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Debugging incoming call

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This commit is contained in:
yanmorin
2005-10-27 21:08:19 +00:00
parent 739560531c
commit a319b4631d
4 changed files with 58 additions and 34 deletions
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3
src/audio/audiortp.cpp
View File

@ -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;
}

24
src/managerimpl.cpp
View File

@ -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;
}

50
src/sipcall.cpp
View File

@ -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) {

15
src/sipvoiplink.cpp
View File

@ -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;