diff --git a/src/audio/audiortp.cpp b/src/audio/audiortp.cpp
index 8db80f7ae35d682d296f14addc3a8a12944dc866..0b166c6a0940efe30a5350b2981f74b9320a2d1d 100644
--- a/src/audio/audiortp.cpp
+++ b/src/audio/audiortp.cpp
@@ -201,7 +201,6 @@ AudioRtpRTX::initAudioRtpSession (void)
remotePort = _ca->getLocalSDP()->get_remote_audio_port();
_debug("Init audio RTP session - remote IP = %s\n", remoteIP.c_str());
ost::InetHostAddress remote_ip(remoteIP.c_str());
- _debug("Init audio RTP session - remote IP = %s\n", remoteIP.c_str());
if (!remote_ip) {
_debug("! ARTP Thread Error: Target IP address [%s] is not correct!\n", remoteIP.data());
return;
diff --git a/src/sdp.cpp b/src/sdp.cpp
index bf2f92858842699ba02748d8b7137408229a5219..537b4b18a801a61300236cd432204465f646f51d 100644
--- a/src/sdp.cpp
+++ b/src/sdp.cpp
@@ -40,12 +40,13 @@ Sdp::Sdp( pj_pool_t *pool )
Sdp::~Sdp() {
- unsigned int k;
+ //unsigned int k;
+ /*
for( k=0; k<_session_media.size(); k++ ){
delete _session_media[k];
_session_media[k] = 0;
- }
+ }*/
//for( k=0; k<_local_media_cap.size(); k++ ){
// delete _local_media_cap[k];
@@ -53,8 +54,8 @@ Sdp::~Sdp() {
//}
}
-void Sdp::set_media_descriptor_line( sdpMedia *media,
- pjmedia_sdp_media** p_med ) {
+void Sdp::set_media_descriptor_line( sdpMedia *media, pjmedia_sdp_media** p_med ) {
+
pjmedia_sdp_media* med;
pjmedia_sdp_rtpmap rtpmap;
pjmedia_sdp_attr *attr;
@@ -354,7 +355,10 @@ void Sdp::set_local_media_capabilities () {
CodecsMap codecs_list;
CodecsMap::iterator iter;
- _debug ("Fetch local media capabilities .......... %i" , get_local_extern_audio_port());
+ // Clean it first
+ _local_media_cap.clear();
+
+ _debug ("Fetch local media capabilities .......... %i\n" , get_local_extern_audio_port());
/* Only one audio media used right now */
audio = new sdpMedia(MIME_TYPE_AUDIO);
@@ -399,7 +403,7 @@ void Sdp::fetch_remote_ip_from_sdp (pjmedia_sdp_session *r_sdp) {
std::string remote_ip;
remote_ip = r_sdp->conn->addr.ptr;
- _debug(" Remote Audio IP: %s\n", remote_ip.c_str());
+ _debug("************************************************** Remote Audio IP: %s\n", remote_ip.c_str());
this->set_remote_ip(remote_ip);
}
diff --git a/src/sipvoiplink.cpp b/src/sipvoiplink.cpp
index a3b308405b75bc33da04ed4808399248003184ae..d9abc2b42dc8c1dc14cc478f21c4774f2df5e474 100644
--- a/src/sipvoiplink.cpp
+++ b/src/sipvoiplink.cpp
@@ -25,7 +25,7 @@
#include "sipaccount.h"
#include "audio/audiortp.h"
-#define CAN_REINVITE 1
+//#define CAN_REINVITE 1
/**************** EXTERN VARIABLES AND FUNCTIONS (callbacks) **************************/
@@ -469,7 +469,7 @@ SIPVoIPLink::newOutgoingCall(const CallID& id, const std::string& toUrl)
return call;
}
- bool
+ bool
SIPVoIPLink::answer(const CallID& id)
{
@@ -500,7 +500,7 @@ SIPVoIPLink::answer(const CallID& id)
PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
status = pjsip_inv_send_msg(inv_session, tdata);
PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
-
+
// Start the RTP sessions
_debug("SIPVoIPLink::answer: Starting AudioRTP when answering : call %s \n", call->getCallId().c_str());
if (_audiortp->createNewSession(call) >= 0) {
@@ -516,7 +516,7 @@ SIPVoIPLink::answer(const CallID& id)
// Create and send a 488/Not acceptable here
// because the SDP negociation failed
status = pjsip_inv_answer( inv_session, PJSIP_SC_NOT_ACCEPTABLE_HERE, NULL, NULL,
- &tdata );
+ &tdata );
PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
status = pjsip_inv_send_msg( inv_session, tdata );
PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
@@ -660,7 +660,7 @@ int SIPVoIPLink::inv_session_reinvite (SIPCall *call, std::string direction) {
// Build the local SDP offer
status = call->getLocalSDP()->create_initial_offer( );
pjmedia_sdp_media_remove_all_attr(local_sdp->media[0], "sendrecv");
- attr = pjmedia_sdp_attr_create(_pool, "sendonly", NULL);
+ attr = pjmedia_sdp_attr_create(_pool, direction.c_str(), NULL);
pjmedia_sdp_media_add_attr(local_sdp->media[0], attr);
PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
@@ -691,7 +691,7 @@ SIPVoIPLink::offhold(const CallID& id)
}
/* Create re-INVITE with new offer */
- status = inv_session_reinvite (call, "sendonly");
+ status = inv_session_reinvite (call, "sendrecv");
if (status != PJ_SUCCESS)
return false;
@@ -1075,7 +1075,7 @@ std::string SIPVoIPLink::getSipTo(const std::string& to_url, std::string hostnam
{
pjmedia_sdp_session *r_sdp;
-
+
//SIPCall* call = dynamic_cast<SIPCall *>(theCall);//findSIPCallWithCid(event->cid);
if (!call) {
_debug("! SIP Failure: unknown call\n");
@@ -1084,13 +1084,16 @@ std::string SIPVoIPLink::getSipTo(const std::string& to_url, std::string hostnam
//call->setDid(event->did);
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");
+ _debug("SIP Failure: no remote sdp session\n");
return;
}
- call->getLocalSDP()->fetch_media_transport_info_from_remote_sdp (r_sdp);
+ _debug ("Get remote media information from offer\n");
+ //call->getLocalSDP()->fetch_media_transport_info_from_remote_sdp (r_sdp);
+ _debug ("Update call state\n");
call->setConnectionState(Call::Connected);
call->setState(Call::Active);
@@ -1512,14 +1515,21 @@ std::string SIPVoIPLink::getSipTo(const std::string& to_url, std::string hostnam
void call_on_state_changed( pjsip_inv_session *inv, pjsip_event *e){
- PJ_UNUSED_ARG(inv);
+ SIPCall *call;
+ AccountID accId;
+ SIPVoIPLink *link;
+ pjsip_rx_data *rdata;
_debug (" ***************************** NEW CALL STATE %i **************************\n", inv->state);
- SIPCall *call = reinterpret_cast<SIPCall*> (inv->mod_data[_mod_ua.id]);
+ /* Retrieve the call information */
+ call = reinterpret_cast<SIPCall*> (inv->mod_data[_mod_ua.id]);
if(!call)
return;
+ //Retrieve the body message
+ rdata = e->body.tsx_state.src.rdata;
+
/* If this is an outgoing INVITE that was created because of
* REFER/transfer, send NOTIFY to transferer.
*/
@@ -1576,6 +1586,54 @@ std::string SIPVoIPLink::getSipTo(const std::string& to_url, std::string hostnam
}
}
}
+ else {
+
+ // The call is ringing
+ if (inv->state == PJSIP_INV_STATE_EARLY){
+ _debug ("*************************** PJSIP_INV_STATE_EARLY - PEER RINGING ***********************************\n");
+ call->setConnectionState(Call::Ringing);
+ Manager::instance().peerRingingCall(call->getCallId());
+ }
+
+ // We receive a ACK - The connection is established
+ else if( inv->state == PJSIP_INV_STATE_CONFIRMED ){
+ _debug ("*************************** PJSIP_INV_STATE_CONFIRMED ***********************************\n");
+ accId = Manager::instance().getAccountFromCall(call->getCallId());
+ link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink(accId));
+ if (link)
+ link->SIPCallAnswered(call, rdata);
+ }
+
+ else if( inv->state == PJSIP_INV_STATE_DISCONNECTED ){
+ _debug ("*************************** PJSIP_INV_STATE_DISCONNECTED %i***********************************\n", inv->cause);
+ switch( inv->cause )
+ {
+ /* The call terminates normally - BYE / CANCEL */
+ case PJSIP_SC_OK:
+ case PJSIP_SC_DECLINE:
+ accId = Manager::instance().getAccountFromCall(call->getCallId());
+ link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink(accId));
+ if (link) {
+ link->SIPCallClosed(call);
+ }
+ break;
+
+ case PJSIP_SC_REQUEST_TIMEOUT:
+ // The host was probably unreachable: bad address, bad port, ...
+ break;
+ case PJSIP_SC_NOT_ACCEPTABLE_HERE:
+ // The SDP negociation failed
+ break;
+ default:
+
+ // The call terminated successfully; normal behaviour
+ // Core notification
+ break;
+ }
+ }
+
+
+ }
}
void call_on_media_update( pjsip_inv_session *inv, pj_status_t status) {
@@ -1591,7 +1649,7 @@ std::string SIPVoIPLink::getSipTo(const std::string& to_url, std::string hostnam
// Get the new sdp, result of the negociation
pjmedia_sdp_neg_get_active_local( inv->neg, &r_sdp );
-
+
call = reinterpret_cast<SIPCall *> (inv->mod_data[getModId()]);
// Clean the resulting sdp offer to create a new one (in case of a reinvite)
call->getLocalSDP()->clean_session_media();
@@ -1605,6 +1663,7 @@ std::string SIPVoIPLink::getSipTo(const std::string& to_url, std::string hostnam
void call_on_tsx_changed(pjsip_inv_session *inv, pjsip_transaction *tsx, pjsip_event *e){
+ /*
pjsip_rx_data *rdata;
AccountID accId;
SIPCall *call;
@@ -1621,7 +1680,6 @@ std::string SIPVoIPLink::getSipTo(const std::string& to_url, std::string hostnam
if (tsx->role == PJSIP_ROLE_UAC) {
-
switch (tsx->state) {
case PJSIP_TSX_STATE_TERMINATED:
if (tsx->status_code == 200 && pjsip_method_cmp(&tsx->method, pjsip_get_refer_method()) != 0) {
@@ -1636,10 +1694,11 @@ std::string SIPVoIPLink::getSipTo(const std::string& to_url, std::string hostnam
accId = Manager::instance().getAccountFromCall(call->getCallId());
link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink(accId));
if (link) {
+ _debug ("SIP transaction successfully terminated\n");
link->SIPCallAnswered(call, rdata);
} else {
}
-
+
} else if (tsx->status_code / 100 == 5) {
_debug("UserAgent: 5xx error message received\n");
}
@@ -1654,7 +1713,6 @@ std::string SIPVoIPLink::getSipTo(const std::string& to_url, std::string hostnam
if (msg->line.status.code == 180) {
_debug("UserAgent: Peer is ringing!\n");
-
call->setConnectionState(Call::Ringing);
Manager::instance().peerRingingCall(call->getCallId());
}
@@ -1721,10 +1779,11 @@ std::string SIPVoIPLink::getSipTo(const std::string& to_url, std::string hostnam
}
break;
default:
- _debug("default: %i\n", tsx->status_code);
+ _debug("transaction terminated: %i\n", tsx->state);
break;
} // end of switch
}
+ */
}
void regc_cb(struct pjsip_regc_cbparam *param){
@@ -2355,7 +2414,7 @@ std::string SIPVoIPLink::getSipTo(const std::string& to_url, std::string hostnam
}
-bool setCallAudioLocal(SIPCall* call, std::string localIP, bool stun, std::string server) {
+ bool setCallAudioLocal(SIPCall* call, std::string localIP, bool stun, std::string server) {
// Setting Audio
unsigned int callLocalAudioPort = RANDOM_LOCAL_PORT;
unsigned int callLocalExternAudioPort = callLocalAudioPort;
@@ -2376,7 +2435,7 @@ bool setCallAudioLocal(SIPCall* call, std::string localIP, bool stun, std::strin
call->getLocalSDP()->attribute_port_to_all_media (callLocalExternAudioPort);
return true;
-}
+ }
diff --git a/src/sipvoiplink.h b/src/sipvoiplink.h
index 693f20fd282bc226e783abc5a9a64c87e9310129..212a835c6bd4e72f4ca9a0bcd4f3ac62a5a310c3 100644
--- a/src/sipvoiplink.h
+++ b/src/sipvoiplink.h
@@ -41,7 +41,7 @@ class AudioRtp;
#define RANDOM_SIP_PORT rand() % 64000 + 1024
// To set the verbosity. From 0 (min) to 6 (max)
-#define PJ_LOG_LEVEL 6
+#define PJ_LOG_LEVEL 1
/**
* @file sipvoiplink.h