diff --git a/src/sipmanager.cpp b/src/sipmanager.cpp
index 3574063854774cbeca743adac48c471798c787b8..81cf90b21e3f21d584a5cc850ecc9aae0cf6f198 100755
--- a/src/sipmanager.cpp
+++ b/src/sipmanager.cpp
@@ -64,9 +64,6 @@ pj_status_t SIPManager::sipCreate() {
return status;
}
- /* Set default sound device ID */
- //pjsua_var.cap_dev = pjsua_var.play_dev = -1;
-
/* Init caching pool. */
pj_caching_pool_init(&_cp, NULL, 0);
@@ -124,6 +121,8 @@ pj_status_t SIPManager::sipInit() {
_debug("SIPManager: Unable to check NAT setting\n");
return false; // hoho we can't use the random sip port too...
}
+ } else {
+ //FIXME! check port number availability
}
_localPort = port;
@@ -311,6 +310,7 @@ pj_status_t SIPManager::sipInit() {
_debug("SIPManager: sflphone version %s for %s initialized\n", pj_get_version(), PJ_OS_NAME);
+ Manager::instance().setSipThreadStatus(false);
status = pj_thread_create(_pool, "sflphone", &worker_thread,
NULL, 0, 0, &_thread);
@@ -421,12 +421,13 @@ bool SIPManager::addAccount(AccountID id, pjsip_regc **regc2, const std::string&
}
tmp = "sip:" + server;
- pj_str_t svr = pj_str((char *) tmp.data());
+ pj_strdup2(_pool, &svr, tmp.data());
+
tmp = "<sip:" + user + "@" + server + ">";
- pj_str_t aor = pj_str((char *) tmp.data());
+ pj_strdup2(_pool, &aor, tmp.data());
sprintf(contactTmp, "<sip:%s@%s:%d>", user.data(), _localExternAddress.data(), _localExternPort);
- pj_str_t contact = pj_str(contactTmp);
+ pj_strdup2(_pool, &contact, contactTmp);
//_debug("SIPManager: Get in %s %d %s\n", svr.ptr, svr.slen, aor.ptr);
_debug("SIPManager: Contact is %s\n", contact.ptr);
@@ -436,7 +437,10 @@ bool SIPManager::addAccount(AccountID id, pjsip_regc **regc2, const std::string&
return status;
}
- AccBaseInfo *info = new AccBaseInfo();
+ AccBaseInfo *info = getAccountInfoFromId(id);
+
+ if(info == NULL)
+ info = new AccBaseInfo();
pj_bzero(&info->cred, sizeof (info->cred));
pj_strdup2(_pool, &info->cred.username, user.data());
@@ -465,6 +469,40 @@ bool SIPManager::addAccount(AccountID id, pjsip_regc **regc2, const std::string&
pj_strdup2(_pool, &info->contact, contact.ptr);
_accBaseInfoList.push_back(info);
+ // associate regc with account
+ *regc2 = regc;
+
+ pj_mutex_unlock(_mutex);
+ return true;
+}
+
+bool SIPManager::removeAccount(pjsip_regc *regc)
+{
+ pj_status_t status = 0;
+ pjsip_tx_data *tdata = NULL;
+
+ pj_mutex_lock(_mutex);
+
+ if(regc) {
+ status = pjsip_regc_unregister(regc, &tdata);
+ if(status != PJ_SUCCESS) {
+ _debug("SIPManager: Unable to unregister regc.\n");
+ pj_mutex_unlock(_mutex);
+ return false;
+ }
+
+ status = pjsip_regc_send( regc, tdata );
+ if(status != PJ_SUCCESS) {
+ _debug("SIPManager: Unable to send regc request.\n");
+ pj_mutex_unlock(_mutex);
+ return false;
+ }
+ } else {
+ _debug("SIPManager: regc is null!\n");
+ pj_mutex_unlock(_mutex);
+ return false;
+ }
+
pj_mutex_unlock(_mutex);
return true;
}
@@ -493,11 +531,11 @@ pj_status_t SIPManager::stunServerResolve() {
stun_status = PJ_EPENDING;
// Init STUN socket
- stun_adr = pj_str((char *) _stunServer.data());
+ pj_strdup2(_pool, &stun_adr, _stunServer.data());
stun_status = pj_sockaddr_in_init(&stun_srv.ipv4, &stun_adr, (pj_uint16_t) 3478);
if (stun_status != PJ_SUCCESS) {
- _debug("SIPManager: Unresolved stud server!\n");
+ _debug("SIPManager: Unresolved stun server!\n");
stun_status = pj_gethostbyname(&stun_adr, &he);
if (stun_status == PJ_SUCCESS) {
@@ -514,6 +552,8 @@ int SIPManager::createUDPServer() {
pj_status_t status;
pj_str_t ipAddr;
pj_sockaddr_in bound_addr;
+ pjsip_host_port a_name;
+ char tmpIP[32];
// Init bound address to ANY
pj_memset(&bound_addr, 0, sizeof (bound_addr));
@@ -536,10 +576,8 @@ int SIPManager::createUDPServer() {
_debug("SIPManager: Use IP: %s\n", _localExternAddress.data());
// Create UDP-Server (default port: 5060)
- pjsip_host_port a_name;
- char tmpIP[32];
strcpy(tmpIP, _localExternAddress.data());
- a_name.host = pj_str(tmpIP);
+ pj_strdup2(_pool, &a_name.host, tmpIP);
a_name.port = (pj_uint16_t) _localExternPort;
status = pjsip_udp_transport_attach(_endpt, _sock, &a_name, 1, NULL);
@@ -559,13 +597,28 @@ void SIPManager::setStunServer(const char *server) {
}
void SIPManager::regc_cb(struct pjsip_regc_cbparam *param) {
+
AccountID *id = static_cast<AccountID *> (param->token);
+ SIPVoIPLink *voipLink;
+
_debug("SIPManager: Account ID is %s, Register result: %d, Status: %d\n", id->data(), param->status, param->code);
+
if (param->status == PJ_SUCCESS) {
if (param->code < 0 || param->code >= 300) {
+ /* Sometimes, the status is OK, but we still failed.
+ * So checking the code for real result
+ */
Manager::instance().getAccountLink(*id)->setRegistrationState(VoIPLink::Error);
} else
- Manager::instance().getAccountLink(*id)->setRegistrationState(VoIPLink::Registered);
+ // Registration/Unregistration is success
+ voipLink = dynamic_cast<SIPVoIPLink *>(Manager::instance().getAccountLink(*id));
+ if(!voipLink)
+ return;
+
+ if(voipLink->isRegister())
+ Manager::instance().getAccountLink(*id)->setRegistrationState(VoIPLink::Registered);
+ else
+ Manager::instance().getAccountLink(*id)->setRegistrationState(VoIPLink::Unregistered);
} else {
Manager::instance().getAccountLink(*id)->setRegistrationState(VoIPLink::Error);
}
@@ -579,7 +632,6 @@ SIPManager::loadSIPLocalIP() {
if (pj_gethostip(pj_AF_INET(), &ip_addr) != PJ_SUCCESS) {
// Update the registration state if no network capabilities found
_debug("SIPManager: Get host ip failed!\n");
- //setRegistrationState( ErrorNetwork );
returnValue = false;
} else {
_localIPAddress = std::string(pj_inet_ntoa(ip_addr.ipv4.sin_addr));
@@ -591,13 +643,11 @@ SIPManager::loadSIPLocalIP() {
/* Worker thread function. */
int SIPManager::worker_thread(void *arg) {
- //enum { TIMEOUT = 10 };
PJ_UNUSED_ARG(arg);
- while (true) {
- //int count;
-
+ // FIXME! maybe we should add a flag for exiting!
+ while (!Manager::instance().getSipThreadStatus()) {
pj_time_val timeout = {0, 10};
pjsip_endpt_handle_events(getInstance()->getEndPoint(), &timeout);
}
@@ -613,10 +663,14 @@ pj_bool_t SIPManager::mod_on_rx_request(pjsip_rx_data *rdata) {
pjsip_tx_data *tdata;
pjmedia_sdp_session *r_sdp;
+ /* Handle the incoming call invite in this function */
_debug("SIPManager: Callback on_rx_request is involved!\n");
PJ_UNUSED_ARG(rdata);
+ /* First, let's got the username and server name from the invite.
+ * We will use them to detect which account is the callee.
+ */
pjsip_uri *uri = rdata->msg_info.to->uri;
pjsip_sip_uri *sip_uri = (pjsip_sip_uri *) pjsip_uri_get_uri(uri);
@@ -625,6 +679,7 @@ pj_bool_t SIPManager::mod_on_rx_request(pjsip_rx_data *rdata) {
_debug("SIPManager: The receiver is : %s@%s\n", userName.data(), server.data());
+ /* Now, it is the time to find the information of the caller */
uri = rdata->msg_info.from->uri;
sip_uri = (pjsip_sip_uri *) pjsip_uri_get_uri(uri);
@@ -635,49 +690,75 @@ pj_bool_t SIPManager::mod_on_rx_request(pjsip_rx_data *rdata) {
/* Respond statelessly any non-INVITE requests with 500 */
if (rdata->msg_info.msg->line.req.method.id != PJSIP_INVITE_METHOD) {
if (rdata->msg_info.msg->line.req.method.id != PJSIP_ACK_METHOD) {
- reason = pj_str((char*) " user agent unable to handle this request ");
+ pj_strdup2(getInstance()->getAppPool(), &reason, "user agent unable to handle this request ");
pjsip_endpt_respond_stateless(getInstance()->getEndPoint(), rdata, MSG_METHOD_NOT_ALLOWED, &reason, NULL,
NULL);
return PJ_TRUE;
}
}
+
// Verify that we can handle the request
status = pjsip_inv_verify_request(rdata, &options, NULL, NULL, getInstance()->getEndPoint(), NULL);
if (status != PJ_SUCCESS) {
- reason = pj_str((char*) " user agent unable to handle this INVITE ");
+ pj_strdup2(getInstance()->getAppPool(), &reason, "user agent unable to handle this INVITE ");
pjsip_endpt_respond_stateless(getInstance()->getEndPoint(), rdata, MSG_METHOD_NOT_ALLOWED, &reason, NULL,
NULL);
return PJ_TRUE;
}
+ // Generate a new call ID for the incoming call!
CallID id = Manager::instance().getNewCallID();
- _debug("SIPManager: ##################################call id is %s\n", id.c_str());
+ _debug("SIPManager: The call id of the incoming call is %s\n", id.c_str());
SIPCall* call = new SIPCall(id, Call::Incoming);
if (!call) {
_debug("SIPManager: unable to create an incoming call");
return PJ_FALSE;
}
+ // Set the codec map, IP, peer number and so on... for the SIPCall object
getInstance()->setCallAudioLocal(call);
call->setCodecMap(Manager::instance().getCodecDescriptorMap());
call->setConnectionState(Call::Progressing);
call->setIp(getInstance()->getLocalIP());
call->setPeerNumber(peerNumber);
+
+ /* Call the SIPCallInvite function to generate the local sdp,
+ * remote sdp and negociator.
+ * This function is also used to set the parameters of audio RTP, including:
+ * local IP and port number
+ * remote IP and port number
+ * possilbe audio codec will be used in this call
+ */
if (call->SIPCallInvite(rdata, getInstance()->getAppPool())) {
+ // Get the account id of callee from username and server
AccountID id = getInstance()->getAccountIdFromNameAndServer(userName, server);
- _debug("SIPManager: ##################################account id is %s\n", id.c_str());
+ if(id == AccountNULL) {
+ _debug("SIPManager: Username %s doesn't match any account!\n");
+ delete call;
+ call = NULL;
+ return PJ_FALSE;
+ }
+ _debug("SIPManager: The callee account id is %s\n", id.c_str());
+
+ // Notify UI there is an incoming call
if (Manager::instance().incomingCall(call, id)) {
+ // Add this call to the callAccountMap in ManagerImpl
Manager::instance().getAccountLink(id)->addCall(call);
- //addCall(call);
- _debug("SIPManager: OK\n");
+ _debug("SIPManager: Notify UI success!\n");
} else {
+ // Fail to notify UI
delete call;
- call = 0;
+ call = NULL;
+ _debug("SIPManager: Fail to notify UI!\n");
+ return PJ_FALSE;
}
} else {
+ // Fail to collect call information
delete call;
- call = 0;
+ call = NULL;
+ _debug("SIPManager: Call SIPCallInvite failed!\n");
+ return PJ_FALSE;
}
/* Create the local dialog (UAS) */
@@ -687,12 +768,13 @@ pj_bool_t SIPManager::mod_on_rx_request(pjsip_rx_data *rdata) {
NULL);
return PJ_TRUE;
}
+
// Specify media capability during invite session creation
- pjsip_inv_session *inv;// = call->getInvSession();
- status = pjsip_inv_create_uas(dialog, rdata,
- call->getLocalSDPSession(), 0, &inv);
+ pjsip_inv_session *inv;
+ status = pjsip_inv_create_uas(dialog, rdata, call->getLocalSDPSession(), 0, &inv);
PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+ // Associate the call in the invite session
inv->mod_data[getInstance()->getModId()] = call;
// Send a 180/Ringing response
@@ -701,19 +783,19 @@ pj_bool_t SIPManager::mod_on_rx_request(pjsip_rx_data *rdata) {
status = pjsip_inv_send_msg(inv, tdata);
PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+ // Associate invite session to the current call
+ call->setInvSession(inv);
+
// Update the connection state
- if (call) {
- call->setInvSession(inv);
- call->setConnectionState(Call::Ringing);
- }
+ call->setConnectionState(Call::Ringing);
/* Done */
-
return PJ_SUCCESS;
}
AccountID SIPManager::getAccountIdFromNameAndServer(const std::string& userName, const std::string& server) {
int size = _accBaseInfoList.size();
+ // Try to find the account id from username and server name by full match
for (int i = 0; i < size; i++) {
if (_accBaseInfoList[i]->userName == userName &&
_accBaseInfoList[i]->server == server) {
@@ -722,6 +804,7 @@ AccountID SIPManager::getAccountIdFromNameAndServer(const std::string& userName,
}
}
+ // We failed! Then only match the username
for (int i = 0; i < size; i++) {
if (_accBaseInfoList[i]->userName == userName) {
//_debug("SIPManager: Username match\n");
@@ -729,6 +812,7 @@ AccountID SIPManager::getAccountIdFromNameAndServer(const std::string& userName,
}
}
+ // Failed again! return AccountNULL
return AccountNULL;
}
@@ -743,13 +827,13 @@ SIPManager::AccBaseInfo* SIPManager::getAccountInfoFromId(AccountID id) {
return NULL;
}
-bool
-SIPManager::setCallAudioLocal(SIPCall* call) {
- // Setting Audio
+bool SIPManager::setCallAudioLocal(SIPCall* call) {
+ // Firstly, we use the local IP and port number
unsigned int callLocalAudioPort = RANDOM_LOCAL_PORT;
unsigned int callLocalExternAudioPort = callLocalAudioPort;
+
if (_useStun) {
- // If use Stun server
+ // If use Stun server, modify them
if (Manager::instance().behindNat(_stunServer, callLocalAudioPort)) {
callLocalExternAudioPort = Manager::instance().getFirewallPort();
}
@@ -769,6 +853,7 @@ int SIPManager::answer(SIPCall *call) {
pj_status_t status;
pjsip_tx_data *tdata;
+ // User answered the incoming call, tell peer this news
if (call->startNegociation(_pool)) {
// Create and send a 200(OK) response
_debug("SIPManager: Negociation success!\n");
@@ -789,20 +874,25 @@ bool SIPManager::makeOutgoingCall(const std::string& strTo, SIPCall* call, const
pjsip_tx_data *tdata;
pj_str_t from, to;
+ // Get the basic information about the callee account
AccBaseInfo* accBase = getAccountInfoFromId(id);
+
+ // Generate the from URI
std::string strFrom = "sip:" + accBase->userName + "@" + accBase->server;
_debug("SIPManager: Make a new call from:%s to %s. Contact is %s\n",
strFrom.data(), strTo.data(), accBase->contact.ptr);
- from = pj_str((char *) strFrom.data());
- to = pj_str((char *) strTo.data());
+ // pjsip need the from and to information in pj_str_t format
+ pj_strdup2(_pool, &from, strFrom.data());
+ pj_strdup2(_pool, &to, strTo.data());
+ // create the dialog (UAC)
status = pjsip_dlg_create_uac(pjsip_ua_instance(), &from,
- &accBase->contact, //NULL,
- &to,
- NULL,
- &dialog);
+ &accBase->contact,
+ &to,
+ NULL,
+ &dialog);
PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
setCallAudioLocal(call);
@@ -811,13 +901,15 @@ bool SIPManager::makeOutgoingCall(const std::string& strTo, SIPCall* call, const
// Building the local SDP offer
call->createInitialOffer(_pool);
+ // Create the invite session for this call
pjsip_inv_session *inv;
- status = pjsip_inv_create_uac(dialog,
- call->getLocalSDPSession(), 0, &inv);
+ status = pjsip_inv_create_uac(dialog, call->getLocalSDPSession(), 0, &inv);
PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+ // Set auth information
pjsip_auth_clt_set_credentials(&dialog->auth_sess, 1, &accBase->cred);
+ // Associate current call in the invite session
inv->mod_data[_mod.id] = call;
status = pjsip_inv_invite(inv, &tdata);
@@ -826,13 +918,15 @@ bool SIPManager::makeOutgoingCall(const std::string& strTo, SIPCall* call, const
status = pjsip_inv_send_msg(inv, tdata);
PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+ // Associate current invite session in the call
call->setInvSession(inv);
return true;
}
void SIPManager::call_on_forked(pjsip_inv_session *inv, pjsip_event *e) {
- //_debug("HERE!\n");
+ PJ_UNUSED_ARG(inv);
+ PJ_UNUSED_ARG(e);
}
void SIPManager::call_on_tsx_changed(pjsip_inv_session *inv, pjsip_transaction *tsx, pjsip_event *e) {
@@ -843,7 +937,6 @@ void SIPManager::call_on_tsx_changed(pjsip_inv_session *inv, pjsip_transaction *
SIPVoIPLink *link;
pjsip_msg *msg;
- PJ_UNUSED_ARG(inv);
_debug("SIPManager: TSX Changed! The tsx->state is %d; tsx->role is %d; code is %d; method id is %.*s.\n",
tsx->state, tsx->role, tsx->status_code, tsx->method.name.slen, tsx->method.name.ptr);
@@ -855,6 +948,7 @@ void SIPManager::call_on_tsx_changed(pjsip_inv_session *inv, pjsip_transaction *
case PJSIP_TSX_STATE_TERMINATED:
if (tsx->status_code == 200 &&
pjsip_method_cmp(&tsx->method, pjsip_get_refer_method()) != 0) {
+ // Peer answered the outgoing call
_debug("SIPManager: Peer answered the outgoing call!\n");
call = reinterpret_cast<SIPCall *> (inv->mod_data[getInstance()->getModId()]);
if (call == NULL)
@@ -869,6 +963,7 @@ void SIPManager::call_on_tsx_changed(pjsip_inv_session *inv, pjsip_transaction *
}
break;
case PJSIP_TSX_STATE_PROCEEDING:
+ // Peer is ringing for the outgoing call
msg = rdata->msg_info.msg;
call = reinterpret_cast<SIPCall *> (inv->mod_data[getInstance()->getModId()]);
@@ -883,7 +978,8 @@ void SIPManager::call_on_tsx_changed(pjsip_inv_session *inv, pjsip_transaction *
}
break;
case PJSIP_TSX_STATE_COMPLETED:
- if (tsx->status_code / 100 == 6 || tsx->status_code == 404) {
+ if (tsx->status_code / 100 == 6 || tsx->status_code / 100 == 4) {
+ // We get error message of outgoing call from server
_debug("SIPManager: Server error message is received!\n");
call = reinterpret_cast<SIPCall *> (inv->mod_data[getInstance()->getModId()]);
if (call == NULL) {
@@ -905,15 +1001,14 @@ void SIPManager::call_on_tsx_changed(pjsip_inv_session *inv, pjsip_transaction *
switch (tsx->state) {
case PJSIP_TSX_STATE_TRYING:
if (pjsip_method_cmp(&tsx->method, pjsip_get_refer_method()) == 0) {
- /*
- * Incoming REFER request.
- */
+ // Peer ask me to transfer call to another number.
_debug("SIPManager: Incoming REFER request!\n");
getInstance()->onCallTransfered(inv, e->body.tsx_state.src.rdata);
}
break;
case PJSIP_TSX_STATE_COMPLETED:
if (tsx->status_code == 200 && tsx->method.id == PJSIP_BYE_METHOD) {
+ // Peer hangup the call
_debug("SIPManager: Peer hangup(bye) message is received!\n");
call = reinterpret_cast<SIPCall *> (inv->mod_data[getInstance()->getModId()]);
if (call == NULL) {
@@ -926,6 +1021,7 @@ void SIPManager::call_on_tsx_changed(pjsip_inv_session *inv, pjsip_transaction *
link->SIPCallClosed(call);
}
} else if (tsx->status_code == 200 && tsx->method.id == PJSIP_CANCEL_METHOD) {
+ // Peer refuse the call
_debug("SIPManager: Cancel message is received!\n");
call = reinterpret_cast<SIPCall *> (inv->mod_data[getInstance()->getModId()]);
if (call == NULL) {
@@ -952,9 +1048,9 @@ void SIPManager::call_on_state_changed(pjsip_inv_session *inv, pjsip_event *e) {
PJ_UNUSED_ARG(inv);
SIPCall *call = reinterpret_cast<SIPCall*> (inv->mod_data[getInstance()->getModId()]);
-
if(!call)
return;
+
/* If this is an outgoing INVITE that was created because of
* REFER/transfer, send NOTIFY to transferer.
*/
@@ -1057,6 +1153,7 @@ bool SIPManager::hangup(SIPCall* call) {
pj_status_t status;
pjsip_tx_data *tdata;
+ // User hangup current call. Notify peer
status = pjsip_inv_end_session(call->getInvSession(), 404, NULL, &tdata);
PJ_ASSERT_RETURN(status == PJ_SUCCESS, false);
@@ -1072,6 +1169,7 @@ bool SIPManager::refuse(SIPCall* call)
pj_status_t status;
pjsip_tx_data *tdata;
+ // User refuse current call. Notify peer
status = pjsip_inv_end_session(call->getInvSession(), PJSIP_SC_DECLINE, NULL, &tdata); //603
PJ_ASSERT_RETURN(status == PJ_SUCCESS, false);
@@ -1105,7 +1203,11 @@ bool SIPManager::transfer(SIPCall *call, const std::string& to)
return false;
}
- /* Associate this call with the client subscription */
+ /* Associate this voiplink of call with the client subscription
+ * We can not just associate call with the client subscription
+ * because after this function, we can not find the cooresponding
+ * voiplink from the call any more. But the voiplink is useful!
+ */
AccountID accId = Manager::instance().getAccountFromCall(call->getCallId());
SIPVoIPLink *link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink(accId));
pjsip_evsub_set_mod_data(sub, _mod.id, link);
@@ -1163,6 +1265,7 @@ void SIPManager::xfer_func_cb( pjsip_evsub *sub, pjsip_event *event)
* transfered successfully.
*/
if (link) {
+ // It's the time to stop the RTP
link->transferStep2();
}
@@ -1210,8 +1313,13 @@ void SIPManager::xfer_func_cb( pjsip_evsub *sub, pjsip_event *event)
return;
}
-
+ // Get current call
SIPCall *call = dynamic_cast<SIPCall *>(link->getCall(Manager::instance().getCurrentCallId()));
+ if(!call) {
+ _debug("SIPManager: Call doesn't exit!\n");
+ return;
+ }
+
/* This better be a NOTIFY request */
if (event->type == PJSIP_EVENT_TSX_STATE &&
event->body.tsx_state.type == PJSIP_EVENT_RX_MSG)
@@ -1283,7 +1391,6 @@ void SIPManager::onCallTransfered(pjsip_inv_session *inv, pjsip_rx_data *rdata)
pj_status_t status;
pjsip_tx_data *tdata;
SIPCall *existing_call;
- //int new_call;
const pj_str_t str_refer_to = { "Refer-To", 8};
const pj_str_t str_refer_sub = { "Refer-Sub", 9 };
const pj_str_t str_ref_by = { "Referred-By", 11 };
@@ -1292,7 +1399,6 @@ void SIPManager::onCallTransfered(pjsip_inv_session *inv, pjsip_rx_data *rdata)
pjsip_hdr *ref_by_hdr;
pj_bool_t no_refer_sub = PJ_FALSE;
char *uri;
- //pjsua_msg_data msg_data;
std::string tmp;
pjsip_status_code code;
pjsip_evsub *sub;
@@ -1465,6 +1571,10 @@ void SIPManager::onCallTransfered(pjsip_inv_session *inv, pjsip_rx_data *rdata)
SIPVoIPLink *link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink(accId));
if(link) {
newCall = dynamic_cast<SIPCall *>(link->getCall(newCallId));
+ if(!newCall) {
+ _debug("SIPManager: can not find the call from sipvoiplink!\n");
+ return;
+ }
}
if (sub) {
diff --git a/src/sipvoiplink.cpp b/src/sipvoiplink.cpp
old mode 100644
new mode 100755
index 487f67b63792e3285d15562d6cdcb0986c0762b0..5cec82447ff05cb33e256eeb840506638137dcf2
--- a/src/sipvoiplink.cpp
+++ b/src/sipvoiplink.cpp
@@ -457,15 +457,14 @@ SIPVoIPLink::sendRegister()
_regc = NULL;
}
+ _bRegister = true;
+
int expire_value = Manager::instance().getRegistrationExpireValue();
_debug("SIP Registration Expire Value = %i\n" , expire_value);
setRegistrationState(Trying);
- SIPManager *sipManager = Manager::instance().getSipManager();
- _debug("Get manager of sip\n");
-
- return sipManager->addAccount(id, _regc, _server, _authname, _password, expire_value);
+ return Manager::instance().getSipManager()->addAccount(id, &_regc, _server, _authname, _password, expire_value);
}
std::string
@@ -497,34 +496,11 @@ bool
SIPVoIPLink::sendUnregister()
{
_debug("SEND UNREGISTER for account %s\n" , getAccountID().c_str());
- if ( _eXosipRegID == EXOSIP_ERROR_STD) return false;
- int eXosipErr = EXOSIP_ERROR_NO;
- osip_message_t *reg = NULL;
-
- eXosip_lock();
- eXosipErr = eXosip_register_build_register (_eXosipRegID, 0, ®);
- eXosip_unlock();
-
- if (eXosipErr != EXOSIP_ERROR_NO) {
- _debug("! SIP Failure: Unable to build registration for sendUnregister");
- return false;
- }
-
- eXosip_lock();
- _debug("< Sending REGISTER (expire=0)\n");
- eXosipErr = eXosip_register_send_register (_eXosipRegID, reg);
- if (eXosipErr == EXOSIP_ERROR_BUILDING) {
- _debug("! SIP Failure: Cannot build registration (unregister), check the setup\n");
- eXosip_unlock();
- return false;
- }
- if (eXosipErr == EXOSIP_ERROR_STD) {
- _debug("! SIP Failure: Unable to send registration (unregister)\n");
- }
- _eXosipRegID = EXOSIP_ERROR_STD;
- eXosip_unlock();
-
+ _bRegister = false;
+
+ Manager::instance().getSipManager()->removeAccount(_regc);
+
return true;
}
@@ -592,7 +568,7 @@ SIPVoIPLink::hangup(const CallID& id)
//eXosip_lock();
//eXosip_call_terminate(call->getCid(), call->getDid());
//eXosip_unlock();
- Manager::instance().getSipManager()->hangup();
+ Manager::instance().getSipManager()->hangup(call);
// Release RTP thread
if (Manager::instance().isCurrentCall(id)) {
@@ -816,7 +792,7 @@ SIPVoIPLink::refuse (const CallID& id)
return false;
}
- Manager::instance().getSipManager()->refuse();
+ Manager::instance().getSipManager()->refuse(call);
/*osip_message_t *answerMessage = NULL;
eXosip_lock();
// not BUSY.. where decline the invitation!
@@ -1439,8 +1415,10 @@ SIPVoIPLink::SIPCallClosed(SIPCall *call)
_debug("* SIP Info: Stopping AudioRTP when closing\n");
_audiortp->closeRtpSession();
}
+ _debug("After close RTP\n");
Manager::instance().peerHungupCall(id);
removeCall(id);
+ _debug("After remove call ID\n");
}
void
diff --git a/src/sipvoiplink.h b/src/sipvoiplink.h
old mode 100644
new mode 100755
index e54c4fbd85e3a63903c1effb119e5a74c57d38c7..6ff0fc81b1feeb13a87170f1184291c932939f7d
--- a/src/sipvoiplink.h
+++ b/src/sipvoiplink.h
@@ -204,7 +204,7 @@ class SIPVoIPLink : public VoIPLink
void setSipServer(const std::string& sipServer);
- //void setS
+ bool isRegister() {return _bRegister;}
public:
@@ -446,6 +446,7 @@ class SIPVoIPLink : public VoIPLink
private:
pjsip_regc *_regc;
std::string _server;
+ bool _bRegister;
};
#endif