diff --git a/sflphone-common/src/sip/sipvoiplink.cpp b/sflphone-common/src/sip/sipvoiplink.cpp
index 87f15b2ac8488d3d06f04f2fe8668c505144460c..e2c0fda5f1d72b4972c35cf944c56b5208df67ca 100644
--- a/sflphone-common/src/sip/sipvoiplink.cpp
+++ b/sflphone-common/src/sip/sipvoiplink.cpp
@@ -480,7 +480,6 @@ int SIPVoIPLink::sendRegister (AccountID id)
SIPAccount *account = NULL;
pjsip_regc *regc;
pjsip_hdr hdr_list;
- pjsip_host_info dest_info;
account = dynamic_cast<SIPAccount *> (Manager::instance().getAccount (id));
@@ -856,6 +855,9 @@ SIPVoIPLink::answer (const CallID& id)
SIPCall *call = getSIPCall (id);
+// AccountID account_id = Manager::instance().getAccountFromCall (id);
+// SIPAccount *account = dynamic_cast<SIPAccount *> (Manager::instance().getAccount (account_id));
+
_debug ("UserAgent: pool capacity %d", pj_pool_get_capacity (_pool));
_debug ("UserAgent: pool size %d", pj_pool_get_used_size (_pool));
@@ -867,6 +869,20 @@ SIPVoIPLink::answer (const CallID& id)
inv_session = call->getInvSession();
+// if (! (account->getServiceRoute().empty())) {
+//
+// pjsip_route_hdr *route_set = pjsip_route_hdr_create (_pool);
+// pjsip_route_hdr *routing = pjsip_route_hdr_create (_pool);
+// pjsip_sip_uri *url = pjsip_sip_uri_create (_pool, 0);
+// routing->name_addr.uri = (pjsip_uri*) url;
+// _error ("UserAgent: Set Service-Route with %s", account->getServiceRoute().c_str());
+// pj_strdup2 (_pool, &url->host, account->getServiceRoute().c_str());
+//
+// pj_list_push_back (&route_set, pjsip_hdr_clone (_pool, routing));
+//
+// pjsip_dlg_set_route_set (inv_session->dlg, route_set);
+// }
+
if (status == PJ_SUCCESS) {
_debug ("SIPVoIPLink: UserAgent: SDP Negociation success! : call %s ", call->getCallId().c_str());
@@ -2056,55 +2072,41 @@ bool SIPVoIPLink::pjsip_init()
// Initialize transaction layer
status = pjsip_tsx_layer_init_module (_endpt);
-
PJ_ASSERT_RETURN (status == PJ_SUCCESS, 1);
// Initialize UA layer module
status = pjsip_ua_init_module (_endpt, NULL);
-
PJ_ASSERT_RETURN (status == PJ_SUCCESS, 1);
// Initialize Replaces support. See the Replaces specification in RFC 3891
status = pjsip_replaces_init_module (_endpt);
-
PJ_ASSERT_RETURN (status == PJ_SUCCESS, 1);
// Initialize 100rel support
status = pjsip_100rel_init_module (_endpt);
-
PJ_ASSERT_RETURN (status == PJ_SUCCESS, 1);
// Initialize and register sflphone module
_mod_ua.name = pj_str ( (char*) name_mod.c_str());
-
_mod_ua.id = -1;
-
_mod_ua.priority = PJSIP_MOD_PRIORITY_APPLICATION;
-
_mod_ua.on_rx_request = &transaction_request_cb;
-
_mod_ua.on_rx_response = &transaction_response_cb;
-
status = pjsip_endpt_register_module (_endpt, &_mod_ua);
-
PJ_ASSERT_RETURN (status == PJ_SUCCESS, 1);
// Init the event subscription module.
// It extends PJSIP by supporting SUBSCRIBE and NOTIFY methods
status = pjsip_evsub_init_module (_endpt);
-
PJ_ASSERT_RETURN (status == PJ_SUCCESS, 1);
// Init xfer/REFER module
status = pjsip_xfer_init_module (_endpt);
-
PJ_ASSERT_RETURN (status == PJ_SUCCESS, 1);
-
PJ_ASSERT_RETURN (status == PJ_SUCCESS, 1);
// Init the callback for INVITE session:
pj_bzero (&inv_cb, sizeof (inv_cb));
-
inv_cb.on_state_changed = &invite_session_state_changed_cb;
inv_cb.on_new_session = &outgoing_request_forked_cb;
inv_cb.on_media_update = &sdp_media_update_cb;
@@ -2114,14 +2116,12 @@ bool SIPVoIPLink::pjsip_init()
// Initialize session invite module
status = pjsip_inv_usage_init (_endpt, &inv_cb);
-
PJ_ASSERT_RETURN (status == PJ_SUCCESS, 1);
_debug ("UserAgent: VOIP callbacks initialized");
// Add endpoint capabilities (INFO, OPTIONS, etc) for this UA
pj_str_t allowed[] = { { (char*) "INFO", 4}, { (char*) "REGISTER", 8}, { (char*) "OPTIONS", 7}, { (char*) "MESSAGE", 7 } }; // //{"INVITE", 6}, {"ACK",3}, {"BYE",3}, {"CANCEL",6}
-
accepted = pj_str ( (char*) "application/sdp");
// Register supported methods
@@ -3180,30 +3180,22 @@ void invite_session_state_changed_cb (pjsip_inv_session *inv, pjsip_event *e)
{
_debug ("UserAgent: Call state changed to %s", invitationStateMap[inv->state]);
- pjsip_rx_data *rdata;
-
/* Retrieve the call information */
- SIPCall * call = NULL;
- call = reinterpret_cast<SIPCall*> (inv->mod_data[_mod_ua.id]);
-
+ SIPCall *call = reinterpret_cast<SIPCall*> (inv->mod_data[_mod_ua.id]);
if (call == NULL) {
_error ("UserAgent: Error: Call is NULL in call state changed callback");
return;
- } else {
}
//Retrieve the body message
- rdata = e->body.tsx_state.src.rdata;
+ pjsip_rx_data *rdata = e->body.tsx_state.src.rdata;
// If the call is a direct IP-to-IP call
- AccountID accId;
-
SIPVoIPLink * link = NULL;
-
if (call->getCallConfiguration () == Call::IPtoIP) {
link = SIPVoIPLink::instance ("");
} else {
- accId = Manager::instance().getAccountFromCall (call->getCallId());
+ AccountID accId = Manager::instance().getAccountFromCall (call->getCallId());
link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink (accId));
}
@@ -3212,47 +3204,26 @@ void invite_session_state_changed_cb (pjsip_inv_session *inv, pjsip_event *e)
return;
}
- /*
- pjsip_hdr *allow_header = NULL;
- std::string *allowed_options = NULL;
-
- char header_buffer[500];
-
- if (e->body.tsx_state.src.rdata->msg_info.msg)
- allow_header = (pjsip_hdr *) pjsip_msg_find_hdr (e->body.tsx_state.src.rdata->msg_info.msg, PJSIP_H_ALLOW, NULL);
-
- if (allow_header) {
- allowed_options = new std::string (allow_header->name.ptr, allow_header->name.slen);
- allow_header->vptr->print_on (allow_header, header_buffer, 5000);
- std::string theHeader (header_buffer);
- }
-
- if (allowed_options)
- delete allowed_options;
- */
-
// If this is an outgoing INVITE that was created because of
// REFER/transfer, send NOTIFY to transferer.
if (call->getXferSub() && e->type==PJSIP_EVENT_TSX_STATE) {
+ _debug("============================= invite_session_state_changed_cb case call->getXferSub() ====================== ");
+
int st_code = -1;
pjsip_evsub_state ev_state = PJSIP_EVSUB_STATE_ACTIVE;
switch (call->getInvSession()->state) {
case PJSIP_INV_STATE_NULL:
-
case PJSIP_INV_STATE_CALLING:
/* Do nothing */
break;
-
case PJSIP_INV_STATE_EARLY:
-
case PJSIP_INV_STATE_CONNECTING:
st_code = e->body.tsx_state.tsx->status_code;
ev_state = PJSIP_EVSUB_STATE_ACTIVE;
break;
-
case PJSIP_INV_STATE_CONFIRMED:
/* When state is confirmed, send the final 200/OK and terminate
* subscription.
@@ -3295,7 +3266,6 @@ void invite_session_state_changed_cb (pjsip_inv_session *inv, pjsip_event *e)
if (inv->state != PJSIP_INV_STATE_CONFIRMED) {
// Update UI with the current status code and description
- //pjsip_transaction * tsx
pjsip_transaction * tsx = NULL;
tsx = e->body.tsx_state.tsx;
int statusCode = 404;
@@ -3313,44 +3283,32 @@ void invite_session_state_changed_cb (pjsip_inv_session *inv, pjsip_event *e)
}
}
- // The call is ringing - We need to handle this case only on outgoing call
if (inv->state == PJSIP_INV_STATE_EARLY && e->body.tsx_state.tsx->role == PJSIP_ROLE_UAC) {
+ // The call is ringing - We need to handle this case only on outgoing call
call->setConnectionState (Call::Ringing);
Manager::instance().peerRingingCall (call->getCallId());
- }
-
-
- // After we sent or received a ACK - The connection is established
- else if (inv->state == PJSIP_INV_STATE_CONFIRMED) {
-
+ } else if (inv->state == PJSIP_INV_STATE_CONFIRMED) {
+ // After we sent or received a ACK - The connection is established
link->SIPCallAnswered (call, rdata);
-
} else if (inv->state == PJSIP_INV_STATE_DISCONNECTED) {
_debug ("UserAgent: State: %s. Cause: %.*s", invitationStateMap[inv->state], (int) inv->cause_text.slen, inv->cause_text.ptr);
- accId = Manager::instance().getAccountFromCall (call->getCallId());
- link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink (accId));
-
- // Make sure link is valid
- assert (link);
+ AccountID accId = Manager::instance().getAccountFromCall (call->getCallId());
+ if((link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink (accId))) == NULL)
+ return;
switch (inv->cause) {
-
- // The call terminates normally - BYE / CANCEL
+ // The call terminates normally - BYE / CANCEL
case PJSIP_SC_OK:
case PJSIP_SC_REQUEST_TERMINATED:
link->SIPCallClosed (call);
break;
-
case PJSIP_SC_DECLINE:
_debug ("UserAgent: Call %s is declined", call->getCallId().c_str());
-
if (inv->role == PJSIP_ROLE_UAC)
link->SIPCallServerFailure (call);
-
break;
-
case PJSIP_SC_NOT_FOUND: /* peer not found */
case PJSIP_SC_REQUEST_TIMEOUT: /* request timeout */
case PJSIP_SC_NOT_ACCEPTABLE_HERE: /* no compatible codecs */
@@ -3362,7 +3320,6 @@ void invite_session_state_changed_cb (pjsip_inv_session *inv, pjsip_event *e)
case PJSIP_SC_ADDRESS_INCOMPLETE:
link->SIPCallServerFailure (call);
break;
-
default:
link->SIPCallServerFailure (call);
_error ("UserAgent: Unhandled call state. This is probably a bug.");
@@ -3383,7 +3340,6 @@ void sdp_media_update_cb (pjsip_inv_session *inv, pj_status_t status)
SIPCall * call;
call = reinterpret_cast<SIPCall *> (inv->mod_data[getModId() ]);
-
if (!call) {
_debug ("UserAgent: Call declined by peer, SDP negociation stopped");
return;
@@ -3760,7 +3716,6 @@ transaction_request_cb (pjsip_rx_data *rdata)
SIPCall* call;
pjsip_inv_session *inv;
pjmedia_sdp_session *r_sdp;
- pjsip_response_addr *res_addr;
_info ("UserAgent: Transaction REQUEST received using transport: %s %s (refcnt=%d)",
rdata->tp_info.transport->obj_name,
@@ -4117,9 +4072,10 @@ transaction_request_cb (pjsip_rx_data *rdata)
return false;
}
- /* Done */
- return true;
}
+
+ /* Done */
+ return true;
}
pj_bool_t transaction_response_cb (pjsip_rx_data *rdata)