| 12345678910111213141516171819202122232425262728293031323334353637 |
- import inc_sip as sip
- import inc_sdp as sdp
- # Ticket https://github.com/pjsip/pjproject/issues/718
- # RTC doesn't put rport in Via, and it is report to have caused segfault.
- complete_msg = \
- """INVITE sip:localhost SIP/2.0
- Via: SIP/2.0/UDP $LOCAL_IP:$LOCAL_PORT;branch=z9hG4bK74a60ee5
- From: <sip:tester@localhost>;tag=as2858a32c
- To: <sip:pjsua@localhost>
- Contact: <sip:tester@$LOCAL_IP:$LOCAL_PORT>
- Call-ID: 123@localhost
- CSeq: 1 INVITE
- Max-Forwards: 70
- Content-Type: application/sdp
- Content-Length: 285
- v=0
- o=root 4236 4236 IN IP4 192.168.1.11
- s=session
- c=IN IP4 192.168.1.11
- t=0 0
- m=audio 14390 RTP/AVP 0 3 8 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:3 GSM/8000
- a=rtpmap:8 PCMA/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=silenceSupp:off - - - -
- a=ptime:20
- a=sendrecv
- """
- sendto_cfg = sip.SendtoCfg( "RTC no rport", "--null-audio --auto-answer 200",
- "", 200, complete_msg=complete_msg)
|
