What steps will reproduce the problem?
1.Build and latest webrtc2sip - as of 4th Jan 2014
2.Run freeswitch 1.2.17 as SIP server/proxy
3.Configure webrtc2sip to use running freeswitch as outbound SIP proxy
4.Go to http://www.sipml5.org/call.htm and login using a SIP user account into
FS by filling appropriate fields
5.Make an audio call another user from sipml5 SIP client who is on freeswitch
What is the expected output? What do you see instead?
- Expected output - call should be established successfully if called party is online else to the voice mailbox of freeswitch (default config of fs)
- What I see - When the call is made, sipml5 starts ringing and then displays "Decline" tearing off the call
- The cause is FS is sending 183 session progress SDP with "\n" (LF) as attribute delimiter which is resulting in SDP parse error at webrtc2sip and tearing off the call. webrtc2sip is expecting \r\n (CRLF)
What version of the product are you using? On what operating system?
- latest version of webrtc2sip as of 4th jan 2014
- OS - CentOS6.5 64 bit
Please provide server logs with DEBUG level equal to INFO
- both webrtc2sip log and pcap are attached
Please provide browser logs
- Not applicable
Original issue reported on code.google.com by vmelache...@gmail.com on 8 Jan 2014 at 3:05
Original issue reported on code.google.com by
vmelache...@gmail.comon 8 Jan 2014 at 3:05Attachments: