diff --git a/src/audio/audiortp.cpp b/src/audio/audiortp.cpp
index 829ca644444f110102a0d9b29c07d8093dc7cdcf..2ef3feea29d4ac4625dc38cd843e90a6fa97cd24 100644
--- a/src/audio/audiortp.cpp
+++ b/src/audio/audiortp.cpp
@@ -133,8 +133,9 @@ AudioRtpRTX::initAudioRtpSession (void)
}
if (!_sym) {
+ std::string localipConfig = _ca->getLocalIp();
ost::InetHostAddress local_ip(localipConfig.c_str());
- if (!_sessionRecv->addDestination (local_ip, (unsigned short), _ca->getLocalAudioPort()) ) {
+ if ( !_sessionRecv->addDestination(local_ip, (unsigned short) _ca->getLocalAudioPort()) ) {
_debug("RTX recv: could not connect to port %d\n", _ca->getLocalAudioPort());
return;
}
diff --git a/src/managerimpl.cpp b/src/managerimpl.cpp
index 88c5fc31788be627de28c6c981cc47780df052c4..3f2d1d1dc80c20f75d0174db835964131dfc02fd 100644
--- a/src/managerimpl.cpp
+++ b/src/managerimpl.cpp
@@ -718,12 +718,14 @@ ManagerImpl::peerAnsweredCall (CALLID id)
{
ost::MutexLock m(_mutex);
Call* call = getCall(id);
- call->setState(Call::Answered);
-
- stopTone();
- // switch current call
- switchCall(id);
- if (_gui) _gui->peerAnsweredCall(id);
+ if (call != 0) {
+ call->setState(Call::Answered);
+
+ stopTone();
+ // switch current call
+ switchCall(id);
+ if (_gui) _gui->peerAnsweredCall(id);
+ }
}
/**
@@ -735,11 +737,13 @@ ManagerImpl::peerRingingCall (CALLID id)
{
ost::MutexLock m(_mutex);
Call* call = getCall(id);
- call->setState(Call::Ringing);
+ if (call != 0) {
+ call->setState(Call::Ringing);
- // ring
- ringback();
- if (_gui) _gui->peerRingingCall(id);
+ // ring
+ ringback();
+ if (_gui) _gui->peerRingingCall(id);
+ }
return 1;
}
diff --git a/src/sipcall.cpp b/src/sipcall.cpp
index 02ce78d86b61823d1dd78e30dc7a7a35e278d71b..63fe2303f07dd293cec319bb322ad56eec2abad8 100644
--- a/src/sipcall.cpp
+++ b/src/sipcall.cpp
@@ -150,28 +150,30 @@ SipCall::setAudioCodec (AudioCodec* ac)
// newIncomingCall is called when the IP-Phone user receives a new call.
int
SipCall::newIncomingCall (eXosip_event_t *event) {
-
- _cid = event->cid;
- _did = event->did;
- _tid = event->tid;
- if (_did < 1 && _cid < 1) {
- return -1; /* not enough information for this event?? */
- }
-
- osip_strncpy (_textinfo, event->textinfo, 255);
+ _cid = event->cid;
+ _did = event->did;
+ _tid = event->tid;
- if (event->response != NULL) {
- _status_code = event->response->status_code;
- snprintf (_reason_phrase, 49, "%s", event->response->reason_phrase);
- }
+ if (_did < 1 && _cid < 1) {
+ return -1; /* not enough information for this event?? */
+ }
+
+ osip_strncpy (_textinfo, event->textinfo, 255);
+
+ if (event->response != NULL) {
+ _status_code = event->response->status_code;
+ snprintf (_reason_phrase, 49, "%s", event->response->reason_phrase);
+ _debug(" Status: %d %s\n", _status_code, _reason_phrase);
+ }
if (event->request != NULL) {
char *tmp = NULL;
osip_from_to_str (event->request->from, &tmp);
if (tmp != NULL) {
- snprintf (_remote_uri, 255, "%s", tmp);
+ snprintf (_remote_uri, 255, "%s\n", tmp);
+ _debug(" Remote URI: %s\n", _remote_uri);
osip_free (tmp);
}
}
@@ -199,6 +201,7 @@ SipCall::newIncomingCall (eXosip_event_t *event) {
sdp_connection_t *conn = eXosip_get_audio_connection (remote_sdp);
if (conn != NULL && conn->c_addr != NULL) {
snprintf (_remote_sdp_audio_ip, 49, "%s", conn->c_addr);
+ _debug(" Remote Audio IP: %s\n", _remote_sdp_audio_ip);
}
sdp_media_t *remote_med = eXosip_get_audio_media (remote_sdp);
@@ -211,6 +214,7 @@ SipCall::newIncomingCall (eXosip_event_t *event) {
}
_remote_sdp_audio_port = atoi (remote_med->m_port);
+ _debug(" Remote Audio Port: %d\n", _remote_sdp_audio_port);
char *tmp = NULL;
if (_remote_sdp_audio_port > 0 && _remote_sdp_audio_ip[0] != '\0') {
@@ -229,6 +233,7 @@ SipCall::newIncomingCall (eXosip_event_t *event) {
}
if (tmp != NULL) {
payload = atoi (tmp);
+ _debug(" Payload: %d\n", payload);
} else {
// Send 415 Unsupported media type
eXosip_call_send_answer (_tid, 415, NULL);
@@ -246,6 +251,7 @@ SipCall::newIncomingCall (eXosip_event_t *event) {
int i;
eXosip_lock ();
+ _debug("< Building Answer 183\n");
i = eXosip_call_build_answer (_tid, 183, &answer);
if (i == 0) {
i = sdp_complete_message (remote_sdp, answer);
@@ -253,6 +259,7 @@ SipCall::newIncomingCall (eXosip_event_t *event) {
osip_message_free (answer);
// Send 415 Unsupported media type
eXosip_call_send_answer (_tid, 415, NULL);
+ _debug("< Sending Answer 415\n");
} else {
/* start sending audio */
if (enable_audio == true) {
@@ -271,14 +278,17 @@ SipCall::newIncomingCall (eXosip_event_t *event) {
conn = eXosip_get_audio_connection (remote_sdp);
if (conn != NULL && conn->c_addr != NULL) {
snprintf (_remote_sdp_audio_ip, 49, "%s", conn->c_addr);
+ _debug(" Remote Audio IP: %s\n", _remote_sdp_audio_ip);
}
remote_med = eXosip_get_audio_media (remote_sdp);
if (remote_med != NULL && remote_med->m_port != NULL) {
_remote_sdp_audio_port = atoi (remote_med->m_port);
+ _debug(" Remote Audio Port: %d\n", _remote_sdp_audio_port);
}
local_med = eXosip_get_audio_media (local_sdp);
if (local_med != NULL && local_med->m_port != NULL) {
audio_port = atoi (local_med->m_port);
+ _debug(" Local Audio Port: %d\n", audio_port);
}
if (_remote_sdp_audio_port > 0
@@ -288,7 +298,7 @@ SipCall::newIncomingCall (eXosip_event_t *event) {
}
if (tmp != NULL) {
payload = atoi (tmp);
- _debug("SipCall::newIncomingCall: For incoming payload = %d\n", payload);
+ _debug(" Remote Payload: %d\n", payload);
setAudioCodec(_cdv->at(0)->alloc(payload, "")); // codec builder for the mic
}
if (tmp != NULL
@@ -301,18 +311,24 @@ SipCall::newIncomingCall (eXosip_event_t *event) {
sdp_analyse_attribute (remote_sdp, remote_med);
_local_sendrecv =
sdp_analyse_attribute (local_sdp, local_med);
+ _debug(" Remote SendRecv: %d\n", _remote_sendrecv);
+ _debug(" Local SendRecv: %d\n", _local_sendrecv);
if (_local_sendrecv == _SENDRECV) {
- if (_remote_sendrecv == _SENDONLY)
+ if (_remote_sendrecv == _SENDONLY) {
_local_sendrecv = _RECVONLY;
- else if (_remote_sendrecv == _RECVONLY)
+ }
+ else if (_remote_sendrecv == _RECVONLY) {
_local_sendrecv = _SENDONLY;
+ }
}
+ _debug(" Final Local SendRecv: %d\n", _local_sendrecv);
}
}
sdp_message_free (local_sdp);
}
i = eXosip_call_send_answer (_tid, 183, answer);
+ _debug(" < Sending answer 183\n");
}
if (i != 0) {
diff --git a/src/sipvoiplink.cpp b/src/sipvoiplink.cpp
index ef942fe897f0593b8eaee9d1d9c95f0eeff578cb..69311b79de4a2daf7cbb5613e09cab9efbb50850 100644
--- a/src/sipvoiplink.cpp
+++ b/src/sipvoiplink.cpp
@@ -628,6 +628,7 @@ SipVoIPLink::getEvent (void)
switch (event->type) {
// IP-Phone user receives a new call
case EXOSIP_CALL_INVITE: //
+ _debug("> INVITE (receive)\n");
checkNetwork();
// Set local random port for incoming call
@@ -646,7 +647,7 @@ SipVoIPLink::getEvent (void)
// Generate id
id = Manager::instance().generateNewCallId();
Manager::instance().pushBackNewCall(id, Incoming);
- _debug("New INVITE Event: call with id %d [cid = %d, did = %d]\n",id, event->cid, event->did);
+ _debug(" ID: %d [cid = %d, did = %d]\n",id, event->cid, event->did);
// Display the callerId-name
osip_from_t *from;
@@ -659,6 +660,7 @@ SipVoIPLink::getEvent (void)
osip_from_to_str (event->request->from, &tmp);
if (tmp != NULL) {
snprintf (sipcall->getRemoteUri(), 256, "%s", tmp);
+ _debug(" Remote URI: %s\n", tmp);
osip_free (tmp);
}
}
@@ -671,7 +673,7 @@ SipVoIPLink::getEvent (void)
urlUsername = url->username;
}
Manager::instance().callSetInfo(id, name, urlUsername);
- _debug("New INVITE Event: From: %s\n", name.c_str());
+ _debug(" Name/Username: %s/%s\n", name.c_str(), urlUsername.c_str());
}
//Don't need this display text message now that we send the name
//inside the Manager to the gui
@@ -681,11 +683,12 @@ SipVoIPLink::getEvent (void)
// Associate an audio port with a call
sipcall->setLocalAudioPort(_localPort);
sipcall->setLocalIp(getLocalIpAddress());
- _debug("New INVITE Event: we set the local audio to: %s:%d\n", getLocalIpAddress().c_str(), _localPort);
+ _debug(" Local listening port: %d\n", _localPort);
+ _debug(" Local listening IP: %s\n", getLocalIpAddress().c_str());
sipcall->newIncomingCall(event);
if (Manager::instance().incomingCall(id) < 0) {
- Manager::instance().displayErrorText(id, "New INVITE Event: Incoming call failed");
+ Manager::instance().displayErrorText(id, " Incoming Call Failed");
}
break;
@@ -800,14 +803,14 @@ SipVoIPLink::getEvent (void)
// Handle 4XX errors
switch (event->response->status_code) {
case AUTH_REQUIRED:
- _debug("SIP Server ask required authentification: loging...\n");
+ _debug("SIP Server ask required authentification: logging...\n");
setAuthentication();
eXosip_lock();
eXosip_automatic_action();
eXosip_unlock();
break;
case UNAUTHORIZED:
- _debug("Request is unauthorized. SIP Server ask authentification: loging...\n");
+ _debug("Request is unauthorized. SIP Server ask authentification: logging...\n");
setAuthentication();
break;