diff --git a/sflphone-common/src/sip/sipaccount.cpp b/sflphone-common/src/sip/sipaccount.cpp
index 2028c1fa87693b68a1ec9fc397df2ebd9700509b..52c7d49cd9cd0b3ace882309eab007659e6bafb1 100644
--- a/sflphone-common/src/sip/sipaccount.cpp
+++ b/sflphone-common/src/sip/sipaccount.cpp
@@ -1297,7 +1297,6 @@ std::string SIPAccount::getToUri (const std::string& username)
std::string hostname = "";
// UDP does not require the transport specification
-
if (_transportType == PJSIP_TRANSPORT_TLS) {
scheme = "sips:";
transport = ";transport=" + std::string (pjsip_transport_get_type_name (_transportType));
diff --git a/sflphone-common/src/sip/sipvoiplink.cpp b/sflphone-common/src/sip/sipvoiplink.cpp
index 82477e6fcaa66b45da5f703fdd8eb8d50189e623..a83fd733287ba6a184757a010e37e5f3b09308d6 100644
--- a/sflphone-common/src/sip/sipvoiplink.cpp
+++ b/sflphone-common/src/sip/sipvoiplink.cpp
@@ -254,15 +254,22 @@ pj_bool_t transaction_request_cb (pjsip_rx_data *rdata);
pj_bool_t transaction_response_cb (pjsip_rx_data *rdata UNUSED) ;
/**
- * Send an ACK message inside a transaction. PJSIP send automatically, non-2xx ACK response.
- * ACK for a 2xx response must be send using this method.
- */
+ * Send an ACK message inside a transaction. PJSIP send automatically, non-2xx ACK response.
+ * ACK for a 2xx response must be send using this method.
+ */
static void sendAck (pjsip_dialog *dlg, pjsip_rx_data *rdata);
+/**
+ * Send a reINVITE inside an active dialog to modify its state
+ * Local SDP session should be modified before calling this method
+ * @param sip call
+ */
+int SIPSessionReinvite(SIPCall *);
+
/*
* Transfer callbacks
*/
-void transfer_function_cb (pjsip_evsub *sub, pjsip_event *event);
+void transfer_client_cb (pjsip_evsub *sub, pjsip_event *event);
void transfer_server_cb (pjsip_evsub *sub, pjsip_event *event);
/**
@@ -643,17 +650,16 @@ Call *SIPVoIPLink::newOutgoingCall (const CallID& id, const std::string& toUrl)
pj_status_t status;
std::string localAddr, addrSdp;
- _debug ("UserAgent: New outgoing call %s to %s", id.c_str(), toUrl.c_str());
-
- _error ("UserAgent: pool capacity %d", pj_pool_get_capacity (_pool));
- _error ("UserAgent: pool size %d", pj_pool_get_used_size (_pool));
+ // _error ("UserAgent: pool capacity %d", pj_pool_get_capacity (_pool));
+ // _error ("UserAgent: pool size %d", pj_pool_get_used_size (_pool));
+ // Create a new SIP call
SIPCall* call = new SIPCall (id, Call::Outgoing, _cp);
if(call == NULL) {
- return NULL;
+ throw VoipLinkException("Could not create new SIP call");
}
-
+ // Find the account associated to this call
account = dynamic_cast<SIPAccount *> (Manager::instance().getAccount (Manager::instance().getAccountFromCall (id)));
if (account == NULL) {
_error ("UserAgent: Error: Could not retrieving account to make call with");
@@ -661,32 +667,47 @@ Call *SIPVoIPLink::newOutgoingCall (const CallID& id, const std::string& toUrl)
call->setState (Call::Error);
delete call;
call = NULL;
- return call;
+ // TODO: We should investigate how we could get rid of this error and create a IP2IP call instead
+ throw VoipLinkException("Could not get account for this call");
}
- std::string toUri = account->getToUri (toUrl);
+ // If toUri is not a well formated sip URI, use account information to process it
+ std::string toUri;
+ if((toUrl.find("sip:") != std::string::npos) ||
+ toUrl.find("sips:") != std::string::npos) {
+ toUri = toUrl;
+ }
+ else {
+ toUri = account->getToUri (toUrl);
+ }
call->setPeerNumber (toUri);
+ _debug ("UserAgent: New outgoing call %s to %s", id.c_str(), toUri.c_str());
localAddr = getInterfaceAddrFromName (account->getLocalInterface ());
+ _debug ("UserAgent: Local address for thi call: %s", localAddr.c_str());
- _debug ("UserAgent: Local address for call: %s", localAddr.c_str());
-
- if (localAddr == "0.0.0.0")
+ if (localAddr == "0.0.0.0") {
loadSIPLocalIP (&localAddr);
+ }
setCallAudioLocal (call, localAddr);
// May use the published address as well
account->isStunEnabled () ? addrSdp = account->getPublishedAddress () : addrSdp = getInterfaceAddrFromName (account->getLocalInterface ());
- if (addrSdp == "0.0.0.0")
+ if (addrSdp == "0.0.0.0") {
loadSIPLocalIP (&addrSdp);
+ }
+ // Initialize the session using ULAW as default codec in case of early media
+ // The session should be ready to receive media once the first INVITE is sent, before
+ // the session initialization is completed
sfl::Codec* audiocodec = Manager::instance().getCodecDescriptorMap().instantiateCodec (PAYLOAD_CODEC_ULAW);
-
if (audiocodec == NULL) {
_error ("UserAgent: Could not instantiate codec");
- throw VoipLinkException ("Could not instantiate codec");
+ delete call;
+ call = NULL;
+ throw VoipLinkException ("Could not instantiate codec for early media");
}
try {
@@ -697,22 +718,19 @@ Call *SIPVoIPLink::newOutgoingCall (const CallID& id, const std::string& toUrl)
_info ("UserAgent: Start audio rtp session");
call->getAudioRtp()->start (static_cast<AudioCodec *>(audiocodec));
} catch (...) {
- _error ("UserAgent: Error: Failed to create rtp thread from newOutGoingCall");
+ throw VoipLinkException ("Could start rtp session for early media");
}
// init file name according to peer phone number
call->initRecFileName (toUrl);
- _debug ("UserAgent: Try to make a call to: %s with call ID: %s", toUrl.data(), id.data());
-
// Building the local SDP offer
call->getLocalSDP()->setLocalIP (addrSdp);
status = call->getLocalSDP()->createOffer (account->getActiveCodecs ());
-
if (status != PJ_SUCCESS) {
delete call;
call = NULL;
- return call;
+ throw VoipLinkException ("Could not create local sdp offer for new call");
}
if (SIPOutgoingInvite (call)) {
@@ -722,9 +740,10 @@ Call *SIPVoIPLink::newOutgoingCall (const CallID& id, const std::string& toUrl)
} else {
delete call;
call = NULL;
+ throw VoipLinkException("Could not send outgoing INVITE request for new call");
}
- return call;
+ return call;
}
bool
@@ -1042,13 +1061,15 @@ SIPVoIPLink::sendTextMessage (sfl::InstantMessaging *module, const std::string&
return status;
}
-int SIPVoIPLink::SIPSessionReinvite (SIPCall *call)
+int SIPSessionReinvite (SIPCall *call)
{
pj_status_t status;
pjsip_tx_data *tdata;
pjmedia_sdp_session *local_sdp;
+ _debug("UserAgent: Sending re-INVITE request");
+
if (call == NULL) {
_error ("UserAgent: Error: Call is NULL in session reinvite");
return !PJ_SUCCESS;
@@ -1086,21 +1107,20 @@ SIPVoIPLink::transfer (const CallID& id, const std::string& to)
pj_status_t status;
SIPCall *call = getSIPCall (id);
+ if (!call) {
+ _error ("UserAgent: Error: Call doesn't exist");
+ return false;
+ }
+
call->stopRecording();
AccountID account_id = Manager::instance().getAccountFromCall (id);
SIPAccount *account = dynamic_cast<SIPAccount *> (Manager::instance().getAccount (account_id));
-
if (!account) {
_error ("UserAgent: Error: Transfer account is null. Returning.");
return false;
}
- if (!call) {
- _error ("UserAgent: Error: Call doesn't exist");
- return false;
- }
-
std::string dest;
pj_str_t pjDest;
@@ -1114,7 +1134,7 @@ SIPVoIPLink::transfer (const CallID& id, const std::string& to)
/* Create xfer client subscription. */
pj_bzero (&xfer_cb, sizeof (xfer_cb));
- xfer_cb.on_evsub_state = &transfer_function_cb;
+ xfer_cb.on_evsub_state = &transfer_client_cb;
status = pjsip_xfer_create_uac (call->getInvSession()->dlg, &xfer_cb, &sub);
@@ -1206,7 +1226,7 @@ bool SIPVoIPLink::attendedTransfer(const CallID& transferId, const CallID& targe
/* Create xfer client subscription. */
pj_bzero (&xfer_cb, sizeof (xfer_cb));
- xfer_cb.on_evsub_state = &transfer_function_cb;
+ xfer_cb.on_evsub_state = &transfer_client_cb;
status = pjsip_xfer_create_uac (transferCall->getInvSession()->dlg, &xfer_cb, &sub);
@@ -1509,9 +1529,11 @@ SIPVoIPLink::SIPStartCall (SIPCall* call, const std::string& subject UNUSED)
return false;
}
+
// Creates URI
std::string fromUri = account->getFromUri();
std::string toUri = call->getPeerNumber(); // expecting a fully well formed sip uri
+ _debug("------------------------------------------ toUri: %s", toUri.c_str());
std::string address = findLocalAddressFromUri (toUri, account->getAccountTransport ());
int port = findLocalPortFromUri (toUri, account->getAccountTransport ());
@@ -1751,7 +1773,10 @@ bool SIPVoIPLink::SIPNewIpToIpCall (const CallID& id, const std::string& to)
// Building the local SDP offer
call->getLocalSDP()->setLocalIP (addrSdp);
- call->getLocalSDP()->createOffer (account->getActiveCodecs ());
+ status = call->getLocalSDP()->createOffer (account->getActiveCodecs ());
+ if(status != PJ_SUCCESS) {
+ _error("UserAgent: Failed to create local offer\n");
+ }
// Init TLS transport if enabled
if (account->isTlsEnabled()) {
@@ -3166,6 +3191,8 @@ void invite_session_state_changed_cb (pjsip_inv_session *inv, pjsip_event *e)
pjsip_tx_data *tdata;
pj_status_t status;
+ _debug(" ----------------------------------------------------------- Xfer NOTIFY!!!!!!\n");
+
status = pjsip_xfer_notify (call->getXferSub(), ev_state, st_code, NULL, &tdata);
if (status != PJ_SUCCESS) {
@@ -3492,6 +3519,7 @@ void transaction_state_changed_cb (pjsip_inv_session *inv UNUSED, pjsip_transact
if (tsx->role==PJSIP_ROLE_UAS && tsx->state==PJSIP_TSX_STATE_TRYING &&
pjsip_method_cmp (&tsx->method, &pjsip_refer_method) ==0) {
/** Handle the refer method **/
+ _debug("-------------------------------------------------------------- Handle REFER method");
onCallTransfered (inv, e->body.tsx_state.src.rdata);
} else if (tsx->role==PJSIP_ROLE_UAS && tsx->state==PJSIP_TSX_STATE_TRYING) {
@@ -3718,6 +3746,8 @@ transaction_request_cb (pjsip_rx_data *rdata)
pjsip_inv_session *inv;
pjmedia_sdp_session *r_sdp;
+ _debug("------------------------------------------------------------------- Transaction Request\n");
+
_info ("UserAgent: Transaction REQUEST received using transport: %s %s (refcnt=%d)",
rdata->tp_info.transport->obj_name,
rdata->tp_info.transport->info,
@@ -3970,7 +4000,15 @@ transaction_request_cb (pjsip_rx_data *rdata)
status = call->getLocalSDP()->recieveOffer (r_sdp, account->getActiveCodecs ());
+ if (status!=PJ_SUCCESS) {
+ delete call;
+ call = NULL;
+ _warn ("UserAgent: fail in receiving initial offer");
+ pjsip_endpt_respond_stateless (_endpt, rdata, PJSIP_SC_INTERNAL_SERVER_ERROR, NULL, NULL, NULL);
+ return false;
+ }
+ // Init default codec for early media session
sfl::Codec* audiocodec = Manager::instance().getCodecDescriptorMap().instantiateCodec (PAYLOAD_CODEC_ULAW);
// Init audio rtp session
@@ -3981,14 +4019,6 @@ transaction_request_cb (pjsip_rx_data *rdata)
_warn ("UserAgent: Error: Failed to create rtp thread from answer");
}
- if (status!=PJ_SUCCESS) {
- delete call;
- call = NULL;
- _warn ("UserAgent: fail in receiving initial offer");
- pjsip_endpt_respond_stateless (_endpt, rdata, PJSIP_SC_INTERNAL_SERVER_ERROR,
- NULL, NULL, NULL);
- return false;
- }
/* Create the local dialog (UAS) */
status = pjsip_dlg_create_uas (pjsip_ua_instance(), rdata, NULL, &dialog);
@@ -4081,6 +4111,8 @@ transaction_request_cb (pjsip_rx_data *rdata)
pj_bool_t transaction_response_cb (pjsip_rx_data *rdata)
{
+ _debug("----------------------------------------------------------- Transaction response");
+
_info ("UserAgent: Transaction response using transport: %s %s (refcnt=%d)",
rdata->tp_info.transport->obj_name,
rdata->tp_info.transport->info,
@@ -4125,7 +4157,7 @@ void onCallTransfered (pjsip_inv_session *inv, pjsip_rx_data *rdata)
pj_status_t status;
pjsip_tx_data *tdata;
- SIPCall *existing_call;
+ SIPCall *currentCall;
const pj_str_t str_refer_to = { (char*) "Refer-To", 8};
const pj_str_t str_refer_sub = { (char*) "Refer-Sub", 9 };
const pj_str_t str_ref_by = { (char*) "Referred-By", 11 };
@@ -4134,16 +4166,21 @@ void onCallTransfered (pjsip_inv_session *inv, pjsip_rx_data *rdata)
pjsip_hdr *ref_by_hdr;
pj_bool_t no_refer_sub = PJ_FALSE;
char *uri;
- std::string tmp;
+ std::string sipUri;
pjsip_status_code code;
pjsip_evsub *sub;
- existing_call = (SIPCall *) inv->mod_data[_mod_ua.id];
+ _debug("----------------------------------------------------------- On Call Transfered called\n");
+
+ currentCall = (SIPCall *) inv->mod_data[_mod_ua.id];
+ if (currentCall == NULL) {
+ _debug ("UserAgent: Call doesn't exist (%s, %s)", __FILE__, __LINE__);
+ return;
+ }
/* Find the Refer-To header */
refer_to = (pjsip_generic_string_hdr*)
pjsip_msg_find_hdr_by_name (rdata->msg_info.msg, &str_refer_to, NULL);
-
if (refer_to == NULL) {
/* Invalid Request.
* No Refer-To header!
@@ -4156,7 +4193,6 @@ void onCallTransfered (pjsip_inv_session *inv, pjsip_rx_data *rdata)
/* Find optional Refer-Sub header */
refer_sub = (pjsip_generic_string_hdr*)
pjsip_msg_find_hdr_by_name (rdata->msg_info.msg, &str_refer_sub, NULL);
-
if (refer_sub) {
if (!pj_strnicmp2 (&refer_sub->hvalue, "true", 4) ==0)
no_refer_sub = PJ_TRUE;
@@ -4172,10 +4208,10 @@ void onCallTransfered (pjsip_inv_session *inv, pjsip_rx_data *rdata)
code = PJSIP_SC_ACCEPTED;
_debug ("UserAgent: Call to %.*s is being transfered to %.*s",
- (int) inv->dlg->remote.info_str.slen,
- inv->dlg->remote.info_str.ptr,
- (int) refer_to->hvalue.slen,
- refer_to->hvalue.ptr);
+ (int) inv->dlg->remote.info_str.slen,
+ inv->dlg->remote.info_str.ptr,
+ (int) refer_to->hvalue.slen,
+ refer_to->hvalue.ptr);
if (no_refer_sub) {
/*
@@ -4189,7 +4225,7 @@ void onCallTransfered (pjsip_inv_session *inv, pjsip_rx_data *rdata)
&tdata);
if (status != PJ_SUCCESS) {
- _debug ("UserAgent: Unable to create 2xx response to REFER -- %d", status);
+ _debug ("UserAgent: Unable to create 2xx response to REFER reques -- %d", status);
return;
}
@@ -4206,7 +4242,7 @@ void onCallTransfered (pjsip_inv_session *inv, pjsip_rx_data *rdata)
tdata);
if (status != PJ_SUCCESS) {
- _debug ("UserAgent: Unable to create 2xx response to REFER -- %d", status);
+ _debug ("UserAgent: Unable to create 2xx response to REFER request -- %d", status);
return;
}
@@ -4222,7 +4258,7 @@ void onCallTransfered (pjsip_inv_session *inv, pjsip_rx_data *rdata)
pj_bzero (&xfer_cb, sizeof (xfer_cb));
xfer_cb.on_evsub_state = &transfer_server_cb;
- /* Init addiTHIS_FILE, THIS_FILE, tional header list to be sent with REFER response */
+ /* Init addional header list to be sent with REFER response */
pj_list_init (&hdr_list);
/* Create transferee event subscription */
@@ -4280,18 +4316,17 @@ void onCallTransfered (pjsip_inv_session *inv, pjsip_rx_data *rdata)
uri[refer_to->hvalue.slen] = '\0';
/* Now make the outgoing call. */
- tmp = std::string (uri);
-
- if (existing_call == NULL) {
- _debug ("UserAgent: Call doesn't exist!");
- return;
- }
+ sipUri = std::string (uri);
- AccountID accId = Manager::instance().getAccountFromCall (existing_call->getCallId());
+ CallID currentCallId = currentCall->getCallId();
+ AccountID accId = Manager::instance().getAccountFromCall (currentCallId);
CallID newCallId = Manager::instance().getNewCallID();
- if (!Manager::instance().outgoingCall (accId, newCallId, tmp)) {
+ // Hangup currently transfered call
+ // Manager::instance().hangupCall(currentCallId);
+
+ if (!Manager::instance().outgoingCall (accId, newCallId, sipUri)) {
/* Notify xferer about the error (if we have subscription) */
if (sub) {
@@ -4314,9 +4349,13 @@ void onCallTransfered (pjsip_inv_session *inv, pjsip_rx_data *rdata)
return;
}
- SIPCall* newCall = 0;
+ SIPCall* newCall = NULL;
SIPVoIPLink *link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink (accId));
+ if(link == NULL) {
+ _debug("UserAgent: Error could not retreive voip link from call");
+ return;
+ }
if (link) {
newCall = dynamic_cast<SIPCall *> (link->getCall (newCallId));
@@ -4335,14 +4374,13 @@ void onCallTransfered (pjsip_inv_session *inv, pjsip_rx_data *rdata)
newCall->setXferSub (sub);
/* Put the invite_data in the subscription. */
- pjsip_evsub_set_mod_data (sub, _mod_ua.id,
- newCall);
+ pjsip_evsub_set_mod_data (sub, _mod_ua.id, newCall);
}
}
-void transfer_function_cb (pjsip_evsub *sub, pjsip_event *event)
+void transfer_client_cb (pjsip_evsub *sub, pjsip_event *event)
{
@@ -4362,7 +4400,7 @@ void transfer_function_cb (pjsip_evsub *sub, pjsip_event *event)
* On incoming NOTIFY, notify application about call transfer progress.
*/
else if (pjsip_evsub_get_state (sub) == PJSIP_EVSUB_STATE_ACTIVE ||
- pjsip_evsub_get_state (sub) == PJSIP_EVSUB_STATE_TERMINATED) {
+ pjsip_evsub_get_state (sub) == PJSIP_EVSUB_STATE_TERMINATED) {
pjsip_msg *msg;
pjsip_msg_body *body;
@@ -4435,13 +4473,11 @@ void transfer_function_cb (pjsip_evsub *sub, pjsip_event *event)
std::map<std::string, CallID>::iterator it = transferCallID.find (transferID);
CallID cid = it->second;
SIPCall *call = dynamic_cast<SIPCall *> (link->getCall (cid));
-
if (!call) {
_warn ("UserAgent: Call with id %s doesn't exit!", cid.c_str());
return;
}
-
/* Notify application */
is_last = (pjsip_evsub_get_state (sub) ==PJSIP_EVSUB_STATE_TERMINATED);
@@ -4489,14 +4525,41 @@ void transfer_server_cb (pjsip_evsub *sub, pjsip_event *event)
* When subscription is terminated, clear the xfer_sub member of
* the inv_data.
*/
+ switch (pjsip_evsub_get_state (sub)) {
+ case PJSIP_EVSUB_STATE_NULL:
+ _debug("--------------------------------------------- Transfer server cb PJSIP_EVSUB_STATE_NULL\n");
+ break;
+ case PJSIP_EVSUB_STATE_SENT:
+ _debug("--------------------------------------------- Transfer server cb PJSIP_EVSUB_STATE_SENT\n");
+ break;
+ case PJSIP_EVSUB_STATE_ACCEPTED:
+ _debug("--------------------------------------------- Transfer server cb PJSIP_EVSUB_STATE_ACCEPTED\n");
+ break;
+ case PJSIP_EVSUB_STATE_PENDING:
+ _debug("--------------------------------------------- Transfer server cb PJSIP_EVSUB_STATE_PENDING\n");
+ break;
+ case PJSIP_EVSUB_STATE_ACTIVE:
+ _debug("--------------------------------------------- Transfer server cb PJSIP_EVSUB_STATE_ACTIVE\n");
+ break;
+ case PJSIP_EVSUB_STATE_TERMINATED:
+ _debug("--------------------------------------------- Transfer server cb PJSIP_EVSUB_STATE_TERMINATED\n");
+ break;
+ case PJSIP_EVSUB_STATE_UNKNOWN:
+ _debug("--------------------------------------------- Transfer server cb PJSIP_EVSUB_STATE_UNKNOWN\n");
+ break;
+ default:
+ break;
+ }
if (pjsip_evsub_get_state (sub) == PJSIP_EVSUB_STATE_TERMINATED) {
SIPCall *call;
call = (SIPCall*) pjsip_evsub_get_mod_data (sub, _mod_ua.id);
- if (!call)
+ if (!call) {
+ _debug("UserAgent: Could not find subscription data");
return;
+ }
pjsip_evsub_set_mod_data (sub, _mod_ua.id, NULL);
diff --git a/sflphone-common/src/sip/sipvoiplink.h b/sflphone-common/src/sip/sipvoiplink.h
index abcd552abb029d7ac9d490e089765125cc649c1d..3bad169dbd6c103f50459ee7c07614cdfb945cad 100644
--- a/sflphone-common/src/sip/sipvoiplink.h
+++ b/sflphone-common/src/sip/sipvoiplink.h
@@ -306,13 +306,6 @@ class SIPVoIPLink : public VoIPLink
*/
void SIPHandleReinvite (SIPCall *call);
- /**
- * Send a reINVITE inside an active dialog to modify its state
- * Local SDP session should be modified before calling this method
- * @param sip call
- */
- int SIPSessionReinvite (SIPCall *call);
-
pj_caching_pool *getMemoryPoolFactory();
/**