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Commit 0266fde6 authored by yanmorin's avatar yanmorin
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Debugging incoming message
It's seems like we don't listen to the good port
parent a319b463
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src/managerimpl.cpp
+ 9
− 8
View file @ 0266fde6
......@@ -679,7 +679,7 @@ ManagerImpl::callCanBeAnswered(CALLID id) {
* SipEvent Thread
*/
int
ManagerImpl::incomingCall (CALLID id)
ManagerImpl::incomingCall (CALLID id, const std::string& name, const std::string& number)
{
ost::MutexLock m(_mutex);
Call* call = getCall(id);
......@@ -699,9 +699,9 @@ ManagerImpl::incomingCall (CALLID id)
// TODO: Account not yet implemented
std::string accountId = "acc1";
std::string from = call->getCallerIdName();
std::string number = call->getCallerIdNumber();
if ( number.length() ) {
std::string from = name; call->setCallerIdName(name);
call->setCallerIdNumber(number);
if ( !number.empty() ) {
from.append(" <");
from.append(number);
from.append(">");
......@@ -873,10 +873,11 @@ void
ManagerImpl::stopTone() {
_toneMutex.enterMutex();
_telephoneTone->setCurrentTone(Tone::TONE_NULL);
// if ( _toneType != ZT_TONE_NULL ) {
// _toneType = ZT_TONE_NULL;
// _tone->stopTone();
// }
// for ringing tone..
if ( _toneType != ZT_TONE_NULL ) {
_toneType = ZT_TONE_NULL;
_tone->stopTone();
}
_toneMutex.leaveMutex();
getAudioDriver()->stopStream();
}
......
src/managerimpl.h
+ 1
− 1
View file @ 0266fde6
......@@ -156,7 +156,7 @@ public:
bool playTone ();
void stopTone();
int incomingCall (CALLID id);
int incomingCall (CALLID id, const std::string& name, const std::string& number);
void peerAnsweredCall (CALLID id);
int peerRingingCall (CALLID id);
int peerHungupCall (CALLID id);
......
src/sipcall.cpp
+ 114
− 166
View file @ 0266fde6
......@@ -46,7 +46,6 @@ SipCall::SipCall (CALLID id, CodecDescriptorVector* cdv) : _localIp("127.0.0.1")
_audiocodec = NULL;
enable_audio = false;
_state = 0;
_local_audio_port = 0;
_remote_sdp_audio_port = 0;
_local_sendrecv = 0; /* _SENDRECV, _SENDONLY, _RECVONLY */
......@@ -158,7 +157,6 @@ SipCall::newIncomingCall (eXosip_event_t *event) {
if (_did < 1 && _cid < 1) {
return -1; /* not enough information for this event?? */
}
osip_strncpy (_textinfo, event->textinfo, 255);
if (event->response != NULL) {
......@@ -167,179 +165,132 @@ SipCall::newIncomingCall (eXosip_event_t *event) {
_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\n", tmp);
_debug(" Remote URI: %s\n", _remote_uri);
osip_free (tmp);
}
}
strcpy(_remote_uri, "");
_name = "";
_number = "";
if (event->request != NULL) {
char *tmp = NULL;
sdp_message_t *remote_sdp = NULL;
/* negotiate payloads */
if (event->request != NULL) {
remote_sdp = eXosip_get_sdp_info (event->request);
}
osip_from_to_str(event->request->from, &tmp);
if (tmp != NULL) {
snprintf (_remote_uri, 255, "%s", tmp);
osip_free (tmp);
if (remote_sdp == NULL) {
_debug("SipCall::newIncomingCall: missing SDP in INVITE request\n");
// Get the name/number
osip_from_t *from;
osip_from_init(&from);
osip_from_parse(from, _remote_uri);
_name = osip_from_get_displayname(from);
osip_uri_t* url = osip_from_get_url(from);
if ( url != NULL ) {
_number = url->username;
}
osip_from_free(from);
}
}
_debug(" Name: %s\n", _name.c_str());
_debug(" Number: %s\n", _number.c_str());
_debug(" Remote URI: %s\n", _remote_uri);
if (remote_sdp != NULL) { /* TODO: else build an offer */
if (remote_sdp == NULL) {
_debug("SipCall::newIncomingCall: No remote SDP body found for call\n");
// Send 400 BAD REQUEST
eXosip_call_send_answer (_tid, 400, NULL);
sdp_message_free (remote_sdp);
return 0;
}
/* negotiate payloads */
sdp_message_t *remote_sdp = NULL;
if (event->request != NULL) {
remote_sdp = eXosip_get_sdp_info (event->request);
}
if (remote_sdp == NULL) {
_debug("SipCall::newIncomingCall: No remote SDP in INVITE request. Sending 400 BAD REQUEST\n");
// Send 400 BAD REQUEST
eXosip_call_send_answer (_tid, 400, NULL);
return 0;
}
/* TODO: else build an offer */
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);
if (remote_med == NULL || remote_med->m_port == NULL) {
/* no audio media proposed */
// Send 415 Unsupported media type
eXosip_call_send_answer (_tid, 415, NULL);
sdp_message_free (remote_sdp);
return 0;
}
// Remote Media IP
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);
}
_remote_sdp_audio_port = atoi (remote_med->m_port);
_debug(" Remote Audio Port: %d\n", _remote_sdp_audio_port);
// Remote Media Port
sdp_media_t *remote_med = eXosip_get_audio_media (remote_sdp);
if (remote_med == NULL || remote_med->m_port == NULL) {
// no audio media proposed
_debug("< Sending 415 Unsupported media type\n");
eXosip_lock();
eXosip_call_send_answer (_tid, 415, NULL);
eXosip_unlock();
sdp_message_free (remote_sdp);
return 0;
}
_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') {
int pos;
pos = 0;
while (!osip_list_eol (remote_med->m_payloads, pos)) {
tmp = (char *) osip_list_get (remote_med->m_payloads, pos);
if (tmp != NULL &&
(0 == osip_strcasecmp (tmp, "0")
|| 0 == osip_strcasecmp (tmp, "8"))) {
break;
}
tmp = NULL;
pos++;
}
}
if (tmp != NULL) {
payload = atoi (tmp);
_debug(" Payload: %d\n", payload);
} else {
// Send 415 Unsupported media type
eXosip_call_send_answer (_tid, 415, NULL);
sdp_message_free (remote_sdp);
return 0;
}
if (tmp != NULL && (payload == 0 || payload == 8)
&& _remote_sdp_audio_port > 0 && _remote_sdp_audio_ip[0] != '\0') {
}
// Remote Payload
char *tmp = NULL;
if (_remote_sdp_audio_port > 0 && _remote_sdp_audio_ip[0] != '\0') {
int pos = 0;
while (!osip_list_eol (remote_med->m_payloads, pos)) {
tmp = (char *) osip_list_get (remote_med->m_payloads, pos);
if (tmp != NULL && ( 0 == osip_strcasecmp (tmp, "0") || 0 == osip_strcasecmp (tmp, "8") )) {
break;
}
tmp = NULL;
pos++;
}
}
if (tmp != NULL) {
payload = atoi (tmp);
_debug(" Payload: %d\n", payload);
setAudioCodec(_cdv->at(0)->alloc(payload, "")); // codec builder for the mic
} else {
_debug("< Sending 415 Unsupported media type\n");
eXosip_lock();
eXosip_call_send_answer (_tid, 415, NULL);
eXosip_unlock();
sdp_message_free (remote_sdp);
return 0;
}
if (remote_sdp != NULL) { /* TODO: else build an offer */
osip_message_t *answer;
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);
if (i != 0) {
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) {
enable_audio = false;
}
if (enable_audio == false) /* audio is started */ {
sdp_message_t *local_sdp;
local_sdp = eXosip_get_sdp_info (answer);
if (remote_sdp != NULL && local_sdp != NULL) {
sdp_connection_t *conn;
sdp_media_t *local_med;
sdp_media_t *remote_med;
char *tmp = NULL;
int audio_port = 0;
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
&& _remote_sdp_audio_ip[0] != '\0'
&& local_med != NULL) {
tmp = (char *) osip_list_get (local_med->m_payloads, 0);
}
if (tmp != NULL) {
payload = atoi (tmp);
_debug(" Remote Payload: %d\n", payload);
setAudioCodec(_cdv->at(0)->alloc(payload, "")); // codec builder for the mic
}
if (tmp != NULL
&& audio_port > 0
&& _remote_sdp_audio_port > 0
&& _remote_sdp_audio_ip[0] != '\0') {
/* search if stream is sendonly or recvonly */
_remote_sendrecv =
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) {
_local_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");
}
osip_message_t *answer = 0;
eXosip_lock();
_debug("< Building Answer 183\n");
if (0 == eXosip_call_build_answer (_tid, 183, &answer)) {
if ( 0 != sdp_complete_message(remote_sdp, answer)) {
osip_message_free(answer);
// Send 415 Unsupported media type
eXosip_call_send_answer (_tid, 415, NULL);
_debug("< Sending Answer 415\n");
} else {
enable_audio = false;
if (i != 0) {
_debug("SipCall::newIncomingCall: cannot send 183 progress?\n");
}
}
eXosip_unlock ();
sdp_message_t *local_sdp = eXosip_get_sdp_info(answer);
sdp_media_t *local_med = NULL;
if (local_sdp != NULL) {
local_med = eXosip_get_audio_media(local_sdp);
}
if (local_sdp != NULL && local_med != NULL) {
/* search if stream is sendonly or recvonly */
_remote_sendrecv = 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) { _local_sendrecv = _RECVONLY; }
else if (_remote_sendrecv == _RECVONLY) { _local_sendrecv = _SENDONLY; }
}
_debug(" Final Local SendRecv: %d\n", _local_sendrecv);
sdp_message_free (local_sdp);
}
_debug(" < Sending answer 183\n");
if (0 != eXosip_call_send_answer (_tid, 183, answer)) {
_debug("SipCall::newIncomingCall: cannot send 183 progress?\n");
}
}
_state = event->type;
}
eXosip_unlock ();
sdp_message_free (remote_sdp);
return 0;
}
......@@ -371,7 +322,6 @@ SipCall::ringingCall (eXosip_event_t *event) {
osip_free (tmp);
}
}
this->_state = event->type;;
return 0;
}
......@@ -380,7 +330,6 @@ SipCall::receivedAck (eXosip_event_t *event)
{
_cid = event->cid;
_did = event->did;
_state = event->type;
return 0;
}
......@@ -442,7 +391,6 @@ SipCall::answeredCall(eXosip_event_t *event) {
}
eXosip_unlock ();
this->_state = event->type;
return 0;
}
......
src/sipcall.h
+ 5
− 1
View file @ 0266fde6
......@@ -110,6 +110,9 @@ public:
inline void setStandBy (bool standby) { _standby = standby; }
inline bool getStandBy (void) { return _standby; }
std::string getName() const { return _name; }
std::string getNumber() const { return _number; }
private:
void alloc (void);
void dealloc (void);
......@@ -137,12 +140,13 @@ private:
char* _remote_uri;
char* _remote_sdp_audio_ip;
int _state;
int _local_audio_port;
int _remote_sdp_audio_port;
int _local_sendrecv; /* _SENDRECV, _SENDONLY, _RECVONLY */
int _remote_sendrecv; /* _SENDRECV, _SENDONLY, _RECVONLY */
std::string _localIp;
std::string _name; // set by incoming call
std::string _number; // set by incoming call
};
#endif // __SIP_CALL_H__
src/sipvoiplink.cpp
+ 2
− 32
View file @ 0266fde6
......@@ -649,45 +649,15 @@ SipVoIPLink::getEvent (void)
Manager::instance().pushBackNewCall(id, Incoming);
_debug(" ID: %d [cid = %d, did = %d]\n",id, event->cid, event->did);
// Display the callerId-name
osip_from_t *from;
osip_from_init(&from);
sipcall = getSipCall(id);
if (event->request != NULL) {
char *tmp = NULL;
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);
}
}
osip_from_parse(from, sipcall->getRemoteUri());
{
std::string name = osip_from_get_displayname(from);
std::string urlUsername("");
osip_uri_t* url = osip_from_get_url(from);
if ( url != NULL ) {
urlUsername = url->username;
}
Manager::instance().callSetInfo(id, name, urlUsername);
_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
//Manager::instance().displayTextMessage(id, name);
osip_from_free(from);
// Associate an audio port with a call
sipcall = getSipCall(id);
sipcall->setLocalAudioPort(_localPort);
sipcall->setLocalIp(getLocalIpAddress());
_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) {
if (Manager::instance().incomingCall(id, sipcall->getName(), sipcall->getNumber()) < 0) {
Manager::instance().displayErrorText(id, " Incoming Call Failed");
}
break;
......
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