pjsua2 crash on incoming call
Hello,
With libpjproject 2.5, my application crashes on every incoming
call with ../src/pjsip-ua/sip_inv.c:2321: pjsip_inv_answer: Assertion inv->last_answer' failed.`
For example the example pjsip-apps/src/samples/pjsua2_demo.cpp;
running mainProg1() from that file (adjusted to connect to my
server) and making a call gives me:
17:28:38.112 pjsua_core.c .RX 934 bytes Request msg
INVITE/cseq=19 (rdata0x7f5778009268) from UDP 192.168.1.1:5060:
INVITE sip:...:5060;ob SIP/2.0 Via: SIP/2.0/UDP
192.168.1.1:5060;branch=z9hG4bKF6C9406CE2B74353 From: "Telefon"
sip:...;tag=02C2A2303BAE5A8F To: sip:...:5060;ob Call-ID:
67633F78A1E5FE72@192.168.1.1 CSeq: 19 INVITE Contact:
sip:38CA76357F60C7729329E698EB24B@192.168.1.1 Max-Forwards: 70
Expires: 120 Supported: 100rel,replaces,timer Allow-Events:
telephone-event,refer Allow:
INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH
Content-Type: application/sdp Accept: application/sdp,
multipart/mixed Accept-Encoding: identity Content-Length: 218
v=0
o=user 13656679 13656679 IN IP4 192.168.1.1
s=call
c=IN IP4 192.168.1.1
t=0 0
m=audio 40012 RTP/AVP 8 18 101
a=sendrecv
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtcp:40013
--end msg--
17:28:38.112 pjsua_call.c .Incoming Request msg INVITE/cseq=19
(rdata0x7f5778009268) 17:28:38.112 pjsua_media.c ..Call 0:
initializing media.. 17:28:38.113 pjsua_media.c ...RTP socket
reachable at 192.168.1.8:4000 17:28:38.113 pjsua_media.c ...RTCP
socket reachable at 192.168.1.8:4001 17:28:38.113 pjsua_media.c
...Media index 0 selected for audio call 0
*** Incoming Call: "Telefon" sip:... [NULL]
17:28:38.113 pjsua_call.c ...Answering call 0: code=200 a.out:
../src/pjsip-ua/sip_inv.c:2321: pjsip_inv_answer: Assertion
`inv->last_answer' failed. [1] 19792 abort (core dumped) ./a.out
The call handler ist:
virtual void onIncomingCall(OnIncomingCallParam &iprm)
{
Call *call = new MyCall(*this, iprm.callId);
CallInfo ci = call->getInfo();
CallOpParam prm;
std::cout << "*** Incoming Call: " << ci.remoteUri << " ["
<< ci.stateText << "]" << std::endl;
calls.push_back(call);
prm.statusCode = (pjsip_status_code)200;
call->answer(prm);
}
Any idea what to do here?
thank you
Gabriel
17:28:38.112 pjsua_call.c .Incoming Request msg INVITE/cseq=19 (rdata0x7f5778009268)
17:28:38.112 pjsua_media.c ..Call 0: initializing media..
17:28:38.113 pjsua_media.c ...RTP socket reachable at 192.168.1.8:4000
17:28:38.113 pjsua_media.c ...RTCP socket reachable at 192.168.1.8:4001
17:28:38.113 pjsua_media.c ...Media index 0 selected for audio call 0
*** Incoming Call: "Telefon" sip:... [NULL]
17:28:38.113 pjsua_call.c ...Answering call 0: code=200 a.out:
../src/pjsip-ua/sip_inv.c:2321: pjsip_inv_answer: Assertion
`inv->last_answer' failed. [1] 19792 abort (core dumped) ./a.out
I managed to track it down a little more. It seems to be a bug in
pj::Endpoint. The first answer to a call has to be made by
'pjsip_inv_initial_answer()' function, which is done by pjsua
automatically before calling the 'on_incoming_call' callback of
pj::Endpoint. Before this happens, the media is being initialized (see
log above) by 'pj::Endpoint::on_create_media_transport()' (line 1233).
In version 2.5 this method calls on it's own 'on_incoming_call', but the
call isn't ready to be answered "normally" yet, so we get the crash.
Below the code of on_create_media_transport. Commenting out like shown
solves the problem for me, but I don't know why the code has been added,
so it might not the best solution:
-------------- pjsua2/Endpoint.cpp:1233+ -------------------------
pjmedia_transport*
Endpoint::on_create_media_transport(pjsua_call_id call_id,
unsigned media_idx,
pjmedia_transport *base_tp,
unsigned flags)
{
Call *call = Call::lookup(call_id);
if (!call) {
//pjsua_call in_call = &pjsua_var.calls[call_id];
//if (in_call->incoming_data) {
/ This can happen when there is an incoming call but the
* on_incoming_call() callback hasn't been called. So we need to
* call the callback here.
*/
//on_incoming_call(in_call->acc_id, call_id,
in_call->incoming_data);
//[> New call should already be created by app. <]
//call = Call::lookup(call_id);
//if (!call) {
//return base_tp;
//}
//} else {
return base_tp;
//}
}
----------- [...] -----------------------------------
regards
Gabriel