Incoming Call in sipml5 with kamailio as SIP Server

[Jason Sia]

I use kamailio websocket module as the websocket server and freeswitch as the media server. I can successfully login/register. I can also make outgoing calls from chrome to web. I can also receive the ringing signal of the incoming call. However when I accept the call it shows Call Rejected. Here are the javascript logs: (Kamailio sip server show no errors)

__tsip_transport_ws_onmessage SIPml-api.js?svn=179:1

recv=INVITE sip:12...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws SIP/2.0 Via: SIP/2.0/WS 192.x.x.x:10080;branch=z9hG4bKe868.77662c27427f0826d5c385379ade01ff.0 From: "2222"<sip:22...@192.x.x.x>;tag=2912790649 To: <sip:12...@192.x.x.x> Contact: "2222"<sip:22...@192.168.1.102:47462;alias=112.208.63.200~47462~1;transport=udp> Call-ID: 0067eaf99bf7b1a0...@192.168.1.102 CSeq: 4515 INVITE Content-Type: application/sdp Content-Length: 319 Record-Route: <sip:192.x.x.x:10080;transport=ws;r2=on;lr=on;nat=yes> Record-Route: <sip:192.x.x.x;r2=on;lr=on;nat=yes> Via: SIP/2.0/UDP 192.168.1.102:47462;rport=47462;received=112.208.63.200;branch=z9hG4bK93c357f5239d6043a5c1f84739594803383334 Max-Forwards: 16 v=0 o=- 1376174762425 1376174762426 IN IP4 192.x.x.x s=- c=IN IP4 192.x.x.x t=0 0 m=audio 43322 RTP/AVP 96 97 3 0 8 127 a=rtpmap:96 GSM-EFR/8000 a=rtpmap:97 AMR/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-15 a=nortpproxy:yes SIPml-api.js?svn=179:1

State machine: tsip_transac_ist_Started_2_Proceeding_X_INVITE SIPml-api.js?svn=179:1

SEND: SIP/2.0 100 Trying (sent from the Transaction Layer) Via: SIP/2.0/WS 192.x.x.x:10080;branch=z9hG4bKe868.77662c27427f0826d5c385379ade01ff.0 From: "2222"<sip:22...@192.x.x.x>;tag=2912790649 To: <sip:12...@192.x.x.x> Call-ID: 0067eaf99bf7b1a0...@192.168.1.102 CSeq: 4515 INVITE Content-Length: 0 Via: SIP/2.0/UDP 192.168.1.102:47462;rport=47462;received=112.208.63.200;branch=z9hG4bK93c357f5239d6043a5c1f84739594803383334 Record-Route: <sip:192.x.x.x:10080;transport=ws;r2=on;lr=on;nat=yes> Record-Route: <sip:192.x.x.x;r2=on;lr=on;nat=yes> SIPml-api.js?svn=179:1

ICE servers:[{"url":"stun:stun.l.google.com:19302"},{"url":"stun:stun.counterpath.net:3478"},{"url":"stun:numb.viagenie.ca:3478"}] SIPml-api.js?svn=179:1

setRemoteDescription(offer) v=0 o=- 1376174762425 1376174762426 IN IP4 192.x.x.x s=- c=IN IP4 192.x.x.x t=0 0 m=audio 43322 RTP/AVP 96 97 3 0 8 127 a=rtpmap:96 GSM-EFR/8000 a=rtpmap:97 AMR/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-15 a=nortpproxy:yes SIPml-api.js?svn=179:1

State machine: s0000_Started_2_Ringing_X_iINVITE SIPml-api.js?svn=179:1

onSetRemoteDescriptionError SIPml-api.js?svn=179:1

1. SetRemoteDescription failed: Called with a SDP without crypto enabled. 
2. tsk_utils_log_errorSIPml-api.js?svn=179:1
3. tmedia_session_jsep01.onSetRemoteDescriptionErrorSIPml-api.js?svn=179:3
4. (anonymous function)
5. SIPml-api.js?svn=179:1

==stack event = m_permission_requested SIPml-api.js?svn=179:1

State machine: tsip_transac_ist_Proceeding_2_Proceeding_X_1xx SIPml-api.js?svn=179:1

SEND: SIP/2.0 180 Ringing Via: SIP/2.0/WS 192.x.x.x:10080;branch=z9hG4bKe868.77662c27427f0826d5c385379ade01ff.0 From: "2222"<sip:22...@192.x.x.x>;tag=2912790649 To: <sip:12...@192.x.x.x>;tag=9FeWJF2hDrLBAHIBVONs Contact: <sip:12...@df7jal23ls0d.invalid;transport=ws> Call-ID: 0067eaf99bf7b1a0...@192.168.1.102 CSeq: 4515 INVITE Content-Length: 0 Via: SIP/2.0/UDP 192.168.1.102:47462;rport=47462;received=112.208.63.200;branch=z9hG4bK93c357f5239d6043a5c1f84739594803383334 Record-Route: <sip:192.x.x.x:10080;transport=ws;r2=on;lr=on;nat=yes> Record-Route: <sip:192.x.x.x;r2=on;lr=on;nat=yes> Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE SIPml-api.js?svn=179:1

==stack event = i_new_call SIPml-api.js?svn=179:1

onGetUserMediaSuccess SIPml-api.js?svn=179:1

createAnswer SIPml-api.js?svn=179:1

onCreateSdpError SIPml-api.js?svn=179:1

1. CreateAnswer failed. 
2. tsk_utils_log_errorSIPml-api.js?svn=179:1
3. tmedia_session_jsep01.onCreateSdpErrorSIPml-api.js?svn=179:3
4. (anonymous function)
5. SIPml-api.js?svn=179:1

State machine: s0000_Ringing_2_Terminated_X_Reject SIPml-api.js?svn=179:1

=== INVITE Dialog terminated === SIPml-api.js?svn=179:1

PeerConnection::stop() SIPml-api.js?svn=179:1

==stack event = m_permission_accepted SIPml-api.js?svn=179:1

1. This/PeerConnection is null: unexpected 
2. tsk_utils_log_errorSIPml-api.js?svn=179:1
3. tmedia_session_jsep01.onIceCandidateSIPml-api.js?svn=179:3
4. o_pc.onicecandidate
5. SIPml-api.js?svn=179:1

State machine: tsip_transac_ist_Proceeding_2_Completed_X_300_to_699 SIPml-api.js?svn=179:1

SEND: SIP/2.0 603 Failed to get local SDP Via: SIP/2.0/WS 192.x.x.x:10080;branch=z9hG4bKe868.77662c27427f0826d5c385379ade01ff.0 From: "2222"<sip:22...@192.x.x.x>;tag=2912790649 To: <sip:12...@192.x.x.x>;tag=9FeWJF2hDrLBAHIBVONs Call-ID: 0067eaf99bf7b1a0...@192.168.1.102 CSeq: 4515 INVITE Content-Length: 0 Via: SIP/2.0/UDP 192.168.1.102:47462;rport=47462;received=112.208.63.200;branch=z9hG4bK93c357f5239d6043a5c1f84739594803383334 Record-Route: <sip:192.x.x.x:10080;transport=ws;r2=on;lr=on;nat=yes> Record-Route: <sip:192.x.x.x;r2=on;lr=on;nat=yes> Reason: SIP; cause=603; text="Failed to get local SDP" SIPml-api.js?svn=179:1

==session event = terminated SIPml-api.js?svn=179:1

State machine: tsip_transac_ist_Any_2_Terminated_X_cancel SIPml-api.js?svn=179:1

__tsip_transport_ws_onmessage SIPml-api.js?svn=179:1

recv=ACK sip:12...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws SIP/2.0 Via: SIP/2.0/WS 192.x.x.x:10080;branch=z9hG4bKe868.77662c27427f0826d5c385379ade01ff.0 From: "2222"<sip:22...@192.x.x.x>;tag=2912790649 To: <sip:12...@192.x.x.x>;tag=9FeWJF2hDrLBAHIBVONs Call-ID: 0067eaf99bf7b1a0...@192.168.1.102 CSeq: 4515 ACK Content-Length: 0 Max-Forwards: 16 SIPml-api.js?svn=179:1

SEND: SIP/2.0 481 Dialog/Transaction Does Not Exist Via: SIP/2.0/WS 192.x.x.x:10080;branch=z9hG4bKe868.77662c27427f0826d5c385379ade01ff.0 From: "2222"<sip:22...@192.x.x.x>;tag=2912790649 To: <sip:12...@192.x.x.x>;tag=9FeWJF2hDrLBAHIBVONs Call-ID: 0067eaf99bf7b1a0...@192.168.1.102 CSeq: 4515 ACK Content-Length: 0

[Mamadou]

WebRTC requires ICE and SRTP(SDES for Chrome and DTLS for Firefox).  The incoming INVITE doesn't include these features.
I'd recommend using webrtc2sip.org or if supported by your server to enable it.

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Mamadou DIOP - Technology Evangelist
Doubango Telecom - Paris, France
http://www.doubango.org
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