.transport->info,
+ (int)pj_atomic_get(rdata->tp_info.transport->ref_cnt));
+
// No need to go any further on incoming ACK
if (rdata->msg_info.msg->line.req.method.id == PJSIP_ACK_METHOD) {
_info("UserAgent: received an ACK");
return true;
}
- // Handle the incoming call invite in this function
- _info("UserAgent: Receiving REQUEST using transport: %s %s (refcnt=%d)",
- rdata->tp_info.transport->obj_name,
- rdata->tp_info.transport->info,
- (int)pj_atomic_get(rdata->tp_info.transport->ref_cnt));
-
/* First, let's got the username and server name from the invite.
* We will use them to detect which account is the callee.
*/
@@ -3523,13 +3510,10 @@ mod_on_rx_request (pjsip_rx_data *rdata)
userName = std::string (sip_uri->user.ptr, sip_uri->user.slen);
server = std::string (sip_uri->host.ptr, sip_uri->host.slen);
- _debug ("UserAgent: mod_on_rx_request: %s@%s", userName.c_str(), server.c_str());
-
// Get the account id of callee from username and server
account_id = Manager::instance().getAccountIdFromNameAndServer (userName, server);
/* If we don't find any account to receive the call */
-
if (account_id == AccountNULL) {
_debug ("UserAgent: Username %s doesn't match any account, using IP2IP!",userName.c_str());
}
@@ -3562,7 +3546,7 @@ mod_on_rx_request (pjsip_rx_data *rdata)
displayName = std::string ("");
}
- _debug ("UserAgent: The receiver is : %s@%s", userName.data(), server.data());
+ _debug ("UserAgent: The receiver is: %s@%s", userName.data(), server.data());
_debug ("UserAgent: The callee account id is %s", account_id.c_str());
/* Now, it is the time to find the information of the caller */
@@ -3580,6 +3564,7 @@ mod_on_rx_request (pjsip_rx_data *rdata)
// Get the server voicemail notification
// Catch the NOTIFY message
if (rdata->msg_info.msg->line.req.method.id == PJSIP_OTHER_METHOD) {
+
method_name = "NOTIFY";
// Retrieve all the message. Should contains only the method name but ...
request = rdata->msg_info.msg->line.req.method.name.ptr;
@@ -3616,10 +3601,10 @@ mod_on_rx_request (pjsip_rx_data *rdata)
get_remote_sdp_from_offer (rdata, &r_sdp);
if(account->getActiveCodecs().empty()) {
- _warn ("UserAgent: Error: No active codec");
- pj_strdup2 (_pool, &reason, "no active codec");
+ _warn ("UserAgent: Error: No active codec");
+ pj_strdup2 (_pool, &reason, "no active codec");
pjsip_endpt_respond_stateless (_endpt, rdata, PJSIP_SC_NOT_ACCEPTABLE_HERE ,
- &reason, NULL, NULL);
+ &reason, NULL, NULL);
return true;
}
@@ -3658,13 +3643,14 @@ mod_on_rx_request (pjsip_rx_data *rdata)
id = Manager::instance().getNewCallID();
call = new SIPCall (id, Call::Incoming, _pool);
- /* If an error occured at the call creation */
+
+ // If an error occured at the call creation
if (!call) {
_warn("UserAgent: Error: Unable to create an incoming call");
- pj_strdup2 (_pool, &reason, "unable to create an incoming call");
+ pj_strdup2 (_pool, &reason, "unable to create an incoming call");
pjsip_endpt_respond_stateless (_endpt, rdata, PJSIP_SC_INTERNAL_SERVER_ERROR,
&reason, NULL, NULL);
- return false;
+ return false;
}
std::string addrToUse, addrSdp ="0.0.0.0";
@@ -3676,17 +3662,17 @@ mod_on_rx_request (pjsip_rx_data *rdata)
// May use the published address as well
addrToUse = SIPVoIPLink::instance("")->getInterfaceAddrFromName(account->getLocalInterface ());
- account->isStunEnabled () ? addrSdp = account->getPublishedAddress () : addrSdp = addrToUse;
- // Set the appropriate transport to have the right VIA header
- link->init_transport_selector (account->getAccountTransport (), &tp);
+ account->isStunEnabled () ? addrSdp = account->getPublishedAddress () : addrSdp = addrToUse;
+ // Set the appropriate transport to have the right VIA header
+ link->init_transport_selector (account->getAccountTransport (), &tp);
- if(account->getAccountTransport()) {
+ if(account->getAccountTransport()) {
- _debug("UserAgent: Process INVITE request using transport: %s %s (refcnt=%i)",
- account->getAccountTransport()->obj_name,
- account->getAccountTransport()->info,
- (int)pj_atomic_get(account->getAccountTransport()->ref_cnt));
- }
+ _debug("UserAgent: SIP transport for this account: %s %s (refcnt=%i)",
+ account->getAccountTransport()->obj_name,
+ account->getAccountTransport()->info,
+ (int)pj_atomic_get(account->getAccountTransport()->ref_cnt));
+ }
}
@@ -3707,22 +3693,6 @@ mod_on_rx_request (pjsip_rx_data *rdata)
call->initRecFileName();
- // Notify UI there is an incoming call
-
- _debug ("UserAgent: Add call to account link");
-
- if (Manager::instance().incomingCall (call, account_id)) {
- // Add this call to the callAccountMap in ManagerImpl
- Manager::instance().getAccountLink (account_id)->addCall (call);
- } else {
- // Fail to notify UI
- delete call; call = NULL;
- _warn ("UserAgent: Fail to notify UI!");
- pj_strdup2 (_pool, &reason, "fail to notify ui");
- pjsip_endpt_respond_stateless (_endpt, rdata, PJSIP_SC_INTERNAL_SERVER_ERROR,
- &reason, NULL, NULL);
- return false;
- }
// Have to do some stuff with the SDP
// Set the codec map, IP, peer number and so on... for the SIPCall object
@@ -3758,7 +3728,7 @@ mod_on_rx_request (pjsip_rx_data *rdata)
pj_strdup2 (_pool, &reason, "fail to create uas dialog");
pjsip_endpt_respond_stateless (_endpt, rdata, PJSIP_SC_INTERNAL_SERVER_ERROR,
&reason, NULL, NULL);
- return true;
+ return false;
}
// Specify media capability during invite session creation
@@ -3772,7 +3742,8 @@ mod_on_rx_request (pjsip_rx_data *rdata)
// Associate the call in the invite session
inv->mod_data[_mod_ua.id] = call;
- // Send a 180/Ringing response
+ // Send a 180 Ringing response
+ _info ("UserAgent: Send a 180 Ringing response");
status = pjsip_inv_initial_answer (inv, rdata, PJSIP_SC_RINGING, NULL, NULL, &tdata);
PJ_ASSERT_RETURN (status == PJ_SUCCESS, 1);
@@ -3787,6 +3758,20 @@ mod_on_rx_request (pjsip_rx_data *rdata)
// Update the connection state
call->setConnectionState (Call::Ringing);
+ _debug ("UserAgent: Add call to account link");
+ if (Manager::instance().incomingCall (call, account_id)) {
+ // Add this call to the callAccountMap in ManagerImpl
+ Manager::instance().getAccountLink (account_id)->addCall (call);
+ } else {
+ // Fail to notify UI
+ delete call; call = NULL;
+ _warn ("UserAgent: Fail to notify UI!");
+ pj_strdup2 (_pool, &reason, "fail to notify ui");
+ pjsip_endpt_respond_stateless (_endpt, rdata, PJSIP_SC_INTERNAL_SERVER_ERROR,
+ &reason, NULL, NULL);
+ return false;
+ }
+
/* Done */
return true;
diff --git a/sippxml/ip2ip_uac_send_hangup.xml b/sippxml/ip2ip_uac_send_hangup.xml
index 49cc63fca06e2cd1f2a40f1f65929de436da7d6b..8b801eb49438bb553a448f28e2c410f86c34a234 100644
--- a/sippxml/ip2ip_uac_send_hangup.xml
+++ b/sippxml/ip2ip_uac_send_hangup.xml
@@ -95,6 +95,4 @@
<recv response="200">
</recv>
- <pause milliseconds="1000"/>
-
</scenario>
diff --git a/sippxml/ip2ip_uac_send_peer_hungup.xml b/sippxml/ip2ip_uac_send_peer_hungup.xml
index 4e811e38262f10680417f9415e960e33585b26ee..048b0d8a062521288ed6afc5c21934a2232dd30a 100644
--- a/sippxml/ip2ip_uac_send_peer_hungup.xml
+++ b/sippxml/ip2ip_uac_send_peer_hungup.xml
@@ -89,6 +89,4 @@
]]>
</send>
- <pause milliseconds="1000"/>
-
</scenario>
diff --git a/sippxml/testsuiteuac.sh b/sippxml/testsuiteuac.sh
index 1ed0db416504d9abc8288a3a56e8b848d0e46263..9e993dfeb80926a0b7131add92bc4d657ab89726 100644
--- a/sippxml/testsuiteuac.sh
+++ b/sippxml/testsuiteuac.sh
@@ -62,29 +62,7 @@ function test_ip2ip_recv_hangup {
# sleep 1;
# start sipp client and send calls
- sipp -sf ip2ip_uac_send_peer_hungup.xml ${REMOTEADDR_lo} -i ${LOCALIP_lo} -p ${LOCALPORT} -l 1 -m 10
-
- # kill every one
- # bashtrap
-}
-
-
-# SCENARIO 1 Test 4
-function test_ip2ip_recv_peer_hungup {
-
- # start sflphoned
- # /usr/lib/sflphone/sflphoned&
-
- # wait some time to make sure sflphoned is started
- # sleep 1;
-
- # python ../tools/pysflphone/pysflphone_testdbus.py &
-
- # wait some time to make sure client is bound
- # sleep 1;
-
- # start sipp client and send calls
- sipp -sf ip2ip_uac_send_hangup.xml ${REMOTEADDR_lo} -i ${LOCALIP_lo} -p ${LOCALPORT} -l 1 -m 10
+ sipp -sf ip2ip_uac_send_peer_hungup.xmlip2ip_uac_send_hangup.xml ${REMOTEADDR_lo} -i ${LOCALIP_lo} -p ${LOCALPORT} -l 1 -m 10
# kill every one
# bashtrap
@@ -230,6 +208,26 @@ function test_account_send_transfer {
}
+# SCENARIO 5 Test 1
+function test_ip2ip_send_refused {
+
+ # start sflphoned
+ # /usr/lib/sflphone/sflphoned&
+
+ # start sipp server to receive calls from sflphone and then hangup
+ sipp -sf ip2ip_uac_send_refused.xml ${REMOTEADDR_lo} -s ${REMOTEADDR_lo} -i ${LOCALIP_lo} -p ${LOCALPORT} -l 1
+
+ # wait some time to make sure sflphoned is started
+ # sleep 1;
+
+ # run python client and script to make calls
+ # python ../tools/pysflphone/pysflphone_testdbus.py &
+
+ # kill every one
+ bashtrap
+}
+
+
# function called if CTRL-C detected
bashtrap()
{
@@ -288,7 +286,7 @@ bashtrap()
# - Put this call on HOLD
# - Off HOLD this call
# - Hangup
-test_ip2ip_send_hold_offhold
+# test_ip2ip_send_hold_offhold
@@ -297,4 +295,11 @@ test_ip2ip_send_hold_offhold
# Test 1: - Send an IP2IP call
# - Transfer this call to another sipp instance
# - Hangup
-# test_account_send_transfer
\ No newline at end of file
+# test_account_send_transfer
+
+
+#SCENARIO 5: Refuse call (IP2IP)
+
+# Test 1: - Receive a call
+# - Refuse (hangup without answer)
+test_ip2ip_send_refused
\ No newline at end of file
diff --git a/tools/pysflphone/pysflphone_testdbus.py b/tools/pysflphone/pysflphone_testdbus.py
index a1c909c531fb3a2b73306c59466039e26bbcefb1..933f8736c9ab8e57183006a92069910119ff66d2 100644
--- a/tools/pysflphone/pysflphone_testdbus.py
+++ b/tools/pysflphone/pysflphone_testdbus.py
@@ -21,21 +21,29 @@ PHONE1="27182"
PHONE2="31416"
PHONE3="14142"
+
# Define function callback to emulate UA behavior on
-# recieving a call
+# recieving a call (peer hangup))
def acceptOnIncomingCall(sflphone):
- time.sleep(0.2)
+
sflphone.Accept(sflphone.currentCallId)
+
# Define function callback to emulate UA behavior on
# receiving a call and hanging up
def acceptOnIncomingCallHangup(sflphone):
- time.sleep(0.2)
- sflphone.Accept(sflphone.currentCallId)
- time.sleep(0.5)
+
+ sflphone.Accept(sflphone.currentCallId)
sflphone.HangUp(sflphone.currentCallId)
+# Define function callback to emulate UA behavior on
+# refusing a call
+def refuseOnIncomingCall(sflphone):
+ # time.sleep(0.5)
+ sflphone.Refuse(sflphone.currentCallId)
+
+
class SflPhoneTests():
def __init__(self, sfl):
@@ -230,6 +238,17 @@ class SflPhoneTests():
i = i+1
+ # SCENARIO 5 Test 1
+ def test_ip2ip_recv_refuse(self):
+ """Receive an incoming IP2IP call, refuse it"""
+
+ # Add callback for this test
+ self.sflphone.onIncomingCall_cb = refuseOnIncomingCall
+
+ # Start Glib mainloop
+ self.sflphone.start()
+
+
# Open sflphone and connect to sflphoned through dbus
sflphone = SflPhoneCtrlSimple(True)
@@ -291,7 +310,7 @@ sflphone.setFirstRegisteredAccount();
# - Put this call on HOLD
# - Off HOLD this call
# - Hangup
-testsuite.test_ip2ip_send_hold_offhold()
+# testsuite.test_ip2ip_send_hold_offhold()
@@ -301,3 +320,11 @@ testsuite.test_ip2ip_send_hold_offhold()
# - Transfer this call to another sipp instance
# - Hangup
# testsuite.test_account_send_transfer()
+
+
+
+# SCENARIO 5: IP2IP Call, Refuse
+
+# Test 1: - Receive an incoming call
+# - Hangup without answer
+testsuite.test_ip2ip_recv_refuse()
diff --git a/tools/pysflphone/sflphonectrlsimple.py b/tools/pysflphone/sflphonectrlsimple.py
index a19d4ab0901f4b28fb2081d3f68fc58a5419b2e1..491e105e8256fb6314c80b95b081fb0d5c7f75fc 100755
--- a/tools/pysflphone/sflphonectrlsimple.py
+++ b/tools/pysflphone/sflphonectrlsimple.py
@@ -66,6 +66,7 @@ class SflPhoneCtrlSimple(Thread):
self.registered = False
self.register()
self.currentCallId = ""
+
self.loop = MainLoop()
self.test = test
@@ -534,11 +535,12 @@ class SflPhoneCtrlSimple(Thread):
def Refuse(self, callid):
"""Refuse an incoming call identified by a CallID"""
- if not self.account:
- self.setFirstRegisteredAccount()
+ print "Refuse call " + callid
+ # if not self.account:
+ # self.setFirstRegisteredAccount()
- if not self.isAccountRegistered():
- raise SflPhoneError("Can't refuse a call without a registered account")
+ # if not self.isAccountRegistered():
+ # raise SflPhoneError("Can't refuse a call without a registered account")
if callid is None or callid == "":
raise SflPhoneError("Invalid callID")
@@ -548,6 +550,7 @@ class SflPhoneCtrlSimple(Thread):
def Accept(self, callid):
"""Accept an incoming call identified by a CallID"""
+ print "Accept call " + callid
if not self.account:
self.setFirstRegisteredAccount()
@@ -603,4 +606,9 @@ class SflPhoneCtrlSimple(Thread):
return callid
def run(self):
- self.loop.run()
+ gobject.threads_init()
+ # self.loop.run()
+ context = self.loop.get_context()
+
+ while 1:
+ context.iteration(True)