| Tristan Matthews | 0a329cc | 2013-07-17 13:20:14 -0400 | [diff] [blame] | 1 | # $Id$ |
| 2 | import inc_sip as sip |
| 3 | import inc_sdp as sdp |
| 4 | |
| 5 | # Ticket http://trac.pjsip.org/repos/ticket/718 |
| 6 | # RTC doesn't put rport in Via, and it is report to have caused segfault. |
| 7 | complete_msg = \ |
| 8 | """INVITE sip:localhost SIP/2.0 |
| 9 | Via: SIP/2.0/UDP $LOCAL_IP:$LOCAL_PORT;branch=z9hG4bK74a60ee5 |
| 10 | From: <sip:tester@localhost>;tag=as2858a32c |
| 11 | To: <sip:pjsua@localhost> |
| 12 | Contact: <sip:tester@$LOCAL_IP:$LOCAL_PORT> |
| 13 | Call-ID: 123@localhost |
| 14 | CSeq: 1 INVITE |
| 15 | Max-Forwards: 70 |
| 16 | Content-Type: application/sdp |
| 17 | Content-Length: 285 |
| 18 | |
| 19 | v=0 |
| 20 | o=root 4236 4236 IN IP4 192.168.1.11 |
| 21 | s=session |
| 22 | c=IN IP4 192.168.1.11 |
| 23 | t=0 0 |
| 24 | m=audio 14390 RTP/AVP 0 3 8 101 |
| 25 | a=rtpmap:0 PCMU/8000 |
| 26 | a=rtpmap:3 GSM/8000 |
| 27 | a=rtpmap:8 PCMA/8000 |
| 28 | a=rtpmap:101 telephone-event/8000 |
| 29 | a=fmtp:101 0-16 |
| 30 | a=silenceSupp:off - - - - |
| 31 | a=ptime:20 |
| 32 | a=sendrecv |
| 33 | """ |
| 34 | |
| 35 | |
| 36 | sendto_cfg = sip.SendtoCfg( "RTC no rport", "--null-audio --auto-answer 200", |
| 37 | "", 200, complete_msg=complete_msg) |
| 38 | |