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Commit 3111a739 authored by yanmorin's avatar yanmorin
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Adding IAXCall and beginning of iax get event

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src/Makefile.am
+ 2
− 2
View file @ 3111a739
...@@ -15,8 +15,8 @@ if USE_IAX ...@@ -15,8 +15,8 @@ if USE_IAX
IAX_LIBS = $(top_builddir)/libs/libiax2/src/libiax.la IAX_LIBS = $(top_builddir)/libs/libiax2/src/libiax.la
IAX_FLAGS = -DUSE_IAX IAX_FLAGS = -DUSE_IAX
IAX_CFLAGS = -I$(top_srcdir)/libs/libiax2/src/ IAX_CFLAGS = -I$(top_srcdir)/libs/libiax2/src/
IAXSOURCES = iaxaccount.cpp iaxvoiplink.cpp IAXSOURCES = iaxaccount.cpp iaxvoiplink.cpp iaxcall.cpp
IAXHEADERS = iaxaccount.h iaxvoiplink.h IAXHEADERS = iaxaccount.h iaxvoiplink.h iaxcall.h
else else
IAX_LIBS = IAX_LIBS =
IAX_FLAGS = IAX_FLAGS =
......
src/iaxcall.cpp 0 → 100644
+ 653
− 0
View file @ 3111a739
/*
* 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
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#include "sipcall.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()
{
}
CodecDescriptorMap&
SIPCall::getCodecMap()
{
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;
}
src/iaxcall.h 0 → 100644
+ 49
− 0
View file @ 3111a739
/*
* Copyright (C) 2006 Savoir-Faire Linux inc.
* Author: Yan Morin <yan.morin@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
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#ifndef IAXCALL_H
#define IAXCALL_H
#include "call.h"
#include <iax-client.h>
/**
* IAXCall are IAX implementation of a normal Call
* @author Yan Morin <yan.morin@gmail.com>
*/
class IAXCall : public Call
{
public:
IAXCall(const CallID& id, Call::CallType type);
~IAXCall();
/** Get the session pointer or 0 */
struct iax_session* getSession() { return _session; }
/** Set the session pointer
* @param session the session pointer to assign
*/
void setSession(struct iax_session* session) { _session = session; }
private:
// each call is associate to a session
struct iax_session* _session;
};
#endif
src/iaxvoiplink.cpp
+ 59
− 0
View file @ 3111a739
...@@ -18,6 +18,8 @@ ...@@ -18,6 +18,8 @@
*/ */
#include "iaxvoiplink.h" #include "iaxvoiplink.h"
#include "global.h" // for _debug #include "global.h" // for _debug
#include "iaxcall.h"
#include "eventthread.h"
#define IAX_SUCCESS 0 #define IAX_SUCCESS 0
#define IAX_FAILURE -1 #define IAX_FAILURE -1
...@@ -25,11 +27,14 @@ ...@@ -25,11 +27,14 @@
IAXVoIPLink::IAXVoIPLink(const AccountID& accountID) IAXVoIPLink::IAXVoIPLink(const AccountID& accountID)
: VoIPLink(accountID) : VoIPLink(accountID)
{ {
_evThread = new EventThread(this);
} }
IAXVoIPLink::~IAXVoIPLink() IAXVoIPLink::~IAXVoIPLink()
{ {
delete _evThread; _evThread = 0;
terminate();
} }
bool bool
...@@ -59,3 +64,57 @@ IAXVoIPLink::terminate() ...@@ -59,3 +64,57 @@ IAXVoIPLink::terminate()
// hangup all call // hangup all call
// iax_hangup(calls[callNo].session,"Dumped Call"); // iax_hangup(calls[callNo].session,"Dumped Call");
} }
void
IAXVoIPLink::getEvent()
{
// mutex here
iax_event* event = 0;
IAXCall* call = 0;
while ( (event = iax_get_event(0)) != 0 ) {
call = iaxFindCallBySession(event->session);
if (call!=0) {
iaxHandleCallEvent(event, call);
}
return;
}
iax_event_free(event);
// thread wait 5 second
// unlock mutex
}
bool
IAXVoIPLink::setRegister()
{
return false;
}
bool
IAXVoIPLink::setUnregister()
{
return false;
}
IAXCall*
IAXVoIPLink::iaxFindCallBySession(struct iax_session* session)
{
// access to callMap shoud use that
// the code below is like findSIPCallWithCid()
ost::MutexLock m(_callMapMutex);
IAXCall* call = 0;
CallMap::iterator iter = _callMap.begin();
while(iter != _callMap.end()) {
call = dynamic_cast<IAXCall*>(iter->second);
if (call && call->getSession() == session) {
return call;
}
iter++;
}
return 0; // not found
}
void
IAXVoIPLink::iaxHandleCallEvent(iax_event* event, IAXCall* call)
{
}
src/iaxvoiplink.h
+ 24
− 5
View file @ 3111a739
...@@ -20,9 +20,10 @@ ...@@ -20,9 +20,10 @@
#define IAXVOIPLINK_H #define IAXVOIPLINK_H
#include "voIPLink.h" #include "voIPLink.h"
#include "iax-client.h" #include <iax-client.h>
class AudioCodec; class EventThread;
class IAXCall;
/** /**
* VoIPLink contains a thread that listen to external events * VoIPLink contains a thread that listen to external events
...@@ -36,13 +37,13 @@ public: ...@@ -36,13 +37,13 @@ public:
~IAXVoIPLink(); ~IAXVoIPLink();
void getEvent (void) { } void getEvent(void);
bool init (void); bool init (void);
bool checkNetwork (void) { return false; } bool checkNetwork (void) { return false; }
void terminate (void); void terminate (void);
bool setRegister (void) { return false; } bool setRegister (void);
bool setUnregister (void) { return false; } bool setUnregister (void);
Call* newOutgoingCall(const CallID& id, const std::string& toUrl) {return 0; } Call* newOutgoingCall(const CallID& id, const std::string& toUrl) {return 0; }
bool answer(const CallID& id) {return false;} bool answer(const CallID& id) {return false;}
...@@ -55,6 +56,24 @@ public: ...@@ -55,6 +56,24 @@ public:
bool refuse (const CallID& id) { return false; } bool refuse (const CallID& id) { return false; }
bool carryingDTMFdigits(const CallID& id, char code) { return false; } bool carryingDTMFdigits(const CallID& id, char code) { return false; }
bool sendMessage(const std::string& to, const std::string& body) { return false; } bool sendMessage(const std::string& to, const std::string& body) { return false; }
private:
/**
* Find a iaxcall by iax session number
* @param session an iax_session valid pointer
* @return iaxcall or 0 if not found
*/
IAXCall* iaxFindCallBySession(struct iax_session *session);
/**
* Handle IAX Event for a call
* @param event An iax_event pointer
* @param call An IAXCall pointer
*/
void iaxHandleCallEvent(iax_event* event, IAXCall* call);
EventThread* _evThread;
}; };
#endif #endif
src/sipcall.cpp
+ 6
− 627
View file @ 3111a739
/* /*
* Copyright (C) 2004-2006 Savoir-Faire Linux inc. * Copyright (C) 2006 Savoir-Faire Linux inc.
* Author: Yan Morin <yan.morin@savoirfairelinux.com> * 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 * 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 * it under the terms of the GNU General Public License as published by
...@@ -18,636 +17,16 @@ ...@@ -18,636 +17,16 @@
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/ */
#include "sipcall.h" #include "iaxcall.h"
#include "global.h" // for _debug #include "global.h" // for _debug
#include <sstream> // for media buffer
#define _SENDRECV 0 IAXCall::IAXCall(const CallID& id, Call::CallType type) : Call(id, type), _session(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()
{
}
CodecDescriptorMap&
SIPCall::getCodecMap()
{
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? IAXCall::~IAXCall()
int
SIPCall::sdp_complete_message(sdp_message_t * remote_sdp, osip_message_t * msg)
{ {
// Format port to a char* _session = 0; // just to be sure to don't have unknown pointer, do not delete it!
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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