jami/jami-daemon
1
0
Fork 0
mirror of https://git.jami.net/savoirfairelinux/jami-daemon.git synced 2025-08-07 22:02:12 +08:00
Code Issues Projects Releases Wiki Activity

[#2906] Update RTP remote ip address during this call_onb_media_update

Browse Source 
Was previously updated during handle_reinvite
This commit is contained in:
Alexandre Savard
2010-02-18 14:03:15 -05:00
parent 06f43f791e
commit 414859e6c8
5 changed files with 50 additions and 73 deletions
Download Patch File Download Diff File Expand all files Collapse all files

45
sflphone-common/src/audio/audiortp/AudioRtpSession.h
View File

@ -208,7 +208,7 @@ namespace sfl {
assert(_ca); assert(_ca);
_debug ("Local audio port %i will be used\n", _ca->getLocalAudioPort()); _debug ("Local audio port %i will be used", _ca->getLocalAudioPort());
//mic, we receive from soundcard in stereo, and we send encoded //mic, we receive from soundcard in stereo, and we send encoded
_audiolayer = _manager->getAudioDriver(); _audiolayer = _manager->getAudioDriver();
@ -223,7 +223,7 @@ namespace sfl {
template <typename D> template <typename D>
AudioRtpSession<D>::~AudioRtpSession() AudioRtpSession<D>::~AudioRtpSession()
{ {
_debug ("Delete AudioRtpSession instance\n"); _debug ("Delete AudioRtpSession instance");
try { try {
terminate(); terminate();
@ -232,7 +232,7 @@ namespace sfl {
throw; throw;
} }
_debug("Unbind audio RTP stream for call id %s\n", _ca->getCallId().c_str()); _debug("Unbind audio RTP stream for call id %s", _ca->getCallId().c_str());
// _audiolayer->getMainBuffer()->unBindAll(_ca->getCallId()); // _audiolayer->getMainBuffer()->unBindAll(_ca->getCallId());
_manager->getAudioDriver()->getMainBuffer()->unBindAll(_ca->getCallId()); _manager->getAudioDriver()->getMainBuffer()->unBindAll(_ca->getCallId());
@ -243,7 +243,7 @@ namespace sfl {
delete [] _spkrDataConverted; delete [] _spkrDataConverted;
delete _time; delete _time;
delete _converter; delete _converter;
_debug ("AudioRtpSession instance deleted\n"); _debug ("AudioRtpSession instance deleted");
} }
template <typename D> template <typename D>
@ -292,24 +292,24 @@ namespace sfl {
_audiocodec = _manager->getCodecDescriptorMap().instantiateCodec(pl); _audiocodec = _manager->getCodecDescriptorMap().instantiateCodec(pl);
if (_audiocodec == NULL) { if (_audiocodec == NULL) {
_debug ("No audiocodec, can't init RTP media\n"); _debug ("No audiocodec, can't init RTP media");
throw AudioRtpSessionException(); throw AudioRtpSessionException();
} }
_debug ("Init audio RTP session: codec payload %i\n", _audiocodec->getPayload()); _debug ("Init audio RTP session: codec payload %i", _audiocodec->getPayload());
_codecSampleRate = _audiocodec->getClockRate(); _codecSampleRate = _audiocodec->getClockRate();
_codecFrameSize = _audiocodec->getFrameSize(); _codecFrameSize = _audiocodec->getFrameSize();
//TODO: figure out why this is necessary. //TODO: figure out why this is necessary.
if (_audiocodec->getPayload() == 9) { if (_audiocodec->getPayload() == 9) {
_debug ("Setting payload format to G722\n"); _debug ("Setting payload format to G722");
static_cast<D*>(this)->setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) _audiocodec->getPayload(), _audiocodec->getClockRate())); static_cast<D*>(this)->setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) _audiocodec->getPayload(), _audiocodec->getClockRate()));
} else if (_audiocodec->hasDynamicPayload()) { } else if (_audiocodec->hasDynamicPayload()) {
_debug ("Setting a dynamic payload format\n"); _debug ("Setting a dynamic payload format");
static_cast<D*>(this)->setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) _audiocodec->getPayload(), _audiocodec->getClockRate())); static_cast<D*>(this)->setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) _audiocodec->getPayload(), _audiocodec->getClockRate()));
} else if (!_audiocodec->hasDynamicPayload() && _audiocodec->getPayload() != 9) { } else if (!_audiocodec->hasDynamicPayload() && _audiocodec->getPayload() != 9) {
_debug ("Setting a static payload format\n"); _debug ("Setting a static payload format");
static_cast<D*>(this)->setPayloadFormat (ost::StaticPayloadFormat ( (ost::StaticPayloadType) _audiocodec->getPayload())); static_cast<D*>(this)->setPayloadFormat (ost::StaticPayloadFormat ( (ost::StaticPayloadType) _audiocodec->getPayload()));
} }
} }
@ -318,18 +318,18 @@ namespace sfl {
void AudioRtpSession<D>::setDestinationIpAddress(void) void AudioRtpSession<D>::setDestinationIpAddress(void)
{ {
if (_ca == NULL) { if (_ca == NULL) {
_debug ("Sipcall is gone.\n"); _debug ("Sipcall is gone.");
throw AudioRtpSessionException(); throw AudioRtpSessionException();
} }
_debug ("Setting IP address for the RTP session\n"); _debug ("Setting IP address for the RTP session");
// Store remote ip in case we would need to forget current destination // Store remote ip in case we would need to forget current destination
_remote_ip = ost::InetHostAddress(_ca->getLocalSDP()->get_remote_ip().c_str()); _remote_ip = ost::InetHostAddress(_ca->getLocalSDP()->get_remote_ip().c_str());
_debug ("Init audio RTP session: remote ip %s\n", _ca->getLocalSDP()->get_remote_ip().data()); _debug ("Init audio RTP session: remote ip %s", _ca->getLocalSDP()->get_remote_ip().data());
if (!_remote_ip) { if (!_remote_ip) {
_debug ("Target IP address [%s] is not correct!\n", _ca->getLocalSDP()->get_remote_ip().data()); _debug ("Target IP address [%s] is not correct!", _ca->getLocalSDP()->get_remote_ip().data());
return; return;
} }
@ -337,7 +337,7 @@ namespace sfl {
_remote_port = (unsigned short) _ca->getLocalSDP()->get_remote_audio_port(); _remote_port = (unsigned short) _ca->getLocalSDP()->get_remote_audio_port();
if (! static_cast<D*>(this)->addDestination (_remote_ip, _remote_port)) { if (! static_cast<D*>(this)->addDestination (_remote_ip, _remote_port)) {
_debug ("Can't add destination to session!\n"); _debug ("Can't add destination to session!");
return; return;
} }
} }
@ -347,6 +347,7 @@ namespace sfl {
{ {
// Destination address are stored in a list in ccrtp // Destination address are stored in a list in ccrtp
// This method clear off this entry // This method clear off this entry
_debug("updateDestinationIpAddress: remove destination %s", _ca->getLocalSDP()->get_remote_ip().c_str());
static_cast<D*>(this)->forgetDestination(_remote_ip, _remote_port); static_cast<D*>(this)->forgetDestination(_remote_ip, _remote_port);
// new destination is stored in call // new destination is stored in call
@ -473,12 +474,12 @@ namespace sfl {
_timestamp += _codecFrameSize; _timestamp += _codecFrameSize;
if (!_audiolayer) { if (!_audiolayer) {
_debug ("No audiolayer available for MIC\n"); _debug ("No audiolayer available for MIC");
return; return;
} }
if (!_audiocodec) { if (!_audiocodec) {
_debug ("No audiocodec available for MIC\n"); _debug ("No audiocodec available for MIC");
return; return;
} }
@ -493,12 +494,12 @@ namespace sfl {
void AudioRtpSession<D>::receiveSpeakerData () void AudioRtpSession<D>::receiveSpeakerData ()
{ {
if (!_audiolayer) { if (!_audiolayer) {
_debug ("No audiolayer available for speaker\n"); _debug ("No audiolayer available for speaker");
return; return;
} }
if (!_audiocodec) { if (!_audiocodec) {
_debug ("No audiocodec available for speaker\n"); _debug ("No audiocodec available for speaker");
return; return;
} }
@ -524,7 +525,7 @@ namespace sfl {
template <typename D> template <typename D>
int AudioRtpSession<D>::startRtpThread () int AudioRtpSession<D>::startRtpThread ()
{ {
_debug("Starting main thread\n"); _debug("Starting main thread");
return start(_mainloopSemaphore); return start(_mainloopSemaphore);
} }
@ -553,7 +554,7 @@ namespace sfl {
if (_audiolayer == NULL) { if (_audiolayer == NULL) {
_debug("For some unknown reason, audiolayer is null, just as \ _debug("For some unknown reason, audiolayer is null, just as \
we were about to start the audio stream\n"); we were about to start the audio stream");
throw AudioRtpSessionException(); throw AudioRtpSessionException();
} }
@ -564,7 +565,7 @@ namespace sfl {
static_cast<D*>(this)->startRunning(); static_cast<D*>(this)->startRunning();
_debug ("Entering RTP mainloop for callid %s\n",_ca->getCallId().c_str()); _debug ("Entering RTP mainloop for callid %s",_ca->getCallId().c_str());
while (!testCancel()) { while (!testCancel()) {
@ -600,7 +601,7 @@ namespace sfl {
TimerPort::incTimer (threadSleep); TimerPort::incTimer (threadSleep);
} }
_debug ("Left RTP main loop for callid %s\n",_ca->getCallId().c_str()); _debug ("Left RTP main loop for callid %s",_ca->getCallId().c_str());
} }
} }

4
sflphone-common/src/audio/mainbuffer.cpp
View File

@ -563,7 +563,7 @@ void MainBuffer::stateInfo()
iter_call_id++; iter_call_id++;
} }
_debug ("%s\n", dbg_str.c_str()); _debug ("%s", dbg_str.c_str());
iter_call++; iter_call++;
} }
@ -597,7 +597,7 @@ void MainBuffer::stateInfo()
} }
} }
_debug ("%s\n", dbg_str.c_str()); _debug ("%s", dbg_str.c_str());
iter_buffer++; iter_buffer++;
} }

24
sflphone-common/src/audio/pulseaudio/pulselayer.cpp
View File

@ -494,8 +494,8 @@ void PulseLayer::writeToSpeaker (void)
urgentAvailBytes = _urgentRingBuffer.AvailForGet(); urgentAvailBytes = _urgentRingBuffer.AvailForGet();
// available bytes to be written in pulseaudio internal buffer
int writeableSize = pa_stream_writable_size (playback->pulseStream()); int writeableSize = pa_stream_writable_size (playback->pulseStream());
// _debug("PulseLayer writablesize : %i", writeableSize);
if (writeableSize < 0) if (writeableSize < 0)
_debug ("PulseLayer playback error : %s", pa_strerror (writeableSize)); _debug ("PulseLayer playback error : %s", pa_strerror (writeableSize));
@ -503,15 +503,10 @@ void PulseLayer::writeToSpeaker (void)
if (urgentAvailBytes > writeableSize) { if (urgentAvailBytes > writeableSize) {
// _debug("urgentAvailBytes: %i", urgentAvailBytes);
// _debug("Play Urgent!");
out = (SFLDataFormat*) pa_xmalloc (writeableSize); out = (SFLDataFormat*) pa_xmalloc (writeableSize);
_urgentRingBuffer.Get (out, writeableSize, 100); _urgentRingBuffer.Get (out, writeableSize, 100);
// pa_threaded_mainloop_lock (m);
pa_stream_write (playback->pulseStream(), out, writeableSize, NULL, 0, PA_SEEK_RELATIVE); pa_stream_write (playback->pulseStream(), out, writeableSize, NULL, 0, PA_SEEK_RELATIVE);
// pa_threaded_mainloop_unlock (m);
pa_xfree (out); pa_xfree (out);
@ -529,17 +524,16 @@ void PulseLayer::writeToSpeaker (void)
if (tone != 0) { if (tone != 0) {
// _debug("PlayTone writeableSize: %i", writeableSize);
if (playback->getStreamState() == PA_STREAM_READY) { if (playback->getStreamState() == PA_STREAM_READY) {
// _debug("Play Sine Tone!"); // _debug("writeableSize %d\n", writeableSize);
out = (SFLDataFormat*) pa_xmalloc (writeableSize); out = (SFLDataFormat*) pa_xmalloc (writeableSize);
int copied = tone->getNext (out, writeableSize / sizeof (SFLDataFormat), 100); int copied = tone->getNext (out, writeableSize / sizeof (SFLDataFormat), 100);
// pa_threaded_mainloop_lock (m);
pa_stream_write (playback->pulseStream(), out, copied * sizeof (SFLDataFormat), NULL, 0, PA_SEEK_RELATIVE); int returnValue = pa_stream_write (playback->pulseStream(), out, copied * sizeof (SFLDataFormat), NULL, 0, PA_SEEK_RELATIVE);
// pa_threaded_mainloop_unlock (m);
// _debug("return value %d\n", returnValue);
pa_xfree (out); pa_xfree (out);
@ -549,15 +543,13 @@ void PulseLayer::writeToSpeaker (void)
else if (file_tone != 0) { else if (file_tone != 0) {
if (playback->getStreamState() == PA_STREAM_READY) { if (playback->getStreamState() == PA_STREAM_READY) {
// _debug("Play File Tone!");
out = (SFLDataFormat*) pa_xmalloc (writeableSize); out = (SFLDataFormat*) pa_xmalloc (writeableSize);
int copied = file_tone->getNext (out, writeableSize / sizeof (SFLDataFormat), 100); int copied = file_tone->getNext (out, writeableSize / sizeof (SFLDataFormat), 100);
// pa_threaded_mainloop_lock (m);
pa_stream_write (playback->pulseStream(), out, copied * sizeof (SFLDataFormat), NULL, 0, PA_SEEK_RELATIVE); pa_stream_write (playback->pulseStream(), out, copied * sizeof (SFLDataFormat), NULL, 0, PA_SEEK_RELATIVE);
pa_xfree (out); pa_xfree (out);
// pa_threaded_mainloop_unlock (m);
} }

2
sflphone-common/src/sip/sdp.h
View File

@ -197,7 +197,7 @@ class Sdp {
* Set remote's IP addr. [not protected] * Set remote's IP addr. [not protected]
* @param ip The remote IP address * @param ip The remote IP address
*/ */
void set_remote_ip(const std::string& ip) { _remote_ip_addr = ip; } void set_remote_ip(const std::string& ip) { _remote_ip_addr = ip; }
/** /**
* Return IP of destination [mutex protected] * Return IP of destination [mutex protected]

48
sflphone-common/src/sip/sipvoiplink.cpp
View File

@ -248,7 +248,7 @@ SIPVoIPLink::SIPVoIPLink (const AccountID& accountID)
SIPVoIPLink::~SIPVoIPLink() SIPVoIPLink::~SIPVoIPLink()
{ {
_debug("SIPVoIPLink destructor called"); _debug("UserAgent: SIPVoIPLink destructor called");
terminate(); terminate();
} }
@ -257,7 +257,7 @@ SIPVoIPLink* SIPVoIPLink::instance (const AccountID& id)
{ {
if (!_instance) { if (!_instance) {
_debug ("Create new SIPVoIPLink instance"); _debug ("UserAgent: Create new SIPVoIPLink instance");
_instance = new SIPVoIPLink (id); _instance = new SIPVoIPLink (id);
} }
@ -270,7 +270,7 @@ void SIPVoIPLink::decrementClients (void)
if (_clients == 0) { if (_clients == 0) {
_debug("No SIP account anymore, terminate SIPVoIPLink"); _debug("UserAgent: No SIP account anymore, terminate SIPVoIPLink");
terminate(); terminate();
SIPVoIPLink::_instance=NULL; SIPVoIPLink::_instance=NULL;
} }
@ -299,10 +299,10 @@ bool SIPVoIPLink::init()
void void
SIPVoIPLink::terminate() SIPVoIPLink::terminate()
{ {
_debug ("Terminating SIPVoIPLink"); _debug ("UserAgent: Terminating SIPVoIPLink");
if (_evThread) { if (_evThread) {
_debug ("Deleting sip eventThread"); _debug ("UserAgent: Deleting sip eventThread");
delete _evThread; delete _evThread;
_evThread = NULL; _evThread = NULL;
} }
@ -310,7 +310,7 @@ SIPVoIPLink::terminate()
/* Clean shutdown of pjsip library */ /* Clean shutdown of pjsip library */
if (initDone()) { if (initDone()) {
_debug ("Shuting down PJSIP"); _debug ("UserAgent: Shuting down PJSIP");
pjsip_shutdown(); pjsip_shutdown();
} }
@ -386,7 +386,7 @@ std::string SIPVoIPLink::getInterfaceAddrFromName(std::string ifaceName) {
struct in_addr *addr_in; struct in_addr *addr_in;
if((fd = socket (AF_INET, SOCK_DGRAM,0)) < 0) if((fd = socket (AF_INET, SOCK_DGRAM,0)) < 0)
_debug("getInterfaceAddrFromName error could not open socket\n"); _debug("UserAgent: getInterfaceAddrFromName error could not open socket\n");
memset (&ifr, 0, sizeof (struct ifreq)); memset (&ifr, 0, sizeof (struct ifreq));
@ -394,7 +394,7 @@ std::string SIPVoIPLink::getInterfaceAddrFromName(std::string ifaceName) {
ifr.ifr_addr.sa_family = AF_INET; ifr.ifr_addr.sa_family = AF_INET;
if((err = ioctl(fd, SIOCGIFADDR, &ifr)) < 0) if((err = ioctl(fd, SIOCGIFADDR, &ifr)) < 0)
_debug("getInterfaceAddrFromName use default interface (0.0.0.0)\n"); _debug("UserAgent: getInterfaceAddrFromName use default interface (0.0.0.0)\n");
saddr_in = (struct sockaddr_in *)&ifr.ifr_addr; saddr_in = (struct sockaddr_in *)&ifr.ifr_addr;
addr_in = &(saddr_in->sin_addr); addr_in = &(saddr_in->sin_addr);
@ -445,7 +445,7 @@ int SIPVoIPLink::sendRegister (AccountID id)
account = dynamic_cast<SIPAccount *> (Manager::instance().getAccount (id)); account = dynamic_cast<SIPAccount *> (Manager::instance().getAccount (id));
if (account == NULL) { if (account == NULL) {
_debug ("In sendRegister: account is null"); _debug ("UserAgent: In sendRegister: account is null");
return false; return false;
} }
@ -1021,8 +1021,8 @@ int SIPVoIPLink::inv_session_reinvite (SIPCall *call, std::string direction)
// TODO Restore Re-Invite // TODO Restore Re-Invite
// status = call->getLocalSDP()->create_initial_offer(); // status = call->getLocalSDP()->create_initial_offer();
if (status != PJ_SUCCESS) // if (status != PJ_SUCCESS)
return 1; // !PJ_SUCCESS // return 1; // !PJ_SUCCESS
pjmedia_sdp_media_remove_all_attr (local_sdp->media[0], "sendrecv"); pjmedia_sdp_media_remove_all_attr (local_sdp->media[0], "sendrecv");
@ -1802,7 +1802,6 @@ pj_status_t SIPVoIPLink::enable_dns_srv_resolver (pjsip_endpoint *endpt, pj_dns_
bool SIPVoIPLink::pjsip_init() bool SIPVoIPLink::pjsip_init()
{ {
pj_status_t status; pj_status_t status;
int errPjsip = 0;
pjsip_inv_callback inv_cb; pjsip_inv_callback inv_cb;
pj_str_t accepted; pj_str_t accepted;
std::string name_mod; std::string name_mod;
@ -2221,7 +2220,7 @@ bool SIPVoIPLink::createSipTransport(AccountID id)
int trns = remoteSipUri.find(";transport"); int trns = remoteSipUri.find(";transport");
std::string remoteAddr = remoteSipUri.substr(sips, trns-sips); std::string remoteAddr = remoteSipUri.substr(sips, trns-sips);
// Nothing to do, TLS listener already created at pjsip's startup and TLS connection\ // Nothing to do, TLS listener already created at pjsip's startup and TLS connection
// is automatically handled in pjsip when sending registration messages. // is automatically handled in pjsip when sending registration messages.
if(createTlsTransport(id, remoteAddr) != PJ_SUCCESS) if(createTlsTransport(id, remoteAddr) != PJ_SUCCESS)
return false; return false;
@ -2749,9 +2748,7 @@ pj_status_t SIPVoIPLink::createAlternateUdpTransport (AccountID id)
void SIPVoIPLink::shutdownSipTransport(const AccountID& accountID) void SIPVoIPLink::shutdownSipTransport(const AccountID& accountID)
{ {
_debug("Shutdown Sip Transport"); _debug("UserAgent: Shutdown Sip Transport");
pj_status_t status = 0;
SIPAccount* account = dynamic_cast<SIPAccount *> (Manager::instance().getAccount (accountID)); SIPAccount* account = dynamic_cast<SIPAccount *> (Manager::instance().getAccount (accountID));
@ -2915,23 +2912,9 @@ void set_voicemail_info (AccountID account, pjsip_msg_body *body)
void SIPVoIPLink::handle_reinvite (SIPCall *call) void SIPVoIPLink::handle_reinvite (SIPCall *call)
{ {
/*
// Close the previous RTP session
call->getAudioRtp()->stop ();
call->setAudioStart (false);
_debug ("Create new rtp session from handle_reinvite : %s:%i", call->getLocalIp().c_str(), call->getLocalAudioPort()); _debug ("UserAgent: handle_reinvite");
try {
call->getAudioRtp()->initAudioRtpSession (call);
} catch (...) {
_debug ("! SIP Failure: Unable to create RTP Session (%s:%d)", __FILE__, __LINE__);
}
*/
_debug("Handle reINVITE");
call->getAudioRtp()->updateDestinationIpAddress();
} }
// This callback is called when the invite session state has changed // This callback is called when the invite session state has changed
@ -3134,7 +3117,7 @@ void call_on_state_changed (pjsip_inv_session *inv, pjsip_event *e)
// This callback is called after SDP offer/answer session has completed. // This callback is called after SDP offer/answer session has completed.
void call_on_media_update (pjsip_inv_session *inv, pj_status_t status) void call_on_media_update (pjsip_inv_session *inv, pj_status_t status)
{ {
_debug ("call_on_media_update"); _debug ("UserAgent: call_on_media_update");
const pjmedia_sdp_session *local_sdp; const pjmedia_sdp_session *local_sdp;
const pjmedia_sdp_session *remote_sdp; const pjmedia_sdp_session *remote_sdp;
@ -3182,6 +3165,7 @@ void call_on_media_update (pjsip_inv_session *inv, pj_status_t status)
// Set remote ip / port // Set remote ip / port
call->getLocalSDP()->set_media_transport_info_from_remote_sdp (remote_sdp); call->getLocalSDP()->set_media_transport_info_from_remote_sdp (remote_sdp);
call->getAudioRtp()->updateDestinationIpAddress();
// Get the crypto attribute containing srtp's cryptographic context (keys, cipher) // Get the crypto attribute containing srtp's cryptographic context (keys, cipher)
CryptoOffer crypto_offer; CryptoOffer crypto_offer;