diff --git a/sflphone-common/src/iax/iaxvoiplink.cpp b/sflphone-common/src/iax/iaxvoiplink.cpp
index 2c3a0b1a7a9707b51178657ab35f301d06a049a7..280d15b082acbd86452439b79105ca2a3d9c6816 100644
--- a/sflphone-common/src/iax/iaxvoiplink.cpp
+++ b/sflphone-common/src/iax/iaxvoiplink.cpp
@@ -536,25 +536,25 @@ IAXVoIPLink::answer (const CallID& id) throw (VoipLinkException)
}
bool
-IAXVoIPLink::hangup (const CallID& id)
+IAXVoIPLink::hangup (const CallID& id) throw (VoipLinkException)
{
_debug ("IAXVoIPLink: Hangup");
+
IAXCall* call = getIAXCall (id);
+ if(call == NULL) {
+ throw VoipLinkException("Could not find call");
+ }
+
std::string reason = "Dumped Call";
CHK_VALID_CALL;
audiolayer->getMainBuffer()->unBindAll (call->getCallId());
_mutexIAX.enterMutex();
-
iax_hangup (call->getSession(), (char*) reason.c_str());
_mutexIAX.leaveMutex();
- call->setSession (NULL);
- if (Manager::instance().isCurrentCall (id)) {
- // stop audio
- // audiolayer->stopStream();
- }
+ call->setSession (NULL);
removeCall (id);
return true;
@@ -562,25 +562,22 @@ IAXVoIPLink::hangup (const CallID& id)
bool
-IAXVoIPLink::peerHungup (const CallID& id)
+IAXVoIPLink::peerHungup (const CallID& id) throw (VoipLinkException)
{
_debug ("IAXVoIPLink: Peer hung up");
+
IAXCall* call = getIAXCall (id);
+ if(call == NULL) {
+ throw VoipLinkException("Could not find call");
+ }
+
std::string reason = "Dumped Call";
CHK_VALID_CALL;
audiolayer->getMainBuffer()->unBindAll (call->getCallId());
- _mutexIAX.enterMutex();
-
- _mutexIAX.leaveMutex();
call->setSession (NULL);
- if (Manager::instance().isCurrentCall (id)) {
- // stop audio
- // audiolayer->stopStream();
- }
-
removeCall (id);
return true;
}
@@ -588,9 +585,12 @@ IAXVoIPLink::peerHungup (const CallID& id)
bool
-IAXVoIPLink::onhold (const CallID& id)
+IAXVoIPLink::onhold (const CallID& id) throw (VoipLinkException)
{
IAXCall* call = getIAXCall (id);
+ if(call == NULL) {
+ throw VoipLinkException("Call does not exist");
+ }
CHK_VALID_CALL;
@@ -607,7 +607,7 @@ IAXVoIPLink::onhold (const CallID& id)
}
bool
-IAXVoIPLink::offhold (const CallID& id)
+IAXVoIPLink::offhold (const CallID& id) throw (VoipLinkException)
{
IAXCall* call = getIAXCall (id);
@@ -625,7 +625,7 @@ IAXVoIPLink::offhold (const CallID& id)
}
bool
-IAXVoIPLink::transfer (const CallID& id, const std::string& to)
+IAXVoIPLink::transfer (const CallID& id, const std::string& to) throw (VoipLinkException)
{
IAXCall* call = getIAXCall (id);
diff --git a/sflphone-common/src/iax/iaxvoiplink.h b/sflphone-common/src/iax/iaxvoiplink.h
index 7efba3be4088112229be528a00d96cc175dd4224..33859a136f1fbbc44a23ae53b5f697739a1dc0b0 100644
--- a/sflphone-common/src/iax/iaxvoiplink.h
+++ b/sflphone-common/src/iax/iaxvoiplink.h
@@ -128,7 +128,7 @@ class IAXVoIPLink : public VoIPLink
* @return bool true on success
* false otherwise
*/
- virtual bool hangup (const CallID& id);
+ virtual bool hangup (const CallID& id) throw (VoipLinkException);
/**
* Peer Hungup a call
@@ -136,7 +136,7 @@ class IAXVoIPLink : public VoIPLink
* @return bool true on success
* false otherwise
*/
- virtual bool peerHungup (const CallID& id);
+ virtual bool peerHungup (const CallID& id) throw (VoipLinkException);
/**
* Cancel a call
@@ -144,7 +144,7 @@ class IAXVoIPLink : public VoIPLink
* @return bool true on success
* false otherwise
*/
- virtual bool cancel (const CallID& id UNUSED) {
+ virtual bool cancel (const CallID& id UNUSED) throw (VoipLinkException){
return false;
}
@@ -154,7 +154,7 @@ class IAXVoIPLink : public VoIPLink
* @return bool true on success
* false otherwise
*/
- virtual bool onhold (const CallID& id);
+ virtual bool onhold (const CallID& id) throw (VoipLinkException);
/**
* Put a call off hold
@@ -162,7 +162,7 @@ class IAXVoIPLink : public VoIPLink
* @return bool true on success
* false otherwise
*/
- virtual bool offhold (const CallID& id);
+ virtual bool offhold (const CallID& id) throw (VoipLinkException);
/**
* Transfer a call
@@ -171,7 +171,7 @@ class IAXVoIPLink : public VoIPLink
* @return bool true on success
* false otherwise
*/
- virtual bool transfer (const CallID& id, const std::string& to);
+ virtual bool transfer (const CallID& id, const std::string& to) throw (VoipLinkException);
/**
* Perform attended transfer
diff --git a/sflphone-common/src/sip/sipvoiplink.cpp b/sflphone-common/src/sip/sipvoiplink.cpp
index da71a56c58e469296969612c6f848edf16633d30..52f1444080f0b4cbf8a58a09abb434cd8ec514a5 100644
--- a/sflphone-common/src/sip/sipvoiplink.cpp
+++ b/sflphone-common/src/sip/sipvoiplink.cpp
@@ -650,9 +650,6 @@ Call *SIPVoIPLink::newOutgoingCall (const CallID& id, const std::string& toUrl)
pj_status_t status;
std::string localAddr, addrSdp;
- // _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) {
@@ -718,7 +715,7 @@ Call *SIPVoIPLink::newOutgoingCall (const CallID& id, const std::string& toUrl)
_info ("UserAgent: Start audio rtp session");
call->getAudioRtp()->start (static_cast<AudioCodec *>(audiocodec));
} catch (...) {
- throw VoipLinkException ("Could start rtp session for early media");
+ throw VoipLinkException ("Could not start rtp session for early media");
}
// init file name according to peer phone number
@@ -756,7 +753,6 @@ SIPVoIPLink::answer (const CallID& id) throw (VoipLinkException)
_debug ("UserAgent: Answering call %s", id.c_str());
SIPCall *call = getSIPCall (id);
-
if (call==NULL) {
throw VoipLinkException("Call is NULL while answering");
}
@@ -789,9 +785,16 @@ SIPVoIPLink::answer (const CallID& id) throw (VoipLinkException)
// Terminate the call
_debug ("UserAgent: SDP Negociation failed, terminate call %s ", call->getCallId().c_str());
- if (call->getAudioRtp()) {
+ if(call->getAudioRtp()) {
+ throw VoipLinkException("No audio rtp session for this call");
+ }
+
+ try {
call->getAudioRtp()->stop ();
}
+ catch(...) {
+ throw VoipLinkException("Could not stop rtp session");
+ }
removeCall (call->getCallId());
@@ -800,53 +803,59 @@ SIPVoIPLink::answer (const CallID& id) throw (VoipLinkException)
}
bool
-SIPVoIPLink::hangup (const CallID& id)
+SIPVoIPLink::hangup (const CallID& id) throw (VoipLinkException)
{
pj_status_t status;
pjsip_tx_data *tdata = NULL;
SIPCall* call = getSIPCall (id);
-
+ if (call == NULL) {
+ throw VoipLinkException("Call is NULL while hanging up");
+ }
AccountID account_id = Manager::instance().getAccountFromCall (id);
SIPAccount *account = dynamic_cast<SIPAccount *> (Manager::instance().getAccount (account_id));
-
- if (call == NULL) {
- _debug ("! SIP Error: Call doesn't exist");
- return false;
+ if(account == NULL) {
+ throw VoipLinkException("Could not find account for this call");
}
+
pjsip_inv_session *inv = call->getInvSession();
+ if(inv == NULL) {
+ throw VoipLinkException("No invite session for this call");
+ }
- // _debug("Some tdata info: %",);
+ // Looks for sip routes
if (! (account->getServiceRoute().empty())) {
pjsip_route_hdr *route_set = createRouteSet(account, inv->pool);
pjsip_dlg_set_route_set (inv->dlg, route_set);
}
// User hangup current call. Notify peer
- status = pjsip_inv_end_session (call->getInvSession(), 404, NULL, &tdata);
-
- if (status != PJ_SUCCESS)
+ status = pjsip_inv_end_session (inv, 404, NULL, &tdata);
+ if (status != PJ_SUCCESS) {
return false;
+ }
-
- if (tdata == NULL)
+ if (tdata == NULL) {
return true;
+ }
- status = pjsip_inv_send_msg (call->getInvSession(), tdata);
-
+ status = pjsip_inv_send_msg (inv, tdata);
if (status != PJ_SUCCESS)
return false;
- // Make sure the pointer is NULL in callbacks
- call->getInvSession()->mod_data[getModId()] = NULL;
-
+ // Make sure user data is NULL in callbacks
+ inv->mod_data[getModId()] = NULL;
// Release RTP thread
- if (Manager::instance().isCurrentCall (id)) {
- _debug ("* SIP Info: Stopping AudioRTP for hangup");
- call->getAudioRtp()->stop();
+ try {
+ if (Manager::instance().isCurrentCall (id)) {
+ call->getAudioRtp()->stop();
+ }
+ }
+ catch(...) {
+ throw VoipLinkException("Could not stop audio rtp session");
}
removeCall (id);
@@ -855,7 +864,7 @@ SIPVoIPLink::hangup (const CallID& id)
}
bool
-SIPVoIPLink::peerHungup (const CallID& id)
+SIPVoIPLink::peerHungup (const CallID& id) throw (VoipLinkException)
{
pj_status_t status;
pjsip_tx_data *tdata = NULL;
@@ -864,15 +873,12 @@ SIPVoIPLink::peerHungup (const CallID& id)
_info ("UserAgent: Peer hungup");
call = getSIPCall (id);
-
- if (call==0) {
- _warn ("UserAgent: Call doesn't exist");
- return false;
+ if (call == NULL) {
+ throw VoipLinkException("Call does not exist");
}
// User hangup current call. Notify peer
status = pjsip_inv_end_session (call->getInvSession(), 404, NULL, &tdata);
-
if (status != PJ_SUCCESS)
return false;
@@ -884,14 +890,18 @@ SIPVoIPLink::peerHungup (const CallID& id)
if (status != PJ_SUCCESS)
return false;
- // Make sure the pointer is NULL in callbacks
+ // Make sure user data is NULL in callbacks
call->getInvSession()->mod_data[getModId() ] = NULL;
-
// Release RTP thread
- if (Manager::instance().isCurrentCall (id)) {
- _debug ("UserAgent: Stopping AudioRTP for hangup");
- call->getAudioRtp()->stop();
+ try {
+ if (Manager::instance().isCurrentCall (id)) {
+ _debug ("UserAgent: Stopping AudioRTP for hangup");
+ call->getAudioRtp()->stop();
+ }
+ }
+ catch(...) {
+ throw VoipLinkException("Could not stop audio rtp session");
}
removeCall (id);
@@ -900,15 +910,13 @@ SIPVoIPLink::peerHungup (const CallID& id)
}
bool
-SIPVoIPLink::cancel (const CallID& id)
+SIPVoIPLink::cancel (const CallID& id) throw (VoipLinkException)
{
_info ("UserAgent: Cancel call %s", id.c_str());
SIPCall* call = getSIPCall (id);
-
if (!call) {
- _warn ("UserAgent: Error: Call doesn't exist");
- return false;
+ throw VoipLinkException("Call does not exist");
}
removeCall (id);
@@ -918,7 +926,7 @@ SIPVoIPLink::cancel (const CallID& id)
bool
-SIPVoIPLink::onhold (const CallID& id)
+SIPVoIPLink::onhold (const CallID& id) throw (VoipLinkException)
{
Sdp *sdpSession;
pj_status_t status;
@@ -927,24 +935,30 @@ SIPVoIPLink::onhold (const CallID& id)
call = getSIPCall (id);
if (call == NULL) {
- _debug ("UserAgent: Error: call doesn't exist in onhold action");
- return false;
+ throw VoipLinkException("Could not find call");
}
// Stop sound
call->setAudioStart (false);
call->setState (Call::Hold);
- call->getAudioRtp()->stop();
+
+ try {
+ call->getAudioRtp()->stop();
+ }
+ catch (...) {
+ throw VoipLinkException("Could not stop audio rtp session");
+ }
_debug ("UserAgent: Stopping RTP session for on hold action");
- if ( (sdpSession = call->getLocalSDP()) == NULL) {
- _debug ("UserAgent: Error: Unable to find local sdp");
- return false;
+ sdpSession = call->getLocalSDP();
+ if (sdpSession == NULL) {
+ throw VoipLinkException("Could not find sdp session");
}
sdpSession->removeAttributeFromLocalAudioMedia("sendrecv");
sdpSession->removeAttributeFromLocalAudioMedia("sendonly");
+
sdpSession->addAttributeToLocalAudioMedia("sendonly");
// Create re-INVITE with new offer
@@ -958,7 +972,7 @@ SIPVoIPLink::onhold (const CallID& id)
}
bool
-SIPVoIPLink::offhold (const CallID& id)
+SIPVoIPLink::offhold (const CallID& id) throw (VoipLinkException)
{
Sdp *sdpSession;
pj_status_t status;
@@ -968,21 +982,20 @@ SIPVoIPLink::offhold (const CallID& id)
call = getSIPCall (id);
- if (call==0) {
- _debug ("! SIP Error: Call doesn't exist");
- return false;
+ if (call == NULL) {
+ throw VoipLinkException("Could not find call");
}
- if ( (sdpSession = call->getLocalSDP()) == NULL) {
- _debug ("UserAgent: Error: Unable to find local sdp");
- return false;
+ sdpSession = call->getLocalSDP();
+ if (sdpSession == NULL) {
+ throw VoipLinkException("Could not find sdp session");
}
// Retreive previously selected codec
sfl::Codec *sessionMedia = sdpSession->getSessionMedia();
- if (!sessionMedia) {
- return false;
+ if (sessionMedia == NULL) {
+ throw VoipLinkException("Could not find session media");
}
// Get PayloadType for this codec
@@ -990,19 +1003,19 @@ SIPVoIPLink::offhold (const CallID& id)
_debug ("UserAgent: Payload from session media %d", pl);
- try {
- // Create a new instance for this codec
- sfl::Codec* audiocodec = Manager::instance().getCodecDescriptorMap().instantiateCodec (pl);
-
- if (audiocodec == NULL)
- _error ("UserAgent: No audiocodec found");
+ // Create a new instance for this codec
+ sfl::Codec* audiocodec = Manager::instance().getCodecDescriptorMap().instantiateCodec (pl);
+ if (audiocodec == NULL) {
+ throw VoipLinkException("Could not instantiate codec");
+ }
+ try {
call->getAudioRtp()->initAudioRtpConfig (call);
call->getAudioRtp()->initAudioRtpSession (call);
call->getAudioRtp()->start (static_cast<AudioCodec *>(audiocodec));
} catch (...) {
- _debug ("! SIP Failure: Unable to create RTP Session (%s:%d)", __FILE__, __LINE__);
+ throw VoipLinkException("Could not create audio rtp session");
}
sdpSession->removeAttributeFromLocalAudioMedia("sendrecv");
@@ -1012,7 +1025,6 @@ SIPVoIPLink::offhold (const CallID& id)
/* Create re-INVITE with new offer */
status = SIPSessionReinvite (call);
-
if (status != PJ_SUCCESS) {
return false;
}
@@ -1095,7 +1107,7 @@ int SIPSessionReinvite (SIPCall *call)
}
bool
-SIPVoIPLink::transfer (const CallID& id, const std::string& to)
+SIPVoIPLink::transfer (const CallID& id, const std::string& to) throw (VoipLinkException)
{
std::string tmp_to;
@@ -1106,28 +1118,26 @@ 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;
+ if (call == NULL) {
+ throw VoipLinkException("Could not find call");
}
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 (account == NULL) {
+ throw VoipLinkException("Could not find account");
}
std::string dest;
-
pj_str_t pjDest;
if (to.find ("@") == std::string::npos) {
dest = account->getToUri (to);
pj_cstr (&pjDest, dest.c_str());
}
+
_info ("UserAgent: Transfering to %s", dest.c_str());
/* Create xfer client subscription. */
@@ -1135,10 +1145,8 @@ SIPVoIPLink::transfer (const CallID& id, const std::string& to)
xfer_cb.on_evsub_state = &transfer_client_cb;
status = pjsip_xfer_create_uac (call->getInvSession()->dlg, &xfer_cb, &sub);
-
if (status != PJ_SUCCESS) {
- _warn ("UserAgent: Unable to create xfer -- %d", status);
- return false;
+ throw VoipLinkException("Could not create xfer request");
}
/* Associate this voiplink of call with the client subscription
@@ -1152,24 +1160,18 @@ SIPVoIPLink::transfer (const CallID& id, const std::string& to)
* Create REFER request.
*/
status = pjsip_xfer_initiate (sub, &pjDest, &tdata);
-
if (status != PJ_SUCCESS) {
- _error ("UserAgent: Unable to create REFER request -- %d", status);
- return false;
+ throw VoipLinkException("Could not create REFER request");
}
// Put SIP call id in map in order to retrieve call during transfer callback
std::string callidtransfer (call->getInvSession()->dlg->call_id->id.ptr, call->getInvSession()->dlg->call_id->id.slen);
- _debug ("%s", callidtransfer.c_str());
transferCallID.insert (std::pair<std::string, CallID> (callidtransfer, call->getCallId()));
-
/* Send. */
status = pjsip_xfer_send_request (sub, tdata);
-
if (status != PJ_SUCCESS) {
- _error ("UserAgent: Unable to send REFER request -- %d", status);
- return false;
+ throw VoipLinkException("Could not send xfer request");
}
return true;
@@ -4167,7 +4169,7 @@ void onCallTransfered (pjsip_inv_session *inv, pjsip_rx_data *rdata)
char *uri;
std::string sipUri;
pjsip_status_code code;
- pjsip_evsub *sub;
+ pjsip_evsub *sub = NULL;
currentCall = (SIPCall *) inv->mod_data[_mod_ua.id];
if (currentCall == NULL) {
@@ -4346,10 +4348,10 @@ void onCallTransfered (pjsip_inv_session *inv, pjsip_rx_data *rdata)
return;
}
- SIPCall* sipCall = dynamic_cast<SIPCall *>(newCall);
-
Manager::instance().hangupCall(currentCall->getCallId());
+// SIPCall* sipCall = dynamic_cast<SIPCall *>(newCall);
+
// SIPVoIPLink *link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink (accId));
// if(link == NULL) {
// _debug("UserAgent: Error could not retreive voip link from call");
diff --git a/sflphone-common/src/sip/sipvoiplink.h b/sflphone-common/src/sip/sipvoiplink.h
index ecb72e6ce82bd95240a081116db40d846026f7ba..ec301e7a5442357851a29f62e05b393b976f1880 100644
--- a/sflphone-common/src/sip/sipvoiplink.h
+++ b/sflphone-common/src/sip/sipvoiplink.h
@@ -160,7 +160,7 @@ class SIPVoIPLink : public VoIPLink
/**
* Answer the call
* @param id The call identifier
- * @return int True on success
+ * @return True on success
*/
virtual bool answer (const CallID& id) throw (VoipLinkException);
@@ -169,35 +169,35 @@ class SIPVoIPLink : public VoIPLink
* @param id The call identifier
* @return bool True on success
*/
- virtual bool hangup (const CallID& id);
+ virtual bool hangup (const CallID& id) throw (VoipLinkException);
/**
* Hang up the call
* @param id The call identifier
* @return bool True on success
*/
- virtual bool peerHungup (const CallID& id);
+ virtual bool peerHungup (const CallID& id) throw (VoipLinkException);
/**
* Cancel the call
* @param id The call identifier
* @return bool True on success
*/
- virtual bool cancel (const CallID& id);
+ virtual bool cancel (const CallID& id) throw (VoipLinkException);
/**
* Put the call on hold
* @param id The call identifier
* @return bool True on success
*/
- virtual bool onhold (const CallID& id);
+ virtual bool onhold (const CallID& id) throw (VoipLinkException);
/**
* Put the call off hold
* @param id The call identifier
* @return bool True on success
*/
- virtual bool offhold (const CallID& id);
+ virtual bool offhold (const CallID& id) throw (VoipLinkException);
/**
* Transfer the call
@@ -205,7 +205,7 @@ class SIPVoIPLink : public VoIPLink
* @param to The recipient of the transfer
* @return bool True on success
*/
- virtual bool transfer (const CallID& id, const std::string& to);
+ virtual bool transfer (const CallID& id, const std::string& to) throw (VoipLinkException);
/**
* Attended transfer
diff --git a/sflphone-common/src/voiplink.h b/sflphone-common/src/voiplink.h
index 2b1cefd6b9909f7f293452b797c25cffa38faa4e..9ff3791bf8a40f08858b4965d842e054671bceb9 100644
--- a/sflphone-common/src/voiplink.h
+++ b/sflphone-common/src/voiplink.h
@@ -141,35 +141,35 @@ class VoIPLink
* @param id The call identifier
* @return bool True on success
*/
- virtual bool hangup (const CallID& id) = 0;
+ virtual bool hangup (const CallID& id) throw (VoipLinkException) = 0;
/**
* Peer Hung up a call
* @param id The call identifier
* @return bool True on success
*/
- virtual bool peerHungup (const CallID& id) = 0;
+ virtual bool peerHungup (const CallID& id) throw (VoipLinkException) = 0;
/**
* Cancel the call dialing
* @param id The call identifier
* @return bool True on success
*/
- virtual bool cancel (const CallID& id) = 0;
+ virtual bool cancel (const CallID& id) throw (VoipLinkException) = 0;
/**
* Put a call on hold
* @param id The call identifier
* @return bool True on success
*/
- virtual bool onhold (const CallID& id) = 0;
+ virtual bool onhold (const CallID& id) throw (VoipLinkException) = 0;
/**
* Resume a call from hold state
* @param id The call identifier
* @return bool True on success
*/
- virtual bool offhold (const CallID& id) = 0;
+ virtual bool offhold (const CallID& id) throw (VoipLinkException) = 0;
/**
* Transfer a call to specified URI
@@ -177,7 +177,7 @@ class VoIPLink
* @param to The recipient of the call
* @return bool True on success
*/
- virtual bool transfer (const CallID& id, const std::string& to) = 0;
+ virtual bool transfer (const CallID& id, const std::string& to) throw (VoipLinkException) = 0;
/**
* Attended transfer