Dear all,
Thank you for your private replies.
I enabled the Null Sound device and made TCP transport and STUN working in my PJSUA2 example.
I was able to compile trunk on my host (macOS) with —without-ssl and —without-sound.
I have a running, stripped down version of pjsua-x86_64-apple-darwin17.5.0. I can answer calls with 180 and 200.
The on_incoming_call callback is called AFTER media_channel_init finished and the Initial answer Response msg 100/INVITE was sent.
The PJSUA2 API behaves differently when a call arrives.
In my PJSUA2 example the on_incoming_callback is called before the initial answer Response msg was created and sent. It is called right after media_channel_init, before media_channel_init returned.
Might this be related to #1923 and #1916?
There is no difference if I try to answer with 180 or 200.
How can this behaviour be fixed so PJSUA2 works as expected?
Thanks for all replies in advance,
best regards
Sascha
Working trace from pjsua-x86_64-apple-darwin17.5.0
14:45:29.174 sip_endpoint.c Processing incoming message: Request msg INVITE/cseq=103 (rdata0x7f979802da28)
14:45:29.174 pjsua_core.c .RX 1313 bytes Request msg INVITE/cseq=103 (rdata0x7f979802da28) from UDP 217.10.79.9:5060:
INVITE sip: XXXXXXXXXX@XXXXXXXXXX:40984;ob SIP/2.0
Record-Route: <sip:217.10.79.9;lr;ftag=as1f7268d3>
Record-Route: <sip:172.20.40.7;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as1f7268d3>
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bKd32b.a06be3fa68fd4436b6ee7002a55d2a5b.0
Via: SIP/2.0/UDP 172.20.40.7;branch=z9hG4bKd32b.15fa37ab70f368719e6acae297bdba12.0
Via: SIP/2.0/UDP 217.10.68.137;branch=z9hG4bKd32b.adc97d37b9c2c1dd5a3fa0462022ea1f.0
Via: SIP/2.0/UDP 217.10.77.105:5060;branch=z9hG4bK19903a75
Max-Forwards: 67
From: “XXXXXXXXXX" <sip: XXXXXXXXXX@xxxxxxxxxx>;tag=as1f7268d3
To: <sip: XXXXXXXXXX@xxxxxxxxxx>
Contact: <sip: XXXXXXXXXX@217.10.77.105:5060>
Call-ID: 67da93b618218d9547ffdf8447b1f4b9@xxxxxxxxxx
CSeq: 103 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Type: application/sdp
Content-Length: 403
v=0
o=root 1371948874 1371948875 IN IP4 217.10.77.105
s=sipgate VoIP GW
c=IN IP4 217.10.77.105
t=0 0
m=audio 28116 RTP/AVP 8 0 3 97 18 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
--end msg--
14:45:29.174 pjsua_call.c .Incoming Request msg INVITE/cseq=103 (rdata0x7f979802da28)
14:45:29.174 pjsua_call.c ..Resetting call_id 0
14:45:29.174 pjsua_call.c ..call = 0x105c0d0b8
14:45:29.174 pjsua_call.c ..call->incoming_data = 0
14:45:29.174 pjsua_call.c ..pjsip_replaces_verify_request is ok.
14:45:29.174 pjsua_call.c ..Invite does not contain a Replaces header.
14:45:29.174 pjsua_call.c ..Clone rdata to call->incoming_data.
14:45:29.175 pjsua_call.c ..Quick SDP check OK.
14:45:29.175 pjsua_call.c ..SDP check OK.
14:45:29.175 pjsua_call.c ..Get contact:
14:45:29.175 pjsua_call.c ..Create Dialog.
14:45:29.175 tsx0x7f979901d2a8 ...Transaction created for Request msg INVITE/cseq=103 (rdata0x7f979802da28)
14:45:29.175 tsx0x7f979901d2a8 ..Incoming Request msg INVITE/cseq=103 (rdata0x7f979802da28) in state Null
14:45:29.175 tsx0x7f979901d2a8 ...State changed from Null to Trying, event=RX_MSG
14:45:29.175 dlg0x7f9799018aa8 ....Transaction tsx0x7f979901d2a8 state changed to Trying
14:45:29.175 dlg0x7f9799018aa8 ..UAS dialog created
14:45:29.175 pjsua_call.c ..Allow rewrite and via_addr set.
14:45:29.175 dlg0x7f9799018aa8 ..Module mod-invite added as dialog usage, data=0x7f9797f2d998
14:45:29.175 dlg0x7f9799018aa8 ...Session count inc to 3 by mod-invite
14:45:29.175 inv0x7f9799018aa8 ..UAS invite session created for dialog dlg0x7f9799018aa8
14:45:29.175 dlg0x7f9799018aa8 ...Session count inc to 3 by mod-pjsua
14:45:29.175 pjsua_call.c ..Trace 1.
14:45:29.175 pjsua_call.c ..Call media_channel_init.
14:45:29.175 pjsua_media.c ..Call 0: initializing media..
14:45:29.211 stun_simple.c ...Warning: unknown attr type 8020 in attr 3. Attribute was ignored.
14:45:29.211 stun_simple.c ...Warning: unknown attr type 8022 in attr 4. Attribute was ignored.
14:45:29.211 stun_simple.c ...Warning: unknown attr type 8020 in attr 3. Attribute was ignored.
14:45:29.211 stun_simple.c ...Warning: unknown attr type 8022 in attr 4. Attribute was ignored.
14:45:29.211 pjsua_media.c ...Note: STUN mapped RTCP port 60115 is not adjacent to RTP port 43711
14:45:29.211 pjsua_media.c ...RTP socket reachable at XXXXXXXXXX:43711
14:45:29.211 pjsua_media.c ...RTCP socket reachable at XXXXXXXXXX:60115
14:45:29.211 srtp0x7f979b801c00 ...SRTP keying SDES created
14:45:29.211 pjsua_media.c ...Media index 0 selected for audio call 0
14:45:29.211 pjsua_media.c ...Call 0: media transport initialization complete: Success
14:45:29.211 pjsua_call.c ..media_channel_init OK.
14:45:29.211 dlg0x7f9799018aa8 ...Session count dec to 3 by mod-pjsua
14:45:29.211 pjsua_call.c ..Timer init OK.
14:45:29.211 pjsua_call.c ..Update NAT.
14:45:29.211 pjsua_call.c ..pjmedia_sdp_neg_get_neg_remote OK.
14:45:29.211 pjsua_call.c ..Call 0: remote NAT type is 0 (Unknown)
14:45:29.211 endpoint ...Response msg 100/INVITE/cseq=103 (tdta0x7f979b8024a8) created
14:45:29.211 dlg0x7f9799018aa8 ...Initial answer Response msg 100/INVITE/cseq=103 (tdta0x7f979b8024a8)
14:45:29.211 pjsua_call.c ..pjsip_inv_initial_answer OK
14:45:29.211 inv0x7f9799018aa8 ...Sending Response msg 100/INVITE/cseq=103 (tdta0x7f979b8024a8)
14:45:29.211 dlg0x7f9799018aa8 ....Sending Response msg 100/INVITE/cseq=103 (tdta0x7f979b8024a8)
14:45:29.211 tsx0x7f979901d2a8 ....Sending Response msg 100/INVITE/cseq=103 (tdta0x7f979b8024a8) in state Trying
14:45:29.211 sip_resolve.c .....Target '217.10.79.9:5060' type=UDP resolved to '217.10.79.9:5060' type=UDP (UDP transport)
14:45:29.211 pjsua_core.c .....TX 688 bytes Response msg 100/INVITE/cseq=103 (tdta0x7f979b8024a8) to UDP 217.10.79.9:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 217.10.79.9;received=217.10.79.9;branch=z9hG4bKd32b.a06be3fa68fd4436b6ee7002a55d2a5b.0
Via: SIP/2.0/UDP 172.20.40.7;branch=z9hG4bKd32b.15fa37ab70f368719e6acae297bdba12.0
Via: SIP/2.0/UDP 217.10.68.137;branch=z9hG4bKd32b.adc97d37b9c2c1dd5a3fa0462022ea1f.0
Via: SIP/2.0/UDP 217.10.77.105:5060;branch=z9hG4bK19903a75
Record-Route: <sip:217.10.79.9;lr;ftag=as1f7268d3>
Record-Route: <sip:172.20.40.7;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as1f7268d3>
Call-ID: 67da93b618218d9547ffdf8447b1f4b9@xxxxxxxxxx
From: "XXXXXXXXXX" <sip: XXXXXXXXXX@xxxxxxxxxx>;tag=as1f7268d3
To: <sip: XXXXXXXXXX@xxxxxxxxxx>
CSeq: 103 INVITE
Content-Length: 0
--end msg--
14:45:29.211 tsx0x7f979901d2a8 .....State changed from Trying to Proceeding, event=TX_MSG
14:45:29.236 dlg0x7f9799018aa8 ......Transaction tsx0x7f979901d2a8 state changed to Proceeding
14:45:29.236 pjsua_call.c ..pjsip_inv_initial_answer OK.
14:45:29.236 pjsua_call.c ..Call on_incoming_call callback now.
Failing trace from PJSUA2
14:46:27.235 sip_endpoint.c Processing incoming message: Request msg INVITE/cseq=103 (rdata0x7f90ce80bc28)
14:46:27.235 pjsua_core.c .RX 1313 bytes Request msg INVITE/cseq=103 (rdata0x7f90ce80bc28) from UDP 217.10.79.9:5060:
INVITE sip: XXXXXXXXXX@ XXXXXXXXXX:40984;ob SIP/2.0
Record-Route: <sip: XXXXXXXXXX;lr;ftag=as1d6c896b>
Record-Route: <sip:172.20.40.7;lr>
Record-Route: <sip: XXXXXXXXXX;lr;ftag=as1d6c896b>
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK6f5f.b2d582d1382df29a93b35dc9758059cf.0
Via: SIP/2.0/UDP 172.20.40.7;branch=z9hG4bK6f5f.efc0b26a3a3307206e7dd88963aa4397.0
Via: SIP/2.0/UDP 217.10.68.137;branch=z9hG4bK6f5f.9b1773a4515b6a2e06cc7d12752a648d.0
Via: SIP/2.0/UDP 217.10.77.107:5060;branch=z9hG4bK703571c7
Max-Forwards: 67
From: "XXXXXXXXXX" <sip: XXXXXXXXXX@xxxxxxxxxx>;tag=as1d6c896b
To: <sip: XXXXXXXXXX@xxxxxxxxxx>
Contact: <sip: XXXXXXXXXX@ XXXXXXXXXX:5060>
Call-ID: 6c2b84e108d10561291bb34d1d46d0ab@xxxxxxxxxx
CSeq: 103 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Type: application/sdp
Content-Length: 403
v=0
o=root 1194626159 1194626160 IN IP4 217.10.77.107
s=sipgate VoIP GW
c=IN IP4 217.10.77.107
t=0 0
m=audio 19442 RTP/AVP 8 0 3 97 18 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
--end msg--
14:46:27.235 pjsua_call.c .Incoming Request msg INVITE/cseq=103 (rdata0x7f90ce80bc28)
14:46:27.235 tsx0x7f90cd826aa8 ...Transaction created for Request msg INVITE/cseq=103 (rdata0x7f90ce80bc28)
14:46:27.235 tsx0x7f90cd826aa8 ..Incoming Request msg INVITE/cseq=103 (rdata0x7f90ce80bc28) in state Null
14:46:27.235 tsx0x7f90cd826aa8 ...State changed from Null to Trying, event=RX_MSG
14:46:27.235 dlg0x7f90cd822aa8 ....Transaction tsx0x7f90cd826aa8 state changed to Trying
14:46:27.235 dlg0x7f90cd822aa8 ..UAS dialog created
14:46:27.235 dlg0x7f90cd822aa8 ..Module mod-invite added as dialog usage, data=0x7f90cd403198
14:46:27.235 dlg0x7f90cd822aa8 ...Session count inc to 3 by mod-invite
14:46:27.235 inv0x7f90cd822aa8 ..UAS invite session created for dialog dlg0x7f90cd822aa8
14:46:27.235 dlg0x7f90cd822aa8 ...Session count inc to 3 by mod-pjsua
14:46:27.235 pjsua_media.c ..Call 0: initializing media..
14:46:27.259 stun_simple.c ...Warning: unknown attr type 8020 in attr 3. Attribute was ignored.
14:46:27.259 stun_simple.c ...Warning: unknown attr type 8022 in attr 4. Attribute was ignored.
14:46:27.260 stun_simple.c ...Warning: unknown attr type 8020 in attr 3. Attribute was ignored.
14:46:27.260 stun_simple.c ...Warning: unknown attr type 8022 in attr 4. Attribute was ignored.
14:46:27.260 pjsua_media.c ...Note: STUN mapped RTCP port 42112 is not adjacent to RTP port 10949
14:46:27.260 pjsua_media.c ...RTP socket reachable at XXXXXXXXXX:10949
14:46:27.260 pjsua_media.c ...RTCP socket reachable at XXXXXXXXXX:42112
14:46:27.260 srtp0x7f90cf003000 ...SRTP keying SDES created
14:46:27.260 pjsua_media.c ...Media index 0 selected for audio call 0
14:46:27.260 pjsua_media.c ...Call 0: media transport initialization complete: Success
*** onIncomingCall callId=0
accId = 0
remAudioCount = 1
remVideoCount = 0
Sleep ...
Answer ...
14:46:28.261 pjsua_call.c ...Answering call 0: code=180
14:46:28.261 pjsua_call.c ....Pending answering call 0 upon completion of media transport
14:46:28.261 pjsua_call.c ...Call 0 hanging up: code=0..
14:46:28.261 pjsua_call.c ....Pending call 0 hangup upon completion of media transport
14:46:28.261 pjsua_call.c ...Call 0 hanging up: code=500..
14:46:28.261 pjsua_call.c ....Pending call 0 hangup upon completion of media transport
14:46:28.261 dlg0x7f90cd822aa8 ...Session count dec to 3 by mod-pjsua
14:46:28.261 pjsua_call.c ..Call 0: remote NAT type is 0 (Unknown)
14:46:28.261 endpoint ...Response msg 100/INVITE/cseq=103 (tdta0x7f90cf00a2a8) created
14:46:28.261 dlg0x7f90cd822aa8 ...Initial answer Response msg 100/INVITE/cseq=103 (tdta0x7f90cf00a2a8)
14:46:28.261 inv0x7f90cd822aa8 ...Sending Response msg 100/INVITE/cseq=103 (tdta0x7f90cf00a2a8)
14:46:28.261 dlg0x7f90cd822aa8 ....Sending Response msg 100/INVITE/cseq=103 (tdta0x7f90cf00a2a8)
14:46:28.261 tsx0x7f90cd826aa8 ....Sending Response msg 100/INVITE/cseq=103 (tdta0x7f90cf00a2a8) in state Trying
14:46:28.261 sip_resolve.c .....Target '217.10.79.9:5060' type=UDP resolved to '217.10.79.9:5060' type=UDP (UDP transport)
14:46:28.261 pjsua_core.c .....TX 688 bytes Response msg 100/INVITE/cseq=103 (tdta0x7f90cf00a2a8) to UDP 217.10.79.9:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 217.10.79.9;received=217.10.79.9;branch=z9hG4bK6f5f.b2d582d1382df29a93b35dc9758059cf.0
Via: SIP/2.0/UDP 172.20.40.7;branch=z9hG4bK6f5f.efc0b26a3a3307206e7dd88963aa4397.0
Via: SIP/2.0/UDP 217.10.68.137;branch=z9hG4bK6f5f.9b1773a4515b6a2e06cc7d12752a648d.0
Via: SIP/2.0/UDP 217.10.77.107:5060;branch=z9hG4bK703571c7
Record-Route: <sip:217.10.79.9;lr;ftag=as1d6c896b>
Record-Route: <sip:172.20.40.7;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as1d6c896b>
Call-ID: 6c2b84e108d10561291bb34d1d46d0ab@xxxxxxxxxx
From: "XXXXXXXXXX" <sip: XXXXXXXXXX@xxxxxxxxxx>;tag=as1d6c896b
To: <sip: XXXXXXXXXX@xxxxxxxxxx>
CSeq: 103 INVITE
Content-Length: 0
--end msg--
14:46:28.262 tsx0x7f90cd826aa8 .....State changed from Trying to Proceeding, event=TX_MSG
14:46:28.262 dlg0x7f90cd822aa8 ......Transaction tsx0x7f90cd826aa8 state changed to Proceeding
14:46:28.262 pjsua_call.c ..Call 0 hanging up: code=500..
14:46:28.262 inv0x7f90cd822aa8 ....Sending Response msg 500/INVITE/cseq=103 (tdta0x7f90cf00a2a8)
14:46:28.262 dlg0x7f90cd822aa8 .....Sending Response msg 500/INVITE/cseq=103 (tdta0x7f90cf00a2a8)
14:46:28.262 tsx0x7f90cd826aa8 .....Sending Response msg 500/INVITE/cseq=103 (tdta0x7f90cf00a2a8) in state Proceeding
14:46:28.262 pjsua_core.c ......TX 719 bytes Response msg 500/INVITE/cseq=103 (tdta0x7f90cf00a2a8) to UDP 217.10.79.9:5060:
SIP/2.0 500
Via: SIP/2.0/UDP 217.10.79.9;received=217.10.79.9;branch=z9hG4bK6f5f.b2d582d1382df29a93b35dc9758059cf.0
Via: SIP/2.0/UDP 172.20.40.7;branch=z9hG4bK6f5f.efc0b26a3a3307206e7dd88963aa4397.0
Via: SIP/2.0/UDP 217.10.68.137;branch=z9hG4bK6f5f.9b1773a4515b6a2e06cc7d12752a648d.0
Via: SIP/2.0/UDP 217.10.77.107:5060;branch=z9hG4bK703571c7
Record-Route: <sip:217.10.79.9;lr;ftag=as1d6c896b>
Record-Route: <sip:172.20.40.7;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as1d6c896b>
Call-ID: 6c2b84e108d10561291bb34d1d46d0ab@xxxxxxxxxx
From: "XXXXXXXXXX" <sip:XXXXXXXXXX@xxxxxxxxxx>;tag=as1d6c896b
To: <sip:XXXXXXXXXX@xxxxxxxxxx>;tag=gPQlSfEw6u0X92Hi1O5lOWLMWJXkglD2
CSeq: 103 INVITE
Content-Length: 0
--end msg--
14:46:28.262 tsx0x7f90cd826aa8 ......State changed from Proceeding to Completed, event=TX_MSG
14:46:28.262 dlg0x7f90cd822aa8 .......Transaction tsx0x7f90cd826aa8 state changed to Completed
14:46:28.262 pjsua_media.c .........Call 0: deinitializing media..
14:46:28.262 pjsua_media.c ..........Call 0: cleaning up provisional media, prov_med_cnt=1, med_cnt=0
14:46:28.262 dlg0x7f90cd822aa8 .........Session count dec to 4 by mod-invite
> On 24. Apr 2018, at 13:44, Sascha <sascha@xxxxxxxxxxx> wrote:
>
> Dear all,
>
> I am currently trying to use PJSUA2 in C++ to answer calls (version 2.7.2, compiled from source on Ubuntu 16.04).
> I created a minimalistic example using the documentation [1].
>
> It registers successfully with the SIP provider (Spigate.de in this case).
> When I receive an incoming call, I receive an INVITE message and the onIncomingCall callback is called. That’s fine.
> Answering the call does not work.
>
> std::cout << "*** onIncomingCall callId=" << iprm.callId << std::endl;
> Call *call = new MyCall(*this, iprm.callId);
> CallOpParam op;
> op.statusCode = PJSIP_SC_RINGING;
> call->answer(op);
>
> Instead, pjsua is replying with 500 and I do not hear it ringing. Instead, I get a “The person you have called is temporarily not available."
>
> 11:38:08.548 pjsua_call.c .Incoming Request msg INVITE/cseq=103 (rdata0x7f69d4001d38)
> 11:38:08.548 pjsua_media.c ..Call 0: initializing media..
> 11:38:08.548 pjsua_media.c ...RTP socket reachable at 172.18.0.2:4000
> 11:38:08.549 pjsua_media.c ...RTCP socket reachable at 172.18.0.2:4001
> 11:38:08.549 pjsua_media.c ...Media index 0 selected for audio call 0
> *** onIncomingCall callId=0
> 11:38:08.550 pjsua_call.c ...Answering call 0: code=180
> 11:38:08.550 pjsua_call.c ....Pending answering call 0 upon completion of media transport
> 11:38:08.550 pjsua_call.c ...Call 0 hanging up: code=0..
> 11:38:08.550 pjsua_call.c ....Pending call 0 hangup upon completion of media transport
> 11:38:08.550 pjsua_call.c ...Call 0 hanging up: code=500..
> 11:38:08.550 pjsua_call.c ....Pending call 0 hangup upon completion of media transport
> 11:38:08.550 pjsua_core.c .....TX 690 bytes Response msg 100/INVITE/cseq=103 (tdta0x7f69d4013198) to UDP 217.10.79.9:5060:
> SIP/2.0 100 Trying
>
> There is a full log of the software running on the Gitlab CI. It actually was connected to Sipgate and I called the number during the run of the CI. Hope not do disclose too much personal information :-) You can find all the compile steps in the Dockerfile and build.sh. Hope this gives a reproducible environment with a full log.
>
> What is the correct way to answer a call?
>
> Thank you very much in advance.
>
> Sascha
>
> [1] https://github.com/strongly-typed/pjsua2-demo/blob/master/main.cpp
> [2] https://gitlab.com/strongly-typed/pjsua2-demo/-/jobs/64690392
>
>
> _______________________________________________
> Visit our blog: http://blog.pjsip.org
>
> pjsip mailing list
> pjsip@xxxxxxxxxxxxxxx
> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
_______________________________________________
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@xxxxxxxxxxxxxxx
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
