From 2953c83c4ab6040d07a196d31874737958a4e6e3 Mon Sep 17 00:00:00 2001
From: yanmorin <yanmorin>
Date: Wed, 26 Jul 2006 16:18:07 +0000
Subject: [PATCH] *** empty log message ***
---
src/eventthread.cpp | 1 +
src/iaxaccount.cpp | 2 +-
src/iaxcall.cpp | 633 +-------------------------------------------
src/iaxvoiplink.cpp | 107 +++++++-
src/iaxvoiplink.h | 8 +-
src/sipcall.cpp | 633 +++++++++++++++++++++++++++++++++++++++++++-
6 files changed, 744 insertions(+), 640 deletions(-)
diff --git a/src/eventthread.cpp b/src/eventthread.cpp
index dd113bad97..401e2e8948 100644
--- a/src/eventthread.cpp
+++ b/src/eventthread.cpp
@@ -43,3 +43,4 @@ EventThread::run (void)
}
}
+
diff --git a/src/iaxaccount.cpp b/src/iaxaccount.cpp
index d8b49a5914..7ef9838bcd 100644
--- a/src/iaxaccount.cpp
+++ b/src/iaxaccount.cpp
@@ -88,7 +88,7 @@ IAXAccount::initConfig(Conf::ConfigTree& config)
std::string type_str("string");
std::string type_int("int");
- config.addConfigTreeItem(section, Conf::ConfigTreeItem(CONFIG_ACCOUNT_TYPE, "AIX", type_str));
+ config.addConfigTreeItem(section, Conf::ConfigTreeItem(CONFIG_ACCOUNT_TYPE, "IAX", type_str));
config.addConfigTreeItem(section, Conf::ConfigTreeItem(CONFIG_ACCOUNT_ENABLE,"1", type_int));
config.addConfigTreeItem(section, Conf::ConfigTreeItem(CONFIG_ACCOUNT_AUTO_REGISTER, "1", type_int));
}
diff --git a/src/iaxcall.cpp b/src/iaxcall.cpp
index cb3fde0926..bfcd6f8e0f 100644
--- a/src/iaxcall.cpp
+++ b/src/iaxcall.cpp
@@ -1,7 +1,6 @@
/*
- * Copyright (C) 2004-2006 Savoir-Faire Linux inc.
+ * Copyright (C) 2006 Savoir-Faire Linux inc.
* Author: Yan Morin <yan.morin@savoirfairelinux.com>
- * Author : Laurielle Lea <laurielle.lea@savoirfairelinux.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
@@ -18,636 +17,16 @@
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
-#include "sipcall.h"
+#include "iaxcall.h"
#include "global.h" // for _debug
-#include <sstream> // for media buffer
-#define _SENDRECV 0
-#define _SENDONLY 1
-#define _RECVONLY 2
-
-SIPCall::SIPCall(const CallID& id, Call::CallType type) : Call(id, type),
- _localIPAddress(""), _remoteIPAddress("")
-{
- _cid = 0;
- _did = 0;
- _tid = 0;
- _audioCodec = 0;
- _localAudioPort = 0;
- _localExternalAudioPort = 0;
- _remoteAudioPort = 0;
-}
-
-SIPCall::~SIPCall()
+IAXCall::IAXCall(const CallID& id, Call::CallType type) : Call(id, type), _session(0)
{
+
}
-CodecDescriptorMap&
-SIPCall::getCodecMap()
+IAXCall::~IAXCall()
{
- return _codecMap;
+ _session = 0; // just to be sure to don't have unknown pointer, do not delete it!
}
-/**
- * Answer incoming call correclty before telling the user
- * @param event eXosip Event
- */
-bool
-SIPCall::SIPCallInvite(eXosip_event_t *event)
-{
- if (event->cid < 1 && event->did < 1) {
- _debug("SIP Failure: Invalid cid and did\n");
- return false;
- }
-
- if (event->request == NULL) {
- _debug("SIP Failure: No request into the event\n");
- return false;
- }
-
- setCid(event->cid);
- setDid(event->did);
- setTid(event->tid);
-
- setPeerInfoFromRequest(event);
-
- sdp_message_t* remote_sdp = getRemoteSDPFromRequest(event);
- if (remote_sdp == 0) {
- return false;
- }
-
- sdp_media_t* remote_med = getRemoteMedia(event->tid, remote_sdp);
- if (remote_med == 0) {
- sdp_message_free (remote_sdp);
- return false;
- }
-
- if (!setRemoteAudioFromSDP(remote_med, remote_sdp)) {
- _debug("SIP Failure: unable to set IP address and port from SDP\n");
- sdp_message_free (remote_sdp);
- return false;
- }
-
- if (!setAudioCodecFromSDP(remote_med, event->tid)) {
- sdp_message_free (remote_sdp);
- return false;
- }
-
- osip_message_t *answer = 0;
- eXosip_lock();
- _debug("< Building Answer 183\n");
- if (0 == eXosip_call_build_answer (event->tid, 183, &answer)) {
- if ( 0 != sdp_complete_message(remote_sdp, answer)) {
- osip_message_free(answer);
- // Send 415 Unsupported media type
- _debug("< Sending Answer 415 : unsupported media type\n");
- eXosip_call_send_answer (event->tid, 415, NULL);
- } else {
-
- 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 */
- int _remote_sendrecv = sdp_analyse_attribute (remote_sdp, remote_med);
- int _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 (event->tid, 183, answer)) {
- _debug("SipCall::newIncomingCall: cannot send 183 progress?\n");
- }
- }
- }
- eXosip_unlock ();
-
- sdp_message_free (remote_sdp);
- return true;
-}
-
-/**
- * newReinviteCall is called when the IP-Phone user receives a change in the call
- * it's almost an newIncomingCall but we send a 200 OK
- * See: 3.7. Session with re-INVITE (IP Address Change)
- * @param event eXosip Event
- * @return true if ok
- */
-bool
-SIPCall::SIPCallReinvite(eXosip_event_t *event)
-{
- if (event->cid < 1 && event->did < 1) {
- _debug("SIP Failure: Invalid cid and did\n");
- return false;
- }
-
- if (event->request == NULL) {
- _debug("SIP Failure: No request into the event\n");
- return false;
- }
-
- setCid(event->cid);
- setDid(event->did);
- setTid(event->tid);
-
- setPeerInfoFromRequest(event);
-
- sdp_message_t* remote_sdp = getRemoteSDPFromRequest(event);
- if (remote_sdp == 0) {
- return false;
- }
-
- sdp_media_t* remote_med = getRemoteMedia(event->tid, remote_sdp);
- if (remote_med == 0) {
- sdp_message_free (remote_sdp);
- return false;
- }
-
- if (!setRemoteAudioFromSDP(remote_med, remote_sdp)) {
- _debug("SIP Failure: unable to set IP address and port from SDP\n");
- sdp_message_free (remote_sdp);
- return false;
- }
-
- if (!setAudioCodecFromSDP(remote_med, event->tid)) {
- sdp_message_free (remote_sdp);
- return false;
- }
-
- osip_message_t *answer = 0;
- eXosip_lock();
- _debug("< Building Answer 200\n");
- if (0 == eXosip_call_build_answer (event->tid, 200, &answer)) {
- if ( 0 != sdp_complete_message(remote_sdp, answer)) {
- osip_message_free(answer);
- // Send 415 Unsupported media type
- eXosip_call_send_answer (event->tid, 415, NULL);
- _debug("< Sending Answer 415\n");
- } else {
-
- 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 */
- int _remote_sendrecv = sdp_analyse_attribute (remote_sdp, remote_med);
- int _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 200\n");
- if (0 != eXosip_call_send_answer (event->tid, 200, answer)) {
- _debug("SipCall::newIncomingCall: cannot send 200 OK?\n");
- }
- }
- }
- eXosip_unlock ();
- sdp_message_free (remote_sdp);
- return true;
-}
-
- /**
- * Peer answered to a call (on hold or not)
- * @param event eXosip Event
- * @return true if ok
- */
-bool
-SIPCall::SIPCallAnswered(eXosip_event_t *event)
-{
- if (event->cid < 1 && event->did < 1) {
- _debug("SIP Failure: Invalid cid and did\n");
- return false;
- }
-
- if (event->request == NULL) {
- _debug("SIP Failure: No request into the event\n");
- return false;
- }
-
- setCid(event->cid);
- setDid(event->did);
-
- //setPeerInfoFromResponse()
-
- eXosip_lock ();
- {
- osip_message_t *ack = NULL;
- int i;
- i = eXosip_call_build_ack (event->did, &ack);
- if (i != 0) {
- _debug("SipCall::answeredCall: Cannot build ACK for call!\n");
- } else {
- sdp_message_t *local_sdp = NULL;
- sdp_message_t *remote_sdp = NULL;
-
- if (event->request != NULL && event->response != NULL) {
- local_sdp = eXosip_get_sdp_info (event->request);
- remote_sdp = eXosip_get_sdp_info (event->response);
- }
- if (local_sdp == NULL && remote_sdp != NULL) {
- /* sdp in ACK */
- i = sdp_complete_message (remote_sdp, ack);
- if (i != 0) {
- _debug("SipCall::answeredCall: Cannot complete ACK with sdp body?!\n");
- }
- }
- sdp_message_free (local_sdp);
- sdp_message_free (remote_sdp);
-
- _debug("< Send ACK\n");
- eXosip_call_send_ack (event->did, ack);
- }
- }
- eXosip_unlock ();
- return true;
-}
- /**
- * We retreive final SDP info if they changed
- * @param event eXosip Event
- * @return true if ok (change / no change) or false on error
- */
-bool
-SIPCall::SIPCallAnsweredWithoutHold(eXosip_event_t *event)
-{
- if (event->response == NULL || event->request == NULL) { return false; }
-
- eXosip_lock();
- sdp_message_t *remote_sdp = eXosip_get_sdp_info (event->response);
- eXosip_unlock();
- if (remote_sdp == NULL) {
- _debug("SIP Failure: no remote sdp\n");
- sdp_message_free(remote_sdp);
- return false;
- }
-
- sdp_media_t *remote_med = getRemoteMedia(event->tid, remote_sdp);
- if (remote_med==NULL) {
- sdp_message_free(remote_sdp);
- return false;
- }
- if ( ! setRemoteAudioFromSDP(remote_med, remote_sdp) ) {
- sdp_message_free(remote_sdp);
- return false;
- }
-
- char *tmp = (char *) osip_list_get (remote_med->m_payloads, 0);
-
- setAudioCodec(0);
- if (tmp != NULL) {
- int payload = atoi (tmp);
- _debug(" Remote Payload: %d\n", payload);
- setAudioCodec(_codecMap.getCodec((CodecType)payload)); // codec builder for the mic
- }
-
-/*
- // 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);
- if (_local_sendrecv == _SENDRECV) {
- if (_remote_sendrecv == _SENDONLY)
- _local_sendrecv = _RECVONLY;
- else if (_remote_sendrecv == _RECVONLY)
- _local_sendrecv = _SENDONLY;
- }
- _debug(" Remote Sendrecv: %d\n", _remote_sendrecv);
- _debug(" Local Sendrecv: %d\n", _local_sendrecv);
-*/
- sdp_message_free (remote_sdp);
- return true;
-}
-
-const std::string&
-SIPCall::getLocalIp()
-{
- ost::MutexLock m(_callMutex);
- return _localIPAddress;
-}
-
-unsigned int
-SIPCall::getLocalAudioPort()
-{
- ost::MutexLock m(_callMutex);
- return _localAudioPort;
-}
-
-unsigned int
-SIPCall::getRemoteAudioPort()
-{
- ost::MutexLock m(_callMutex);
- return _remoteAudioPort;
-}
-
-const std::string&
-SIPCall::getRemoteIp()
-{
- ost::MutexLock m(_callMutex);
- return _remoteIPAddress;
-}
-
-AudioCodec*
-SIPCall::getAudioCodec()
-{
- ost::MutexLock m(_callMutex);
- return _audioCodec;
-}
-
-void
-SIPCall::setAudioStart(bool start)
-{
- ost::MutexLock m(_callMutex);
- _audioStarted = start;
-}
-
-bool
-SIPCall::isAudioStarted()
-{
- ost::MutexLock m(_callMutex);
- return _audioStarted;
-}
-
- //TODO: humm?
-int
-SIPCall::sdp_complete_message(sdp_message_t * remote_sdp, osip_message_t * msg)
-{
- // Format port to a char*
- if (remote_sdp == NULL) {
- _debug("SipCall::sdp_complete_message: No remote SDP body found for call\n");
- return -1;
- }
- if (msg == NULL) {
- _debug("SipCall::sdp_complete_message: No message to complete\n");
- return -1;
- }
-
- std::ostringstream media;
-
- // for each medias
- int iMedia = 0;
- char *tmp = NULL;
- while (!osip_list_eol(remote_sdp->m_medias, iMedia)) {
- sdp_media_t *remote_med = (sdp_media_t *)osip_list_get(remote_sdp->m_medias, iMedia);
- if (remote_med == 0) { continue; }
-
- if (0 != osip_strcasecmp (remote_med->m_media, "audio")) {
- // if this is not an "audio" media, we set it to 0
- media << "m=" << remote_med->m_media << " 0 " << remote_med->m_proto << " \r\n";
- } else {
- std::ostringstream listCodec;
- std::ostringstream listRtpMap;
-
- // search for compatible codec: foreach payload
- int iPayload = 0;
- while (!osip_list_eol(remote_med->m_payloads, iPayload)) {
- tmp = (char *)osip_list_get(remote_med->m_payloads, iPayload);
- if (tmp!=NULL) {
- int payload = atoi(tmp);
- AudioCodec* audiocodec = _codecMap.getCodec((CodecType)payload);
- if (audiocodec!=0 && audiocodec->isActive()) {
- listCodec << payload << " ";
- listRtpMap << "a=rtpmap: " << payload << " " << audiocodec->getCodecName() << "/" << audiocodec->getClockRate();
- if ( audiocodec->getChannel() != 1) {
- listRtpMap << "/" << audiocodec->getChannel();
- }
- listRtpMap << "\r\n";
- }
- }
- iPayload++;
- }
- if (listCodec.str().empty()) {
- media << "m=" << remote_med->m_media << " 0 " << remote_med->m_proto << " \r\n";
- } else {
- // we add the media line + a=rtpmap list
- media << "m=" << remote_med->m_media << " " << getLocalExternAudioPort() << " RTP/AVP " << listCodec.str() << "\r\n";
- media << listRtpMap.str();
- }
- }
- iMedia++;
- }
- char buf[4096];
- snprintf (buf, 4096,
- "v=0\r\n"
- "o=user 0 0 IN IP4 %s\r\n"
- "s=session\r\n"
- "c=IN IP4 %s\r\n"
- "t=0 0\r\n"
- "%s\n", getLocalIp().c_str(), getLocalIp().c_str(), media.str().c_str());
-
- osip_message_set_body (msg, buf, strlen (buf));
- osip_message_set_content_type (msg, "application/sdp");
- _debug(" sdp: %s", buf);
- return 0;
-}
-
-
- // TODO: hum???
-int
-SIPCall::sdp_analyse_attribute (sdp_message_t * sdp, sdp_media_t * med)
-{
- int pos;
- int pos_media;
-
- /* test media attributes */
- pos = 0;
- while (!osip_list_eol (med->a_attributes, pos)) {
- sdp_attribute_t *at;
-
- at = (sdp_attribute_t *) osip_list_get (med->a_attributes, pos);
- if (at->a_att_field != NULL &&
- 0 == strcmp (at->a_att_field, "sendonly")) {
- return _SENDONLY;
- } else if (at->a_att_field != NULL &&
- 0 == strcmp (at->a_att_field, "recvonly")) {
- return _RECVONLY;
- } else if (at->a_att_field != NULL &&
- 0 == strcmp (at->a_att_field, "sendrecv")) {
- return _SENDRECV;
- }
- pos++;
- }
-
- /* test global attributes */
- pos_media = -1;
- pos = 0;
- while (!osip_list_eol (sdp->a_attributes, pos)) {
- sdp_attribute_t *at;
-
- at = (sdp_attribute_t *) osip_list_get (sdp->a_attributes, pos);
- if (at->a_att_field != NULL &&
- 0 == strcmp (at->a_att_field, "sendonly")) {
- return _SENDONLY;
- } else if (at->a_att_field != NULL &&
- 0 == strcmp (at->a_att_field, "recvonly")) {
- return _RECVONLY;
- } else if (at->a_att_field != NULL &&
- 0 == strcmp (at->a_att_field, "sendrecv")) {
- return _SENDRECV;
- }
- pos++;
- }
-
- return _SENDRECV;
-}
-
-bool
-SIPCall::setPeerInfoFromRequest(eXosip_event_t *event)
-{
- // event->request should not be NULL!
- char remote_uri[256] = "";
- std::string name("");
- std::string number("");
-
- char *tmp = NULL;
- osip_from_to_str(event->request->from, &tmp);
- if (tmp != NULL) {
- snprintf (remote_uri, 255, "%s", tmp);
- remote_uri[255] = '\0';
- osip_free (tmp);
-
- // Get the name/number
- osip_from_t *from;
- osip_from_init(&from);
- osip_from_parse(from, remote_uri);
- char *tmpname = osip_from_get_displayname(from);
- if ( tmpname != NULL ) {
- name = tmpname;
- }
- osip_uri_t* url = osip_from_get_url(from);
- if ( url != NULL && url->username != 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);
-
- setPeerName(name);
- setPeerNumber(number);
- return true;
-}
-
-sdp_message_t*
-SIPCall::getRemoteSDPFromRequest(eXosip_event_t *event)
-{
- // event->request should not be null!
- /* negotiate payloads */
- sdp_message_t *remote_sdp = NULL;
- if (event->request != NULL) {
- eXosip_lock();
- remote_sdp = eXosip_get_sdp_info (event->request);
- eXosip_unlock();
- }
- if (remote_sdp == NULL) {
- _debug("SIP Failure: No SDP into the request\n");
- _debug("< Sending 400 Bad Request (no SDP)\n");
- eXosip_lock();
- eXosip_call_send_answer (event->tid, 400, NULL);
- eXosip_unlock();
- return 0;
- }
- return remote_sdp;
-}
-
-sdp_media_t*
-SIPCall::getRemoteMedia(int tid, sdp_message_t* remote_sdp)
-{
- // Remote Media Port
- eXosip_lock();
- sdp_media_t *remote_med = eXosip_get_audio_media(remote_sdp);
- eXosip_unlock();
-
- if (remote_med == NULL || remote_med->m_port == NULL) {
- // no audio media proposed
- _debug("SIP Failure: unsupported media\n");
- _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;
- }
- return remote_med;
-}
-
-bool
-SIPCall::setRemoteAudioFromSDP(sdp_media_t* remote_med, sdp_message_t* remote_sdp)
-{
- // Remote Media IP
- eXosip_lock();
- sdp_connection_t *conn = eXosip_get_audio_connection(remote_sdp);
- eXosip_unlock();
- if (conn != NULL && conn->c_addr != NULL) {
- char _remote_sdp_audio_ip[50] = "";
- snprintf (_remote_sdp_audio_ip, 49, "%s", conn->c_addr);
- _remote_sdp_audio_ip[49] = '\0';
- _debug(" Remote Audio IP: %s\n", _remote_sdp_audio_ip);
- setRemoteIP(_remote_sdp_audio_ip);
- if (_remote_sdp_audio_ip[0] == '\0') {
- setRemoteAudioPort(0);
- return false;
- }
- }
-
- // Remote port
- int _remote_sdp_audio_port = atoi(remote_med->m_port);
- _debug(" Remote Audio Port: %d\n", _remote_sdp_audio_port);
- setRemoteAudioPort(_remote_sdp_audio_port);
-
- if (_remote_sdp_audio_port == 0) {
- return false;
- }
- return true;
-}
-
-bool
-SIPCall::setAudioCodecFromSDP(sdp_media_t* remote_med, int tid)
-{
- // Remote Payload
- char *tmp = NULL;
- 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 ) {
- int payload = atoi(tmp);
- // stop if we find a correct codec
- if (0 != _codecMap.getCodec((CodecType)payload)){
- break;
- }
- }
- tmp = NULL;
- pos++;
- }
-
- setAudioCodec(0);
- if (tmp != NULL) {
- int payload = atoi (tmp);
- _debug(" Payload: %d\n", payload);
- setAudioCodec(_codecMap.getCodec((CodecType)payload)); // codec builder for the mic
- }
- if (getAudioCodec() == 0) {
- _debug("SIPCall Failure: Unable to set codec\n");
- _debug("< Sending 415 Unsupported media type\n");
- eXosip_lock();
- eXosip_call_send_answer(tid, 415, NULL);
- eXosip_unlock();
- return false;
- }
- return true;
-}
diff --git a/src/iaxvoiplink.cpp b/src/iaxvoiplink.cpp
index b02732d3cc..3a5a0d8cd8 100644
--- a/src/iaxvoiplink.cpp
+++ b/src/iaxvoiplink.cpp
@@ -28,12 +28,14 @@ IAXVoIPLink::IAXVoIPLink(const AccountID& accountID)
: VoIPLink(accountID)
{
_evThread = new EventThread(this);
+ _regSession = 0;
}
IAXVoIPLink::~IAXVoIPLink()
{
delete _evThread; _evThread = 0;
+ _regSession = 0; // shall not delete it
terminate();
}
@@ -53,6 +55,8 @@ IAXVoIPLink::init()
_debug("IAX Info: listening on port %d\n", port);
_localPort = port;
returnValue = true;
+
+ _evThread->start();
}
return returnValue;
}
@@ -75,17 +79,58 @@ IAXVoIPLink::getEvent()
call = iaxFindCallBySession(event->session);
if (call!=0) {
iaxHandleCallEvent(event, call);
+ } else if (event->session != 0 && event->session == _regSession) {
+ // in iaxclient, there is many session handling, here, only one
+ iaxHandleRegReply(event);
+ } else {
+ switch(e->etype) {
+ case IAX_EVENT_REGACK:
+ case IAX_EVENT_REGREJ:
+ _debug("Unknown IAX Registration Event\n");
+ break;
+
+ case IAX_EVENT_REGREQ:
+ _debug("Registration by a peer, don't allow it\n");
+ break;
+ case IAX_EVENT_CONNECT: // new call
+ // New incoming call!
+ break;
+
+ case IAX_EVENT_TIMEOUT: // timeout for an unknown session
+
+ break;
+
+ default:
+ _debug("Unknown event type: %d\n", event->type);
+ }
}
- return;
+ iax_event_free(event);
}
- iax_event_free(event);
- // thread wait 5 second
- // unlock mutex
+ // unlock mutex here
+
+ //iaxRefreshRegistrations();
+
+ // thread wait 5 millisecond
+ _evThread->sleep(5);
}
bool
IAXVoIPLink::setRegister()
{
+ if (_regSession==0) {
+ // lock
+
+ _regSession = iax_session_new();
+
+ if (!_regSession) {
+ _debug("error when generating new session for register");
+ } else {
+ // refresh
+ // last reg
+ }
+
+ // unlock
+ }
return false;
}
@@ -116,5 +161,57 @@ IAXVoIPLink::iaxFindCallBySession(struct iax_session* session)
void
IAXVoIPLink::iaxHandleCallEvent(iax_event* event, IAXCall* call)
{
-
+ // call should not be 0
+ // note activity?
+ //
+ switch(event->type) {
+ case IAX_EVENT_HANGUP:
+ break;
+
+ case IAX_EVENT_REJECT:
+ break;
+
+ case IAX_EVENT_ACCEPT:
+ break;
+
+ case IAX_EVENT_ANSWER:
+ break;
+
+ case IAX_EVENT_BUSY:
+ break;
+
+ case IAX_EVENT_VOICE:
+ break;
+
+ case IAX_EVENT_TEXT:
+ break;
+
+ case IAX_EVENT_RINGA:
+ break;
+
+ case IAX_EVENT_PONG:
+ break;
+
+ case IAX_EVENT_URL:
+ break;
+
+ case IAX_EVENT_CNG:
+ break;
+
+ case IAX_EVENT_TIMEOUT:
+ break;
+
+ case IAX_EVENT_TRANSFER:
+ break;
+
+ default:
+ _debug("Unknown event type: %d\n", event->type);
+
+ }
+}
+
+void
+IAXVoIPLink::iaxHandleRegReply(iax_event* event)
+{
+ // use _regSession here, should be equal to event->session;
}
diff --git a/src/iaxvoiplink.h b/src/iaxvoiplink.h
index c0329a7401..47d0d3e227 100644
--- a/src/iaxvoiplink.h
+++ b/src/iaxvoiplink.h
@@ -63,7 +63,7 @@ private:
* @param session an iax_session valid pointer
* @return iaxcall or 0 if not found
*/
- IAXCall* iaxFindCallBySession(struct iax_session *session);
+ IAXCall* iaxFindCallBySession(struct iax_session* session);
/**
* Handle IAX Event for a call
@@ -72,8 +72,14 @@ private:
*/
void iaxHandleCallEvent(iax_event* event, IAXCall* call);
+ /**
+ * Handle IAX Registration Reply event
+ * @param event An iax_event pointer
+ */
+ void iaxHandleRegReply(iax_event* event);
EventThread* _evThread;
+ struct iax_session* _regSession;
};
#endif
diff --git a/src/sipcall.cpp b/src/sipcall.cpp
index bfcd6f8e0f..cb3fde0926 100644
--- a/src/sipcall.cpp
+++ b/src/sipcall.cpp
@@ -1,6 +1,7 @@
/*
- * Copyright (C) 2006 Savoir-Faire Linux inc.
+ * Copyright (C) 2004-2006 Savoir-Faire Linux inc.
* Author: Yan Morin <yan.morin@savoirfairelinux.com>
+ * Author : Laurielle Lea <laurielle.lea@savoirfairelinux.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
@@ -17,16 +18,636 @@
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
-#include "iaxcall.h"
+#include "sipcall.h"
#include "global.h" // for _debug
+#include <sstream> // for media buffer
-IAXCall::IAXCall(const CallID& id, Call::CallType type) : Call(id, type), _session(0)
+#define _SENDRECV 0
+#define _SENDONLY 1
+#define _RECVONLY 2
+
+SIPCall::SIPCall(const CallID& id, Call::CallType type) : Call(id, type),
+ _localIPAddress(""), _remoteIPAddress("")
+{
+ _cid = 0;
+ _did = 0;
+ _tid = 0;
+ _audioCodec = 0;
+ _localAudioPort = 0;
+ _localExternalAudioPort = 0;
+ _remoteAudioPort = 0;
+}
+
+SIPCall::~SIPCall()
{
-
}
-IAXCall::~IAXCall()
+CodecDescriptorMap&
+SIPCall::getCodecMap()
{
- _session = 0; // just to be sure to don't have unknown pointer, do not delete it!
+ return _codecMap;
}
+/**
+ * Answer incoming call correclty before telling the user
+ * @param event eXosip Event
+ */
+bool
+SIPCall::SIPCallInvite(eXosip_event_t *event)
+{
+ if (event->cid < 1 && event->did < 1) {
+ _debug("SIP Failure: Invalid cid and did\n");
+ return false;
+ }
+
+ if (event->request == NULL) {
+ _debug("SIP Failure: No request into the event\n");
+ return false;
+ }
+
+ setCid(event->cid);
+ setDid(event->did);
+ setTid(event->tid);
+
+ setPeerInfoFromRequest(event);
+
+ sdp_message_t* remote_sdp = getRemoteSDPFromRequest(event);
+ if (remote_sdp == 0) {
+ return false;
+ }
+
+ sdp_media_t* remote_med = getRemoteMedia(event->tid, remote_sdp);
+ if (remote_med == 0) {
+ sdp_message_free (remote_sdp);
+ return false;
+ }
+
+ if (!setRemoteAudioFromSDP(remote_med, remote_sdp)) {
+ _debug("SIP Failure: unable to set IP address and port from SDP\n");
+ sdp_message_free (remote_sdp);
+ return false;
+ }
+
+ if (!setAudioCodecFromSDP(remote_med, event->tid)) {
+ sdp_message_free (remote_sdp);
+ return false;
+ }
+
+ osip_message_t *answer = 0;
+ eXosip_lock();
+ _debug("< Building Answer 183\n");
+ if (0 == eXosip_call_build_answer (event->tid, 183, &answer)) {
+ if ( 0 != sdp_complete_message(remote_sdp, answer)) {
+ osip_message_free(answer);
+ // Send 415 Unsupported media type
+ _debug("< Sending Answer 415 : unsupported media type\n");
+ eXosip_call_send_answer (event->tid, 415, NULL);
+ } else {
+
+ 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 */
+ int _remote_sendrecv = sdp_analyse_attribute (remote_sdp, remote_med);
+ int _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 (event->tid, 183, answer)) {
+ _debug("SipCall::newIncomingCall: cannot send 183 progress?\n");
+ }
+ }
+ }
+ eXosip_unlock ();
+
+ sdp_message_free (remote_sdp);
+ return true;
+}
+
+/**
+ * newReinviteCall is called when the IP-Phone user receives a change in the call
+ * it's almost an newIncomingCall but we send a 200 OK
+ * See: 3.7. Session with re-INVITE (IP Address Change)
+ * @param event eXosip Event
+ * @return true if ok
+ */
+bool
+SIPCall::SIPCallReinvite(eXosip_event_t *event)
+{
+ if (event->cid < 1 && event->did < 1) {
+ _debug("SIP Failure: Invalid cid and did\n");
+ return false;
+ }
+
+ if (event->request == NULL) {
+ _debug("SIP Failure: No request into the event\n");
+ return false;
+ }
+
+ setCid(event->cid);
+ setDid(event->did);
+ setTid(event->tid);
+
+ setPeerInfoFromRequest(event);
+
+ sdp_message_t* remote_sdp = getRemoteSDPFromRequest(event);
+ if (remote_sdp == 0) {
+ return false;
+ }
+
+ sdp_media_t* remote_med = getRemoteMedia(event->tid, remote_sdp);
+ if (remote_med == 0) {
+ sdp_message_free (remote_sdp);
+ return false;
+ }
+
+ if (!setRemoteAudioFromSDP(remote_med, remote_sdp)) {
+ _debug("SIP Failure: unable to set IP address and port from SDP\n");
+ sdp_message_free (remote_sdp);
+ return false;
+ }
+
+ if (!setAudioCodecFromSDP(remote_med, event->tid)) {
+ sdp_message_free (remote_sdp);
+ return false;
+ }
+
+ osip_message_t *answer = 0;
+ eXosip_lock();
+ _debug("< Building Answer 200\n");
+ if (0 == eXosip_call_build_answer (event->tid, 200, &answer)) {
+ if ( 0 != sdp_complete_message(remote_sdp, answer)) {
+ osip_message_free(answer);
+ // Send 415 Unsupported media type
+ eXosip_call_send_answer (event->tid, 415, NULL);
+ _debug("< Sending Answer 415\n");
+ } else {
+
+ 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 */
+ int _remote_sendrecv = sdp_analyse_attribute (remote_sdp, remote_med);
+ int _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 200\n");
+ if (0 != eXosip_call_send_answer (event->tid, 200, answer)) {
+ _debug("SipCall::newIncomingCall: cannot send 200 OK?\n");
+ }
+ }
+ }
+ eXosip_unlock ();
+ sdp_message_free (remote_sdp);
+ return true;
+}
+
+ /**
+ * Peer answered to a call (on hold or not)
+ * @param event eXosip Event
+ * @return true if ok
+ */
+bool
+SIPCall::SIPCallAnswered(eXosip_event_t *event)
+{
+ if (event->cid < 1 && event->did < 1) {
+ _debug("SIP Failure: Invalid cid and did\n");
+ return false;
+ }
+
+ if (event->request == NULL) {
+ _debug("SIP Failure: No request into the event\n");
+ return false;
+ }
+
+ setCid(event->cid);
+ setDid(event->did);
+
+ //setPeerInfoFromResponse()
+
+ eXosip_lock ();
+ {
+ osip_message_t *ack = NULL;
+ int i;
+ i = eXosip_call_build_ack (event->did, &ack);
+ if (i != 0) {
+ _debug("SipCall::answeredCall: Cannot build ACK for call!\n");
+ } else {
+ sdp_message_t *local_sdp = NULL;
+ sdp_message_t *remote_sdp = NULL;
+
+ if (event->request != NULL && event->response != NULL) {
+ local_sdp = eXosip_get_sdp_info (event->request);
+ remote_sdp = eXosip_get_sdp_info (event->response);
+ }
+ if (local_sdp == NULL && remote_sdp != NULL) {
+ /* sdp in ACK */
+ i = sdp_complete_message (remote_sdp, ack);
+ if (i != 0) {
+ _debug("SipCall::answeredCall: Cannot complete ACK with sdp body?!\n");
+ }
+ }
+ sdp_message_free (local_sdp);
+ sdp_message_free (remote_sdp);
+
+ _debug("< Send ACK\n");
+ eXosip_call_send_ack (event->did, ack);
+ }
+ }
+ eXosip_unlock ();
+ return true;
+}
+ /**
+ * We retreive final SDP info if they changed
+ * @param event eXosip Event
+ * @return true if ok (change / no change) or false on error
+ */
+bool
+SIPCall::SIPCallAnsweredWithoutHold(eXosip_event_t *event)
+{
+ if (event->response == NULL || event->request == NULL) { return false; }
+
+ eXosip_lock();
+ sdp_message_t *remote_sdp = eXosip_get_sdp_info (event->response);
+ eXosip_unlock();
+ if (remote_sdp == NULL) {
+ _debug("SIP Failure: no remote sdp\n");
+ sdp_message_free(remote_sdp);
+ return false;
+ }
+
+ sdp_media_t *remote_med = getRemoteMedia(event->tid, remote_sdp);
+ if (remote_med==NULL) {
+ sdp_message_free(remote_sdp);
+ return false;
+ }
+ if ( ! setRemoteAudioFromSDP(remote_med, remote_sdp) ) {
+ sdp_message_free(remote_sdp);
+ return false;
+ }
+
+ char *tmp = (char *) osip_list_get (remote_med->m_payloads, 0);
+
+ setAudioCodec(0);
+ if (tmp != NULL) {
+ int payload = atoi (tmp);
+ _debug(" Remote Payload: %d\n", payload);
+ setAudioCodec(_codecMap.getCodec((CodecType)payload)); // codec builder for the mic
+ }
+
+/*
+ // 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);
+ if (_local_sendrecv == _SENDRECV) {
+ if (_remote_sendrecv == _SENDONLY)
+ _local_sendrecv = _RECVONLY;
+ else if (_remote_sendrecv == _RECVONLY)
+ _local_sendrecv = _SENDONLY;
+ }
+ _debug(" Remote Sendrecv: %d\n", _remote_sendrecv);
+ _debug(" Local Sendrecv: %d\n", _local_sendrecv);
+*/
+ sdp_message_free (remote_sdp);
+ return true;
+}
+
+const std::string&
+SIPCall::getLocalIp()
+{
+ ost::MutexLock m(_callMutex);
+ return _localIPAddress;
+}
+
+unsigned int
+SIPCall::getLocalAudioPort()
+{
+ ost::MutexLock m(_callMutex);
+ return _localAudioPort;
+}
+
+unsigned int
+SIPCall::getRemoteAudioPort()
+{
+ ost::MutexLock m(_callMutex);
+ return _remoteAudioPort;
+}
+
+const std::string&
+SIPCall::getRemoteIp()
+{
+ ost::MutexLock m(_callMutex);
+ return _remoteIPAddress;
+}
+
+AudioCodec*
+SIPCall::getAudioCodec()
+{
+ ost::MutexLock m(_callMutex);
+ return _audioCodec;
+}
+
+void
+SIPCall::setAudioStart(bool start)
+{
+ ost::MutexLock m(_callMutex);
+ _audioStarted = start;
+}
+
+bool
+SIPCall::isAudioStarted()
+{
+ ost::MutexLock m(_callMutex);
+ return _audioStarted;
+}
+
+ //TODO: humm?
+int
+SIPCall::sdp_complete_message(sdp_message_t * remote_sdp, osip_message_t * msg)
+{
+ // Format port to a char*
+ if (remote_sdp == NULL) {
+ _debug("SipCall::sdp_complete_message: No remote SDP body found for call\n");
+ return -1;
+ }
+ if (msg == NULL) {
+ _debug("SipCall::sdp_complete_message: No message to complete\n");
+ return -1;
+ }
+
+ std::ostringstream media;
+
+ // for each medias
+ int iMedia = 0;
+ char *tmp = NULL;
+ while (!osip_list_eol(remote_sdp->m_medias, iMedia)) {
+ sdp_media_t *remote_med = (sdp_media_t *)osip_list_get(remote_sdp->m_medias, iMedia);
+ if (remote_med == 0) { continue; }
+
+ if (0 != osip_strcasecmp (remote_med->m_media, "audio")) {
+ // if this is not an "audio" media, we set it to 0
+ media << "m=" << remote_med->m_media << " 0 " << remote_med->m_proto << " \r\n";
+ } else {
+ std::ostringstream listCodec;
+ std::ostringstream listRtpMap;
+
+ // search for compatible codec: foreach payload
+ int iPayload = 0;
+ while (!osip_list_eol(remote_med->m_payloads, iPayload)) {
+ tmp = (char *)osip_list_get(remote_med->m_payloads, iPayload);
+ if (tmp!=NULL) {
+ int payload = atoi(tmp);
+ AudioCodec* audiocodec = _codecMap.getCodec((CodecType)payload);
+ if (audiocodec!=0 && audiocodec->isActive()) {
+ listCodec << payload << " ";
+ listRtpMap << "a=rtpmap: " << payload << " " << audiocodec->getCodecName() << "/" << audiocodec->getClockRate();
+ if ( audiocodec->getChannel() != 1) {
+ listRtpMap << "/" << audiocodec->getChannel();
+ }
+ listRtpMap << "\r\n";
+ }
+ }
+ iPayload++;
+ }
+ if (listCodec.str().empty()) {
+ media << "m=" << remote_med->m_media << " 0 " << remote_med->m_proto << " \r\n";
+ } else {
+ // we add the media line + a=rtpmap list
+ media << "m=" << remote_med->m_media << " " << getLocalExternAudioPort() << " RTP/AVP " << listCodec.str() << "\r\n";
+ media << listRtpMap.str();
+ }
+ }
+ iMedia++;
+ }
+ char buf[4096];
+ snprintf (buf, 4096,
+ "v=0\r\n"
+ "o=user 0 0 IN IP4 %s\r\n"
+ "s=session\r\n"
+ "c=IN IP4 %s\r\n"
+ "t=0 0\r\n"
+ "%s\n", getLocalIp().c_str(), getLocalIp().c_str(), media.str().c_str());
+
+ osip_message_set_body (msg, buf, strlen (buf));
+ osip_message_set_content_type (msg, "application/sdp");
+ _debug(" sdp: %s", buf);
+ return 0;
+}
+
+
+ // TODO: hum???
+int
+SIPCall::sdp_analyse_attribute (sdp_message_t * sdp, sdp_media_t * med)
+{
+ int pos;
+ int pos_media;
+
+ /* test media attributes */
+ pos = 0;
+ while (!osip_list_eol (med->a_attributes, pos)) {
+ sdp_attribute_t *at;
+
+ at = (sdp_attribute_t *) osip_list_get (med->a_attributes, pos);
+ if (at->a_att_field != NULL &&
+ 0 == strcmp (at->a_att_field, "sendonly")) {
+ return _SENDONLY;
+ } else if (at->a_att_field != NULL &&
+ 0 == strcmp (at->a_att_field, "recvonly")) {
+ return _RECVONLY;
+ } else if (at->a_att_field != NULL &&
+ 0 == strcmp (at->a_att_field, "sendrecv")) {
+ return _SENDRECV;
+ }
+ pos++;
+ }
+
+ /* test global attributes */
+ pos_media = -1;
+ pos = 0;
+ while (!osip_list_eol (sdp->a_attributes, pos)) {
+ sdp_attribute_t *at;
+
+ at = (sdp_attribute_t *) osip_list_get (sdp->a_attributes, pos);
+ if (at->a_att_field != NULL &&
+ 0 == strcmp (at->a_att_field, "sendonly")) {
+ return _SENDONLY;
+ } else if (at->a_att_field != NULL &&
+ 0 == strcmp (at->a_att_field, "recvonly")) {
+ return _RECVONLY;
+ } else if (at->a_att_field != NULL &&
+ 0 == strcmp (at->a_att_field, "sendrecv")) {
+ return _SENDRECV;
+ }
+ pos++;
+ }
+
+ return _SENDRECV;
+}
+
+bool
+SIPCall::setPeerInfoFromRequest(eXosip_event_t *event)
+{
+ // event->request should not be NULL!
+ char remote_uri[256] = "";
+ std::string name("");
+ std::string number("");
+
+ char *tmp = NULL;
+ osip_from_to_str(event->request->from, &tmp);
+ if (tmp != NULL) {
+ snprintf (remote_uri, 255, "%s", tmp);
+ remote_uri[255] = '\0';
+ osip_free (tmp);
+
+ // Get the name/number
+ osip_from_t *from;
+ osip_from_init(&from);
+ osip_from_parse(from, remote_uri);
+ char *tmpname = osip_from_get_displayname(from);
+ if ( tmpname != NULL ) {
+ name = tmpname;
+ }
+ osip_uri_t* url = osip_from_get_url(from);
+ if ( url != NULL && url->username != 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);
+
+ setPeerName(name);
+ setPeerNumber(number);
+ return true;
+}
+
+sdp_message_t*
+SIPCall::getRemoteSDPFromRequest(eXosip_event_t *event)
+{
+ // event->request should not be null!
+ /* negotiate payloads */
+ sdp_message_t *remote_sdp = NULL;
+ if (event->request != NULL) {
+ eXosip_lock();
+ remote_sdp = eXosip_get_sdp_info (event->request);
+ eXosip_unlock();
+ }
+ if (remote_sdp == NULL) {
+ _debug("SIP Failure: No SDP into the request\n");
+ _debug("< Sending 400 Bad Request (no SDP)\n");
+ eXosip_lock();
+ eXosip_call_send_answer (event->tid, 400, NULL);
+ eXosip_unlock();
+ return 0;
+ }
+ return remote_sdp;
+}
+
+sdp_media_t*
+SIPCall::getRemoteMedia(int tid, sdp_message_t* remote_sdp)
+{
+ // Remote Media Port
+ eXosip_lock();
+ sdp_media_t *remote_med = eXosip_get_audio_media(remote_sdp);
+ eXosip_unlock();
+
+ if (remote_med == NULL || remote_med->m_port == NULL) {
+ // no audio media proposed
+ _debug("SIP Failure: unsupported media\n");
+ _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;
+ }
+ return remote_med;
+}
+
+bool
+SIPCall::setRemoteAudioFromSDP(sdp_media_t* remote_med, sdp_message_t* remote_sdp)
+{
+ // Remote Media IP
+ eXosip_lock();
+ sdp_connection_t *conn = eXosip_get_audio_connection(remote_sdp);
+ eXosip_unlock();
+ if (conn != NULL && conn->c_addr != NULL) {
+ char _remote_sdp_audio_ip[50] = "";
+ snprintf (_remote_sdp_audio_ip, 49, "%s", conn->c_addr);
+ _remote_sdp_audio_ip[49] = '\0';
+ _debug(" Remote Audio IP: %s\n", _remote_sdp_audio_ip);
+ setRemoteIP(_remote_sdp_audio_ip);
+ if (_remote_sdp_audio_ip[0] == '\0') {
+ setRemoteAudioPort(0);
+ return false;
+ }
+ }
+
+ // Remote port
+ int _remote_sdp_audio_port = atoi(remote_med->m_port);
+ _debug(" Remote Audio Port: %d\n", _remote_sdp_audio_port);
+ setRemoteAudioPort(_remote_sdp_audio_port);
+
+ if (_remote_sdp_audio_port == 0) {
+ return false;
+ }
+ return true;
+}
+
+bool
+SIPCall::setAudioCodecFromSDP(sdp_media_t* remote_med, int tid)
+{
+ // Remote Payload
+ char *tmp = NULL;
+ 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 ) {
+ int payload = atoi(tmp);
+ // stop if we find a correct codec
+ if (0 != _codecMap.getCodec((CodecType)payload)){
+ break;
+ }
+ }
+ tmp = NULL;
+ pos++;
+ }
+
+ setAudioCodec(0);
+ if (tmp != NULL) {
+ int payload = atoi (tmp);
+ _debug(" Payload: %d\n", payload);
+ setAudioCodec(_codecMap.getCodec((CodecType)payload)); // codec builder for the mic
+ }
+ if (getAudioCodec() == 0) {
+ _debug("SIPCall Failure: Unable to set codec\n");
+ _debug("< Sending 415 Unsupported media type\n");
+ eXosip_lock();
+ eXosip_call_send_answer(tid, 415, NULL);
+ eXosip_unlock();
+ return false;
+ }
+ return true;
+}
--
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