diff --git a/sflphone-common/src/audio/audiortp.cpp b/sflphone-common/src/audio/audiortp.cpp
index 51f7f7286e7662f22c2ae4cdd8c18131e6307fef..6ed965723b49260918130b4dc79ecb39d8685059 100644
--- a/sflphone-common/src/audio/audiortp.cpp
+++ b/sflphone-common/src/audio/audiortp.cpp
@@ -127,14 +127,6 @@ AudioRtp::closeRtpSession ()
return true;
}
-
-AudioRtpRTX*
-AudioRtp::getRTX()
-{
- return _RTXThread;
-}
-
-
void
AudioRtp::setRecording()
{
@@ -144,10 +136,6 @@ AudioRtp::setRecording()
}
-
-
-
-
////////////////////////////////////////////////////////////////////////////////
// AudioRtpRTX Class //
////////////////////////////////////////////////////////////////////////////////
diff --git a/sflphone-common/src/audio/audiortp.h b/sflphone-common/src/audio/audiortp.h
index e40f7a39798c0f154afdf32de62418d81b2b7055..49330c2fd8a90781baadef125d3ada416dec4697 100644
--- a/sflphone-common/src/audio/audiortp.h
+++ b/sflphone-common/src/audio/audiortp.h
@@ -256,11 +256,6 @@ class AudioRtp {
*/
bool closeRtpSession( void );
- /**
- * Return a pointer to the RTP thead
- */
- AudioRtpRTX* getRTX();
-
/**
* Start recording
*/
@@ -268,6 +263,10 @@ class AudioRtp {
friend class RtpTest;
+ /**
+ *
+ */
+ inline AudioRtpRTX * getAudioRtpRtxThread(void) { return _RTXThread; }
private:
// copy constructor
AudioRtp(const AudioRtp& rh);
diff --git a/sflphone-common/src/sdp.cpp b/sflphone-common/src/sdp.cpp
index 1a4005d4c722e06fa23ba7e58e18a2674a182820..9256b238b8ca8db8dd1cf6341e296940d5b1c4da 100644
--- a/sflphone-common/src/sdp.cpp
+++ b/sflphone-common/src/sdp.cpp
@@ -195,7 +195,7 @@ int Sdp::receiving_initial_offer (pjmedia_sdp_session* remote)
}
// Retrieve some useful remote information
- this->fetch_media_transport_info_from_remote_sdp (remote);
+ this->set_media_transport_info_from_remote_sdp (remote);
status = pjmedia_sdp_neg_create_w_remote_offer (_pool,
get_local_sdp_session(), remote, &_negociator);
@@ -314,8 +314,6 @@ void Sdp::set_negociated_offer (const pjmedia_sdp_session *sdp)
_negociated_offer = (pjmedia_sdp_session*) sdp;
- //this->fetch_remote_ip_from_sdp ((pjmedia_sdp_session*)sdp);
-
codecs_list = Manager::instance().getCodecDescriptorMap().getCodecsMap();
// retrieve the media information
@@ -358,7 +356,7 @@ AudioCodec* Sdp::get_session_media (void)
AudioCodec *codec = NULL;
std::vector<sdpMedia*> media_list;
- _debug ("sdp line 314 - get_session_media ()\n");
+ _debug ("sdp line %d - get_session_media ()\n", __LINE__);
media_list = get_session_media_list ();
nb_media = media_list.size();
@@ -470,17 +468,15 @@ std::string Sdp::convert_int_to_string (int value)
return result.str();
}
-void Sdp::fetch_remote_ip_from_sdp (pjmedia_sdp_session *r_sdp)
+void Sdp::set_remote_ip_from_sdp (const pjmedia_sdp_session *r_sdp)
{
- std::string remote_ip;
-
- remote_ip = r_sdp->conn->addr.ptr;
+ std::string remote_ip(r_sdp->conn->addr.ptr, r_sdp->conn->addr.slen);
_debug (" Remote IP from fetching SDP: %s\n", remote_ip.c_str());
this->set_remote_ip (remote_ip);
}
-void Sdp::fetch_remote_audio_port_from_sdp (pjmedia_sdp_media *r_media)
+void Sdp::set_remote_audio_port_from_sdp (pjmedia_sdp_media *r_media)
{
int remote_port;
@@ -490,7 +486,7 @@ void Sdp::fetch_remote_audio_port_from_sdp (pjmedia_sdp_media *r_media)
this->set_remote_audio_port (remote_port);
}
-void Sdp::fetch_media_transport_info_from_remote_sdp (pjmedia_sdp_session *remote_sdp)
+void Sdp::set_media_transport_info_from_remote_sdp (const pjmedia_sdp_session *remote_sdp)
{
_debug ("Fetching media from sdp\n");
@@ -504,12 +500,12 @@ void Sdp::fetch_media_transport_info_from_remote_sdp (pjmedia_sdp_session *remot
return;
}
- this->fetch_remote_audio_port_from_sdp (r_media);
+ this->set_remote_audio_port_from_sdp (r_media);
- this->fetch_remote_ip_from_sdp (remote_sdp);
+ this->set_remote_ip_from_sdp (remote_sdp);
}
-void Sdp::get_remote_sdp_media_from_offer (pjmedia_sdp_session* remote_sdp, pjmedia_sdp_media** r_media)
+void Sdp::get_remote_sdp_media_from_offer (const pjmedia_sdp_session* remote_sdp, pjmedia_sdp_media** r_media)
{
int count, i;
diff --git a/sflphone-common/src/sdp.h b/sflphone-common/src/sdp.h
index 3a0eee5c6e67f4ed7c90757d21368bf2539d1598..ec3178955b57c5370872d43ada2c92302f9f18af 100644
--- a/sflphone-common/src/sdp.h
+++ b/sflphone-common/src/sdp.h
@@ -179,7 +179,7 @@ class Sdp {
*/
unsigned int get_remote_audio_port() { return _remote_audio_port; }
- void fetch_media_transport_info_from_remote_sdp (pjmedia_sdp_session *remote_sdp);
+ void set_media_transport_info_from_remote_sdp (const pjmedia_sdp_session *remote_sdp);
std::vector<sdpMedia*> get_session_media_list (void) { return _session_media; }
@@ -298,11 +298,11 @@ class Sdp {
std::string convert_int_to_string (int value);
- void fetch_remote_ip_from_sdp (pjmedia_sdp_session *r_sdp);
+ void set_remote_ip_from_sdp (const pjmedia_sdp_session *r_sdp);
- void fetch_remote_audio_port_from_sdp (pjmedia_sdp_media *r_media);
+ void set_remote_audio_port_from_sdp (pjmedia_sdp_media *r_media);
- void get_remote_sdp_media_from_offer (pjmedia_sdp_session* r_sdp, pjmedia_sdp_media** r_media);
+ void get_remote_sdp_media_from_offer (const pjmedia_sdp_session* r_sdp, pjmedia_sdp_media** r_media);
//////////////////////////////////////////////////////////////////3
////////////////////////////////////////////////////////////////////
diff --git a/sflphone-common/src/sipvoiplink.cpp b/sflphone-common/src/sipvoiplink.cpp
index 6d8d8f4b964a7754c4ce4baa051be474303c00ad..7692fab7d5544a55a96dd1c68df8e0f632fb8d13 100644
--- a/sflphone-common/src/sipvoiplink.cpp
+++ b/sflphone-common/src/sipvoiplink.cpp
@@ -341,8 +341,8 @@ int SIPVoIPLink::sendRegister (AccountID id)
account = dynamic_cast<SIPAccount *> (Manager::instance().getAccount (id));
hostname = account->getHostname();
username = account->getUsername();
- password = account->getPassword();
-
+ password = account->getPassword();
+
_mutexSIP.enterMutex();
/* Get the client registration information for this particular account */
@@ -545,16 +545,9 @@ SIPVoIPLink::newOutgoingCall (const CallID& id, const std::string& toUrl)
}
if (SIPOutgoingInvite (call)) {
-
call->setConnectionState (Call::Progressing);
call->setState (Call::Active);
addCall (call);
-
- // _audiortp->start();
-
- // call->setAudioStart ( true );
-
-
} else {
delete call;
call = 0;
@@ -841,17 +834,8 @@ SIPVoIPLink::offhold (const CallID& id)
if (status != PJ_SUCCESS)
return false;
- // Enable audio
- _debug ("* SIP Info: Starting AudioRTP when offhold\n");
-
call->setState (Call::Active);
- // it's sure that this is the current call id...
-
- _audiortp->start();
-
- call->setAudioStart (true);
-
return true;
}
@@ -1287,51 +1271,16 @@ SIPVoIPLink::SIPCallAnswered (SIPCall *call, pjsip_rx_data *rdata)
_debug ("SIPCallAnswered\n");
- pjmedia_sdp_session *r_sdp;
-
if (!call) {
_debug ("! SIP Failure: unknown call\n");
return;
}
if (call->getConnectionState() != Call::Connected) {
- _debug ("Get remote SDP from offer\n");
- get_remote_sdp_from_offer (rdata, &r_sdp);
-
- if (r_sdp==NULL) {
- _debug ("SIP Failure: no remote sdp session\n");
- return;
- }
-
- _debug ("Get remote media information from offer\n");
-
- call->getLocalSDP()->fetch_media_transport_info_from_remote_sdp (r_sdp);
-
- // _audiortp->getRTX()->setRtpSessionRemoteIp();
-
_debug ("Update call state , id = %s\n", call->getCallId().c_str());
call->setConnectionState (Call::Connected);
call->setState (Call::Active);
-
Manager::instance().peerAnsweredCall (call->getCallId());
-
- if (Manager::instance().isCurrentCall (call->getCallId())) {
- /*
- _debug("CREATE NEW RTP SESSION FROM SIPCALLANSWERED\n");
- try {
- _audiortp->createNewSession (call);
- call->setAudioStart(true);
- } catch (...) {
- _debug("Failed to create rtp thread from answer\n");
- }
- */
-
- call->setAudioStart (true);
-
- _audiortp->start();
-
- }
-
} else {
_debug ("* SIP Info: Answering call (on/off hold to send ACK)\n");
}
@@ -1405,7 +1354,6 @@ bool SIPVoIPLink::new_ip_to_ip_call (const CallID& id, const std::string& to)
_debug ("! SIP Failure: Unable to create RTP Session in SIPVoIPLink::new_ip_to_ip_call (%s:%d)\n", __FILE__, __LINE__);
}
-
// Generate the contact URI
// uri_contact << "<" << uri_from << ":" << call->getLocalSDP()->get_local_extern_audio_port() << ">";
uri_contact << "<" << uri_from << ":" << _localExternPort << ">";
@@ -1450,10 +1398,6 @@ bool SIPVoIPLink::new_ip_to_ip_call (const CallID& id, const std::string& to)
call->setState (Call::Active);
addCall (call);
- // _audiortp->start();
-
- // call->setAudioStart ( true );
-
return true;
} else
return false;
@@ -1813,10 +1757,6 @@ int SIPVoIPLink::createUDPServer (void)
return PJ_SUCCESS;
}
-
-
-
-
bool SIPVoIPLink::loadSIPLocalIP()
{
@@ -1940,7 +1880,6 @@ void set_voicemail_info (AccountID account, pjsip_msg_body *body)
void SIPVoIPLink::handle_reinvite (SIPCall *call)
{
-
// Close the previous RTP session
_audiortp->closeRtpSession ();
call->setAudioStart (false);
@@ -1952,12 +1891,6 @@ void SIPVoIPLink::handle_reinvite (SIPCall *call)
} catch (...) {
_debug ("! SIP Failure: Unable to create RTP Session (%s:%d)\n", __FILE__, __LINE__);
}
-
- _audiortp->start();
-
- call->setAudioStart (true);
-
-
}
@@ -2121,21 +2054,18 @@ void call_on_state_changed (pjsip_inv_session *inv, pjsip_event *e)
break;
}
}
-
-
}
}
void call_on_media_update (pjsip_inv_session *inv, pj_status_t status)
{
-
_debug ("--------------------- call_on_media_update --------------------- \n");
-
- AccountID accId;
- SIPVoIPLink *link;
-
- const pjmedia_sdp_session *r_sdp;
- SIPCall *call;
+
+ const pjmedia_sdp_session *local_sdp;
+ const pjmedia_sdp_session *remote_sdp;
+
+ SIPVoIPLink * link = NULL;
+ SIPCall * call;
if (status != PJ_SUCCESS) {
_debug ("Error while negociating the offer\n");
@@ -2143,48 +2073,38 @@ void call_on_media_update (pjsip_inv_session *inv, pj_status_t status)
}
// Get the new sdp, result of the negociation
- pjmedia_sdp_neg_get_active_local (inv->neg, &r_sdp);
-
+ pjmedia_sdp_neg_get_active_local (inv->neg, &local_sdp);
+ pjmedia_sdp_neg_get_active_remote(inv->neg, &remote_sdp);
+
call = reinterpret_cast<SIPCall *> (inv->mod_data[getModId() ]);
if (!call) {
_debug ("Call declined by peer, SDP negociation stopped\n");
return;
}
-
+
// Clean the resulting sdp offer to create a new one (in case of a reinvite)
call->getLocalSDP()->clean_session_media();
-
+
// Set the fresh negociated one
- call->getLocalSDP()->set_negociated_offer (r_sdp);
-
- // call->getLocalSDP()->fetch_media_transport_info_from_remote_sdp( r_sdp );
- // call->getLocalSDP()->fetch_remote_ip_from_sdp( r_sdp );
- // call->getLocalSDP()->fetch_remote_audio_port_from_sdp( r_sdp );
-
- accId = Manager::instance().getAccountFromCall (call->getCallId());
-
- link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink (accId));
-
- if (!link) {
- _debug ("No Voip Account Link, impossible to init RTP session with SDP info");
+ call->getLocalSDP()->set_negociated_offer (local_sdp);
+
+ // Set remote ip / port
+ call->getLocalSDP()->set_media_transport_info_from_remote_sdp (remote_sdp);
+
+ link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getSIPAccountLink());
+ if(link == NULL) {
+ _debug ("Failed to get sip link\n");
return;
}
-
- if (call->getState() != Call::Hold) {
- _debug ("Set media parameters in RTP session\n");
- // link->_audiortp->getRTX()->setRtpSessionMedia();
- // link->_audiortp->getRTX()->setRtpSessionRemoteIp();
-
- // link->_audiortp->start();
-
- // call->setAudioStart ( true );
- // link->_audiortp->start();
- } else {
- _debug ("Didn't set RTP parameters since call is on hold\n");
+
+ try {
+ link->getAudioRtp()->start();
+ call->setAudioStart (true);
+ } catch(exception& rtpException) {
+ _debug("%s\n", rtpException.what());
}
-
}
void call_on_forked (pjsip_inv_session *inv, pjsip_event *e)
@@ -2924,7 +2844,7 @@ void on_rx_offer (pjsip_inv_session *inv, const pjmedia_sdp_session *offer)
#ifdef CAN_REINVITE
- _debug ("reinvite SIP\n");
+ _debug ("%s (%d): on_rx_offer REINVITE\n", __FILE__, __LINE__);
SIPCall *call;
pj_status_t status;
diff --git a/sflphone-common/src/sipvoiplink.h b/sflphone-common/src/sipvoiplink.h
index 028fcd8a2fd6173135816df5c1682348813daaa4..3a0387c36ef7f044f3a5952601c6b5f6b27b8bd9 100644
--- a/sflphone-common/src/sipvoiplink.h
+++ b/sflphone-common/src/sipvoiplink.h
@@ -284,8 +284,8 @@ class SIPVoIPLink : public VoIPLink
/** when we init the listener, how many times we try to bind a port? */
int _nbTryListenAddr;
- /** Starting sound */
- AudioRtp* _audiortp;
+ /** Returns a pointer to the AudioRtp object */
+ inline AudioRtp * getAudioRtp(void) { return _audiortp; }
/** Increment the number of SIP account connected to this link */
void incrementClients (void) { _clients++; }
@@ -363,6 +363,9 @@ class SIPVoIPLink : public VoIPLink
std::string getLocalIP() {return _localExternAddress;}
+ /** Starting sound */
+ AudioRtp* _audiortp;
+
/** For registration use only */
int _regPort;