From 0266fde66fcb83c186867a667223b417ee8446d3 Mon Sep 17 00:00:00 2001
From: yanmorin <yanmorin>
Date: Thu, 27 Oct 2005 22:47:59 +0000
Subject: [PATCH] Debugging incoming message It's seems like we don't listen to
the good port
---
src/managerimpl.cpp | 17 +--
src/managerimpl.h | 2 +-
src/sipcall.cpp | 280 ++++++++++++++++++--------------------------
src/sipcall.h | 6 +-
src/sipvoiplink.cpp | 34 +-----
5 files changed, 131 insertions(+), 208 deletions(-)
diff --git a/src/managerimpl.cpp b/src/managerimpl.cpp
index 3f2d1d1dc8..369cc73879 100644
--- a/src/managerimpl.cpp
+++ b/src/managerimpl.cpp
@@ -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();
}
diff --git a/src/managerimpl.h b/src/managerimpl.h
index dcfa29a435..dd903e726d 100644
--- a/src/managerimpl.h
+++ b/src/managerimpl.h
@@ -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);
diff --git a/src/sipcall.cpp b/src/sipcall.cpp
index 63fe2303f0..c6eb8e8744 100644
--- a/src/sipcall.cpp
+++ b/src/sipcall.cpp
@@ -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;
}
diff --git a/src/sipcall.h b/src/sipcall.h
index 95626bc963..f8d9904219 100644
--- a/src/sipcall.h
+++ b/src/sipcall.h
@@ -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__
diff --git a/src/sipvoiplink.cpp b/src/sipvoiplink.cpp
index 69311b79de..44a399b7cd 100644
--- a/src/sipvoiplink.cpp
+++ b/src/sipvoiplink.cpp
@@ -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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