diff --git a/src/sipaccount.cpp b/src/sipaccount.cpp
index 766d6d397e6dcc5906199d28053f021eeec128ba..5923493ec2b0156b06d222062a4136c67d0e62ef 100644
--- a/src/sipaccount.cpp
+++ b/src/sipaccount.cpp
@@ -1,3 +1,6 @@
+
+#include "voiplink.h"
+
/*
* Copyright (C) 2006-2007 Savoir-Faire Linux inc.
* Author: Alexandre Bourget <alexandre.bourget@savoirfairelinux.com>
@@ -44,12 +47,15 @@ SIPAccount::registerVoIPLink()
SIPVoIPLink* thislink = dynamic_cast<SIPVoIPLink*> (_link);
thislink->setStunServer(Manager::instance().getConfigString(_accountID,SIP_STUN_SERVER));
thislink->setUseStun( useStun!=0 ? true : false);
-
+
+ //SIPVoIPLink* thislink = dynamic_cast<SIPVoIPLink*> (_link);
_link->init();
+
// Stuff needed for SIP registration.
thislink->setProxy (Manager::instance().getConfigString(_accountID,SIP_PROXY));
thislink->setAuthName(Manager::instance().getConfigString(_accountID,SIP_USER));
thislink->setPassword(Manager::instance().getConfigString(_accountID,SIP_PASSWORD));
+ thislink->setSipServer(Manager::instance().getConfigString(_accountID,SIP_HOST));
_link->sendRegister();
}
diff --git a/src/sipaccount.h b/src/sipaccount.h
index e542032b02cd4ede155d3548d120aaf4e9c5cb6d..d547f8bbf0463c47b96933728ee78620e0c79d3f 100644
--- a/src/sipaccount.h
+++ b/src/sipaccount.h
@@ -22,6 +22,7 @@
#include "account.h"
+class SIPVoIPLink;
/**
* @file sipaccount.h
diff --git a/src/sipvoiplink.cpp b/src/sipvoiplink.cpp
index 3334bf226a7cd5efc9e2da79889e07f04d18c085..487f67b63792e3285d15562d6cdcb0986c0762b0 100644
--- a/src/sipvoiplink.cpp
+++ b/src/sipvoiplink.cpp
@@ -25,6 +25,8 @@
#include "sipcall.h"
#include <sstream> // for ostringstream
#include "sipaccount.h"
+#include "sipmanager.h"
+#include "audio/audiortp.h"
#include "manager.h"
#include "user_cfg.h" // SIGNALISATION / PULSE #define
@@ -56,7 +58,7 @@
SIPVoIPLink::SIPVoIPLink(const AccountID& accountID)
: VoIPLink(accountID), _localExternAddress("") , eXosip_running( false )
{
- _evThread = new EventThread(this);
+ _evThread = NULL;//new EventThread(this);
_nMsgVoicemail = 0;
@@ -65,6 +67,7 @@ SIPVoIPLink::SIPVoIPLink(const AccountID& accountID)
_nbTryListenAddr = 2; // number of times to try to start SIP listener
_localExternPort = 0;
+ _audiortp = new AudioRtp();
// to get random number for RANDOM_PORT
srand (time(NULL));
}
@@ -78,14 +81,14 @@ SIPVoIPLink::~SIPVoIPLink()
bool
SIPVoIPLink::init()
{
- if( eXosip_running ){
+ /*if( eXosip_running ){
delete _evThread;
_evThread=0;
_evThread= new EventThread( this );
eXosip_quit();
- }
+ }*/
- if (!_initDone) {
+ /*if (!_initDone) {
if (0 != eXosip_init()) {
_debug("! SIP Failure: Could not initialize eXosip\n");
return false;
@@ -152,15 +155,16 @@ SIPVoIPLink::init()
}
_initDone = true;
- eXosip_running = true;
- // Useless
+ eXosip_running = true;*/
+ _regc = NULL;
+ _initDone = true;
return true;
}
void
SIPVoIPLink::terminate()
{
- terminateSIPCall();
+ //terminateSIPCall();
// TODO The next line makes the daemon crash on
// account delete if at least one account is registered.
// It should called only when the last account
@@ -201,16 +205,16 @@ SIPVoIPLink::loadSIPLocalIP()
{
bool returnValue = true;
if (_localIPAddress == "127.0.0.1") {
- char* myIPAddress = new char[65];
- if (eXosip_guess_localip(AF_INET, myIPAddress, 64) == EXOSIP_ERROR_STD) {
+ pj_sockaddr ip_addr;
+ if (pj_gethostip(pj_AF_INET(), &ip_addr) != PJ_SUCCESS) {
// Update the registration state if no network capabilities found
+ _debug("Get host ip failed!\n");
setRegistrationState( ErrorNetwork );
returnValue = false;
} else {
- _localIPAddress = std::string(myIPAddress);
- _debug(" SIP Info: Checking network, setting local IP address to: %s\n", myIPAddress);
+ _localIPAddress = std::string(pj_inet_ntoa(ip_addr.ipv4.sin_addr));
+ _debug(" SIP Info: Checking network, setting local IP address to: %s\n", _localIPAddress.data());
}
- delete [] myIPAddress; myIPAddress = NULL;
}
return returnValue;
}
@@ -218,6 +222,7 @@ SIPVoIPLink::loadSIPLocalIP()
void
SIPVoIPLink::getEvent()
{
+#ifdef AAAAA
char* tmp2;
eXosip_event_t* event = eXosip_event_wait(0, 50);
eXosip_lock();
@@ -438,72 +443,29 @@ SIPVoIPLink::getEvent()
break;
}
eXosip_event_free(event);
+#endif
}
bool
SIPVoIPLink::sendRegister()
{
- int expire_value = Manager::instance().getRegistrationExpireValue();
- _debug("SIP Registration Expire Value = %i\n" , expire_value);
-
- if (_eXosipRegID != EXOSIP_ERROR_STD) {
- return false;
- }
-
- std::string hostname = getHostName();
- if (hostname.empty()) {
- return false;
- }
-
- if (_authname.empty()) {
- return false;
- }
-
- std::string proxy = "sip:" + _proxy;
- hostname = "sip:" + hostname;
- std::string from = SIPFromHeader(_authname, getHostName());
+ AccountID id = getAccountID();
+ pj_status_t status;
- osip_message_t *reg = NULL;
- eXosip_lock();
- if (!_proxy.empty()) {
- _debug("* SIP Info: Register from: %s to %s\n", from.data(), proxy.data());
- _eXosipRegID = eXosip_register_build_initial_register(from.data(),
- proxy.data(), NULL, expire_value, ®);
- } else {
- _debug("* SIP Info: Register from: %s to %s\n", from.data(), hostname.data());
- _eXosipRegID = eXosip_register_build_initial_register(from.data(),
- hostname.data(), NULL, expire_value, ®);
- }
- eXosip_unlock();
- if (_eXosipRegID < EXOSIP_ERROR_NO ) {
- return false;
- }
-
- if (!sendSIPAuthentification()) {
- _debug("* SIP Info: register without authentication\n");
- return false;
+ if(_regc) {
+ pjsip_regc_destroy(_regc);
+ _regc = NULL;
}
- osip_message_set_header (reg, "Event", "Registration");
- osip_message_set_header (reg, "Allow-Events", "presence");
-
- eXosip_lock();
- int eXosipErr = eXosip_register_send_register(_eXosipRegID, reg);
- if (eXosipErr == EXOSIP_ERROR_BUILDING) {
- _debug("! SIP Failure: Cannot build registration, check the setup\n");
- eXosip_unlock();
- return false;
- }
- if (eXosipErr == EXOSIP_ERROR_STD) {
- _debug("! SIP Failure: Registration sending failed\n");
- eXosip_unlock();
- return false;
- }
+ int expire_value = Manager::instance().getRegistrationExpireValue();
+ _debug("SIP Registration Expire Value = %i\n" , expire_value);
setRegistrationState(Trying);
- eXosip_unlock();
- return true;
+ SIPManager *sipManager = Manager::instance().getSipManager();
+ _debug("Get manager of sip\n");
+
+ return sipManager->addAccount(id, _regc, _server, _authname, _password, expire_value);
}
std::string
@@ -572,6 +534,7 @@ SIPVoIPLink::newOutgoingCall(const CallID& id, const std::string& toUrl)
SIPCall* call = new SIPCall(id, Call::Outgoing);
if (call) {
call->setPeerNumber(toUrl);
+ _debug("Try to make a call to: %s with call ID: %s\n", toUrl.data(), id.data());
// we have to add the codec before using it in SIPOutgoingInvite...
call->setCodecMap(Manager::instance().getCodecDescriptorMap());
if ( SIPOutgoingInvite(call) ) {
@@ -596,38 +559,16 @@ SIPVoIPLink::answer(const CallID& id)
return false;
}
- // Send 200 OK
- osip_message_t *answerMessage = NULL;
- eXosip_lock();
- int i = eXosip_call_build_answer(call->getTid(), SIP_OK, &answerMessage);
+ int i = Manager::instance().getSipManager()->answer(call);
+
if (i != 0) {
_debug("< SIP Building Error: send 400 Bad Request\n");
- eXosip_call_send_answer (call->getTid(), SIP_BAD_REQUEST, NULL);
+ //eXosip_call_send_answer (call->getTid(), SIP_BAD_REQUEST, NULL);
} else {
// use exosip, bug locked
i = 0;
- sdp_message_t *remote_sdp = eXosip_get_remote_sdp(call->getDid());
- if (remote_sdp!=NULL) {
- i = call->sdp_complete_message(remote_sdp, answerMessage);
- if (i!=0) {
- osip_message_free(answerMessage);
- }
- sdp_message_free(remote_sdp);
- }
- if (i != 0) {
- _debug("< SIP Error: send 415 Unsupported Media Type\n");
- eXosip_call_send_answer (call->getTid(), SIP_UNSUPPORTED_MEDIA_TYPE, NULL);
- } else {
- _debug("< SIP send 200 OK\n");
- eXosip_call_send_answer (call->getTid(), SIP_OK, answerMessage);
- }
- }
- eXosip_unlock();
-
- if(i==0) {
- // Incoming call is answered, start the sound thread.
_debug("* SIP Info: Starting AudioRTP when answering\n");
- if (_audiortp.createNewSession(call) >= 0) {
+ if (_audiortp->createNewSession(call) >= 0) {
call->setAudioStart(true);
call->setConnectionState(Call::Connected);
call->setState(Call::Active);
@@ -646,16 +587,17 @@ SIPVoIPLink::hangup(const CallID& id)
SIPCall* call = getSIPCall(id);
if (call==0) { _debug("! SIP Error: Call doesn't exist\n"); return false; }
- _debug("- SIP Action: Hang up call %s [cd: %3d %3d]\n", id.data(), call->getCid(), call->getDid());
+ //_debug("- SIP Action: Hang up call %s [cd: %3d %3d]\n", id.data(), call->getCid(), call->getDid());
// Release SIP stack.
- eXosip_lock();
- eXosip_call_terminate(call->getCid(), call->getDid());
- eXosip_unlock();
-
+ //eXosip_lock();
+ //eXosip_call_terminate(call->getCid(), call->getDid());
+ //eXosip_unlock();
+ Manager::instance().getSipManager()->hangup();
+
// Release RTP thread
if (Manager::instance().isCurrentCall(id)) {
_debug("* SIP Info: Stopping AudioRTP for hangup\n");
- _audiortp.closeRtpSession();
+ _audiortp->closeRtpSession();
}
removeCall(id);
return true;
@@ -683,15 +625,19 @@ SIPVoIPLink::onhold(const CallID& id)
SIPCall* call = getSIPCall(id);
if (call==0) { _debug("! SIP Error: call doesn't exist\n"); return false; }
+
// Stop sound
call->setAudioStart(false);
call->setState(Call::Hold);
_debug("* SIP Info: Stopping AudioRTP for onhold action\n");
- _audiortp.closeRtpSession();
+ _audiortp->closeRtpSession();
- int did = call->getDid();
+ //int did = call->getDid();
+ Manager::instance().getSipManager()->onhold(call);
+
+/*
eXosip_lock ();
sdp_message_t *local_sdp = eXosip_get_local_sdp(did);
eXosip_unlock ();
@@ -712,8 +658,9 @@ SIPVoIPLink::onhold(const CallID& id)
_debug("! SIP Failure: unable to build invite method to hold call\n");
return false;
}
-
+*/
/* add sdp body */
+ /*
{
char *tmp = NULL;
@@ -743,7 +690,7 @@ SIPVoIPLink::onhold(const CallID& id)
eXosip_lock ();
exosipErr = eXosip_call_send_request (did, invite);
eXosip_unlock ();
-
+ */
return true;
}
@@ -751,8 +698,11 @@ bool
SIPVoIPLink::offhold(const CallID& id)
{
SIPCall* call = getSIPCall(id);
- if (call==0) { _debug("! SIP Error: Call doesn't exist\n"); return false; }
+ if (call==0) { _debug("! SIP Error: Call doesn't exist\n"); return false; }
+
+ Manager::instance().getSipManager()->offhold(call);
+/*
int did = call->getDid();
eXosip_lock ();
@@ -774,8 +724,9 @@ SIPVoIPLink::offhold(const CallID& id)
sdp_message_free(local_sdp);
return EXOSIP_ERROR_STD;
}
-
+*/
/* add sdp body */
+/*
{
char *tmp = NULL;
@@ -803,12 +754,12 @@ SIPVoIPLink::offhold(const CallID& id)
eXosip_lock ();
exosipErr = eXosip_call_send_request (did, invite);
eXosip_unlock ();
-
+*/
// Enable audio
_debug("* SIP Info: Starting AudioRTP when offhold\n");
call->setState(Call::Active);
// it's sure that this is the current call id...
- if (_audiortp.createNewSession(call) < 0) {
+ if (_audiortp->createNewSession(call) < 0) {
_debug("! SIP Failure: Unable to start sound (%s:%d)\n", __FILE__, __LINE__);
return false;
}
@@ -826,7 +777,8 @@ SIPVoIPLink::transfer(const CallID& id, const std::string& to)
tmp_to = tmp_to + "@" + getHostName();
}
- osip_message_t *refer;
+ _debug("In transfer, tmp_to is %s\n", tmp_to.data());
+ /*osip_message_t *refer;
eXosip_lock();
// Build transfer request
int exosipErr = eXosip_call_build_refer(call->getDid(), (char*)tmp_to.data(), &refer);
@@ -835,14 +787,22 @@ SIPVoIPLink::transfer(const CallID& id, const std::string& to)
_debug("< SIP send transfer request to %s\n", tmp_to.data());
exosipErr = eXosip_call_send_request(call->getDid(), refer);
}
- eXosip_unlock();
+ eXosip_unlock();*/
+
+ Manager::instance().getSipManager()->transfer(call, tmp_to);
- _audiortp.closeRtpSession();
+ //_audiortp->closeRtpSession();
// shall we delete the call?
//removeCall(id);
return true;
}
+bool SIPVoIPLink::transferStep2()
+{
+ _audiortp->closeRtpSession();
+ return true;
+}
+
bool
SIPVoIPLink::refuse (const CallID& id)
{
@@ -856,15 +816,16 @@ SIPVoIPLink::refuse (const CallID& id)
return false;
}
-
- osip_message_t *answerMessage = NULL;
+ Manager::instance().getSipManager()->refuse();
+ /*osip_message_t *answerMessage = NULL;
eXosip_lock();
// not BUSY.. where decline the invitation!
int exosipErr = eXosip_call_build_answer(call->getTid(), SIP_DECLINE, &answerMessage);
if (exosipErr == 0) {
exosipErr = eXosip_call_send_answer(call->getTid(), SIP_DECLINE, answerMessage);
}
- eXosip_unlock();
+ eXosip_unlock();*/
+
return true;
}
@@ -1049,20 +1010,24 @@ SIPVoIPLink::SIPStartCall(SIPCall* call, const std::string& subject)
if (!call) return false;
std::string to = getSipTo(call->getPeerNumber());
- std::string from = getSipFrom();
std::string route = getSipRoute();
- _debug(" From: %s\n", from.data());
+ _debug(" To: %s\n", to.data());
_debug(" Route: %s\n", route.data());
- if (!SIPCheckUrl(from)) {
+ /*if (!SIPCheckUrl(from)) {
_debug("! SIP Error: Source address is invalid %s\n", from.data());
return false;
}
if (!SIPCheckUrl(to)) {
return false;
- }
+ }*/
- osip_message_t *invite;
+ //setCallAudioLocal(call);
+ AccountID accId = getAccountID();
+
+ Manager::instance().getSipManager()->makeOutgoingCall(to, call, accId);
+
+ /*osip_message_t *invite;
eXosip_lock();
int eXosipError = eXosip_call_build_initial_invite (&invite, (char*)to.data(),
(char*)from.data(),
@@ -1074,8 +1039,7 @@ SIPVoIPLink::SIPStartCall(SIPCall* call, const std::string& subject)
return false; // error when building the invite
}
- setCallAudioLocal(call);
-
+
std::ostringstream media_audio;
std::ostringstream rtpmap_attr;
AudioCodecType payload;
@@ -1104,14 +1068,14 @@ SIPVoIPLink::SIPStartCall(SIPCall* call, const std::string& subject)
}
// go to next codec
//*iter++;
- }
+ }*/
// http://www.antisip.com/documentation/eXosip2/group__howto1__initialize.html
// tell sip if we support SIP extension like 100rel
// osip_message_set_supported (invite, "100rel");
/* add sdp body */
- {
+ /*{
char tmp[4096];
snprintf (tmp, 4096,
"v=0\r\n"
@@ -1141,7 +1105,7 @@ SIPVoIPLink::SIPStartCall(SIPCall* call, const std::string& subject)
_debug("* SIP Info: Outgoing callID is %s, cid=%d\n", call->getCallId().data(), cid);
eXosip_call_set_reference (cid, NULL);
}
- eXosip_unlock();
+ eXosip_unlock();*/
return true;
}
@@ -1160,10 +1124,10 @@ SIPVoIPLink::getSipFrom() {
std::string
SIPVoIPLink::getSipTo(const std::string& to_url) {
// Form the From header field basis on configuration panel
- bool isRegistered = (_eXosipRegID == EXOSIP_ERROR_STD) ? false : true;
+ //bool isRegistered = (_eXosipRegID == EXOSIP_ERROR_STD) ? false : true;
// add a @host if we are registered and there is no one inside the url
- if (to_url.find("@") == std::string::npos && isRegistered) {
+ if (to_url.find("@") == std::string::npos) {// && isRegistered) {
std::string host = getHostName();
if(!host.empty()) {
return SIPToHeader(to_url + "@" + host);
@@ -1239,33 +1203,6 @@ SIPVoIPLink::setCallAudioLocal(SIPCall* call)
void
SIPVoIPLink::SIPCallInvite(eXosip_event_t *event)
{
- _debug("> INVITE (receive)\n");
- CallID id = Manager::instance().getNewCallID();
-
- SIPCall* call = new SIPCall(id, Call::Incoming);
- if (!call) {
- _debug("! SIP Failure: unable to create an incoming call");
- return;
- }
-
- setCallAudioLocal(call);
- call->setCodecMap(Manager::instance().getCodecDescriptorMap());
- call->setConnectionState(Call::Progressing);
- if (call->SIPCallInvite(event)) {
- if (Manager::instance().incomingCall(call, getAccountID())) {
- addCall(call);
- } else {
- delete call; call = 0;
- }
- } else {
- delete call; call = 0;
- }
- // Send 180 RINGING
- _debug("< Send 180 Ringing\n");
- eXosip_lock ();
- eXosip_call_send_answer(event->tid, 180, NULL);
- eXosip_unlock ();
- call->setConnectionState(Call::Ringing);
}
void
@@ -1286,7 +1223,7 @@ SIPVoIPLink::SIPCallReinvite(eXosip_event_t *event)
Manager::instance().stopTone(true);
// STOP old rtp session
_debug("* SIP Info: Stopping AudioRTP when reinvite\n");
- _audiortp.closeRtpSession();
+ _audiortp->closeRtpSession();
call->setAudioStart(false);
}
call->SIPCallReinvite(event);
@@ -1310,18 +1247,18 @@ SIPVoIPLink::SIPCallRinging(eXosip_event_t *event)
}
void
-SIPVoIPLink::SIPCallAnswered(eXosip_event_t *event)
+SIPVoIPLink::SIPCallAnswered(SIPCall *call, pjsip_rx_data *rdata)
{
- SIPCall* call = findSIPCallWithCid(event->cid);
+ //SIPCall* call = dynamic_cast<SIPCall *>(theCall);//findSIPCallWithCid(event->cid);
if (!call) {
_debug("! SIP Failure: unknown call\n");
return;
}
- call->setDid(event->did);
+ //call->setDid(event->did);
if (call->getConnectionState() != Call::Connected) {
- call->SIPCallAnswered(event);
- call->SIPCallAnsweredWithoutHold(event);
+ //call->SIPCallAnswered(event);
+ call->SIPCallAnsweredWithoutHold(rdata);
call->setConnectionState(Call::Connected);
call->setState(Call::Active);
@@ -1329,7 +1266,7 @@ SIPVoIPLink::SIPCallAnswered(eXosip_event_t *event)
Manager::instance().peerAnsweredCall(call->getCallId());
if (Manager::instance().isCurrentCall(call->getCallId())) {
_debug("* SIP Info: Starting AudioRTP when answering\n");
- if ( _audiortp.createNewSession(call) < 0) {
+ if ( _audiortp->createNewSession(call) < 0) {
_debug("RTP Failure: unable to create new session\n");
} else {
call->setAudioStart(true);
@@ -1337,7 +1274,7 @@ SIPVoIPLink::SIPCallAnswered(eXosip_event_t *event)
}
} else {
_debug("* SIP Info: Answering call (on/off hold to send ACK)\n");
- call->SIPCallAnswered(event);
+ //call->SIPCallAnswered(event);
}
}
@@ -1401,21 +1338,22 @@ SIPVoIPLink::SIPCallRequestFailure(eXosip_event_t *event)
}
void
-SIPVoIPLink::SIPCallServerFailure(eXosip_event_t *event)
+SIPVoIPLink::SIPCallServerFailure(SIPCall *call)
{
- if (!event->response) { return; }
- switch(event->response->status_code) {
- case SIP_SERVICE_UNAVAILABLE: // 500
- case SIP_BUSY_EVRYWHERE: // 600
- case SIP_DECLINE: // 603
- SIPCall* call = findSIPCallWithCid(event->cid);
+ //if (!event->response) { return; }
+ //switch(event->response->status_code) {
+ //case SIP_SERVICE_UNAVAILABLE: // 500
+ //case SIP_BUSY_EVRYWHERE: // 600
+ //case SIP_DECLINE: // 603
+ //SIPCall* call = findSIPCallWithCid(event->cid);
if (call != 0) {
+ _debug("Server error!\n");
CallID id = call->getCallId();
Manager::instance().callFailure(id);
removeCall(id);
}
- break;
- }
+ //break;
+ //}
}
void
@@ -1449,7 +1387,7 @@ SIPVoIPLink::SIPCallAck(eXosip_event_t *event)
if (!call->isAudioStarted()) {
if (Manager::instance().isCurrentCall(call->getCallId())) {
_debug("* SIP Info: Starting AudioRTP when ack\n");
- if ( _audiortp.createNewSession(call) ) {
+ if ( _audiortp->createNewSession(call) ) {
call->setAudioStart(true);
}
}
@@ -1488,29 +1426,29 @@ SIPVoIPLink::SIPCallMessageNew(eXosip_event_t *event)
}
void
-SIPVoIPLink::SIPCallClosed(eXosip_event_t *event)
+SIPVoIPLink::SIPCallClosed(SIPCall *call)
{
// it was without did before
- SIPCall* call = findSIPCallWithCid(event->cid);
+ //SIPCall* call = findSIPCallWithCid(event->cid);
if (!call) { return; }
CallID id = call->getCallId();
- call->setDid(event->did);
+ //call->setDid(event->did);
if (Manager::instance().isCurrentCall(id)) {
call->setAudioStart(false);
_debug("* SIP Info: Stopping AudioRTP when closing\n");
- _audiortp.closeRtpSession();
+ _audiortp->closeRtpSession();
}
Manager::instance().peerHungupCall(id);
removeCall(id);
}
void
-SIPVoIPLink::SIPCallReleased(eXosip_event_t *event)
+SIPVoIPLink::SIPCallReleased(SIPCall *call)
{
// do cleanup if exists
// only cid because did is always 0 in these case..
- SIPCall* call = findSIPCallWithCid(event->cid);
+ //SIPCall* call = findSIPCallWithCid(event->cid);
if (!call) { return; }
// if we are here.. something when wrong before...
@@ -1790,3 +1728,36 @@ SIPVoIPLink::sdp_off_hold_call (sdp_message_t * sdp)
return 0;
}
+
+void SIPVoIPLink::setAuthName(const std::string& authname)
+{
+ _authname = authname;
+ //_cred.username = pj_str((char *)authname.data());
+}
+
+void SIPVoIPLink::setPassword(const std::string& password)
+{
+ _password = password;
+ /*_cred.data_type = PJSIP_CRED_DATA_PLAIN_PASSWD;
+ _cred.data = pj_str((char *)password.data());
+ _cred.realm = pj_str("*");
+ _cred.scheme = pj_str("digest");*/
+}
+
+void SIPVoIPLink::setSipServer(const std::string& sipServer)
+{
+ //std::string tmp;
+ _debug("Set sip server %s\n", sipServer.data());
+ _server = sipServer;
+ //_registrar = pj_str((char *)tmp.data());
+ //_user = "<sip:" + _authname + "@" + sipServer + ">";
+ //_user = pj_str((char *)tmp.data());
+}
+
+pj_str_t SIPVoIPLink::string2PJStr(const std::string &value)
+{
+ char tmp[256];
+
+ strcpy(tmp, value.data());
+ return pj_str(tmp);
+}
diff --git a/src/sipvoiplink.h b/src/sipvoiplink.h
index 441e34b2c69e68f2e46138703c4d2db6f1f67114..e54c4fbd85e3a63903c1effb119e5a74c57d38c7 100644
--- a/src/sipvoiplink.h
+++ b/src/sipvoiplink.h
@@ -24,11 +24,15 @@
#include "voiplink.h"
#include <string>
#include <eXosip2/eXosip.h>
-#include "audio/audiortp.h"
#include <osip2/osip_mt.h>
+struct pjsip_regc;
+struct pj_str_t;
+struct pjmedia_sdp_session;
+
class EventThread;
class SIPCall;
+class AudioRtp;
/**
* @file sipvoiplink.h
@@ -139,6 +143,9 @@ class SIPVoIPLink : public VoIPLink
*/
bool transfer(const CallID& id, const std::string& to);
+ /** Handle the incoming refer msg, not finished yet */
+ bool transferStep2();
+
/**
* Refuse the call
* @param id The call identifier
@@ -187,15 +194,19 @@ class SIPVoIPLink : public VoIPLink
* Set the authentification name
* @param authname The authentification name
*/
- void setAuthName(const std::string& authname) { _authname = authname; }
+ void setAuthName(const std::string& authname); //{ _authname = authname; }
/**
* Set the password
* @param password Password
*/
- void setPassword(const std::string& password) { _password = password; }
+ void setPassword(const std::string& password); //{ _password = password; }
+
+ void setSipServer(const std::string& sipServer);
+
+ //void setS
- private:
+ public:
/**
* Terminate every call not hangup | brutal | Protected by mutex
@@ -305,7 +316,8 @@ class SIPVoIPLink : public VoIPLink
* @param event eXosip Event
*/
void SIPCallAnswered(eXosip_event_t *event);
-
+ void SIPCallAnswered(SIPCall *call, pjsip_rx_data *rdata);
+
/**
* Handling 4XX error
* @param event eXosip Event
@@ -316,7 +328,7 @@ class SIPVoIPLink : public VoIPLink
* Handling 5XX/6XX error
* @param event eXosip Event
*/
- void SIPCallServerFailure(eXosip_event_t *event);
+ void SIPCallServerFailure(SIPCall *call);
/**
* Handle registration failure cases ( SIP_FORBIDDEN , SIP_UNAUTHORIZED )
@@ -346,14 +358,14 @@ class SIPVoIPLink : public VoIPLink
* Peer close the connection
* @param event eXosip Event
*/
- void SIPCallClosed(eXosip_event_t *event);
+ void SIPCallClosed(SIPCall *call);
/**
* The call pointer was released
* If the call was not cleared before, report an error
* @param event eXosip Event
*/
- void SIPCallReleased(eXosip_event_t *event);
+ void SIPCallReleased(SIPCall *call);
/**
* Receive a new Message request
@@ -390,7 +402,7 @@ class SIPVoIPLink : public VoIPLink
/** To build sdp when call is off-hold */
int sdp_off_hold_call (sdp_message_t * sdp);
-
+
/** EventThread get every incoming events */
EventThread* _evThread;
@@ -428,7 +440,12 @@ class SIPVoIPLink : public VoIPLink
std::string _password;
/** Starting sound */
- AudioRtp _audiortp;
+ AudioRtp* _audiortp;
+
+ pj_str_t string2PJStr(const std::string &value);
+private:
+ pjsip_regc *_regc;
+ std::string _server;
};
#endif
diff --git a/src/voiplink.h b/src/voiplink.h
index 72eb61462aa3b5f0bef7bf8adf0d9c390b51202d..be69aaefbf3e520679b037a10828464c141432e2 100644
--- a/src/voiplink.h
+++ b/src/voiplink.h
@@ -29,6 +29,8 @@
class AudioCodec;
+struct pjsip_rx_data;
+
/** Define AccountID type */
typedef std::string AccountID;
@@ -277,7 +279,8 @@ class VoIPLink {
*/
int _registrationError;
- protected:
+ //protected:
+public:
/** Add a call to the call map (protected by mutex)
* @param call A call pointer with a unique pointer
* @return bool True if the call was unique and added
@@ -295,7 +298,9 @@ class VoIPLink {
* @return bool True on success
*/
bool clearCallMap();
-
+
+
+protected:
/** Contains all the calls for this Link, protected by mutex */
CallMap _callMap;