[OpenSIPS-Devel] WebRtc and opensips

2015-08-21 Thread Герман Чичикин
 Hi, Razvan, thanks for the fast response .

ngep shows:

#
U 192.168.10.62:5060 - 192.168.10.82:5060
  INVITE sip:11@192.168.10.82:5060 SIP/2.0..Record-Route: sip:192.168.10
  .62;r2=on;lr..Record-Route: sip:192.168.10.62:8091;transport=ws;r2=on;lr
  ..Via: SIP/2.0/UDP 192.168.10.62:5060;branch=z9hG4bKa9a4.7d75f496.0;i=2..Vi
  a: SIP/2.0/WS fvnm7l6n6841.invalid;received=192.168.10.36;branch=z9hG4bK775
  6386..Max-Forwards: 68..To: sip: 11@192.168.10.62 ..From: 22 sip
  : 22@192.168.10.62 ;tag=pisvn8sql7..Call-ID: 48akg6r345bbtdiofjcg..CSeq:
   3313 INVITE..X-Can-Renegotiate: false..Contact: sip: p06q144b@192.168.10.3
  6:4110;transport=ws;ob..Content-Type: application/sdp..Session-Expires: 90
  ..Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER..Supported: tim
  er,ice,replaces,outbound..User-Agent: JsSIP 0.7.4..Content-Length: 1096
  v=0..o=mozilla...THIS_IS_SDPARTA-40.0.2 4294967295 0 IN IP4 0.0.0.0..s=-..t
  =0 0..a=msid-semantic:WMS *..m=audio 5 RTP/AVP 109 9 0 8..c=IN IP4 192.
  168.10.62..a=end-of-candidates..a=msid:{c2d03933-6c9c-4078-9eeb-b8ac29caea1
  6} {38e37d34-8ac9-4b3f-bdaf-228eeb111dd2}..a=rtpmap:109 opus/48000/2..a=rtp
  map:9 G722/8000/1..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=ssrc:12013
  51425 cname:{ab765227-cdd6-4ce5-aed8-fb7840d18020}..a=sendrecv..a=rtcp:5000
  1..a=rtcp-mux..m=video 50020 RTP/AVP 120 126 97..c=IN IP4 192.168.10.62..a=
  end-of-candidates..a=fmtp:120 max-fs=12288;max-fr=60..a=fmtp:126 profile-le
  vel-id=42e01f;level-asymmetry-allowed=1;packetization-mode=1..a=fmtp:97 pro
  file-level-id=42e01f;level-asymmetry-allowed=1.   
#
U 192.168.10.82:5060 - 192.168.10.62:5060
  SIP/2.0 100 Trying..Via: SIP/2.0/UDP 192.168.10.62:5060;branch=z9hG4bKa9a4.
  7d75f496.0;i=2..Via: SIP/2.0/WS fvnm7l6n6841.invalid;received=192.168.10.36
  ;branch=z9hG4bK7756386..Record-Route: sip:192.168.10.62;r2=on;lr..Record-
  Route: sip:192.168.10.62:8091;transport=ws;r2=on;lr..From: 22 sip:
   22@192.168.10.62 ;tag=pisvn8sql7..To: sip: 11@192.168.10.62 
..Call-
  ID: 48akg6r345bbtdiofjcg..CSeq: 3313 INVITE..Allow: INVITE, OPTIONS, ACK, B
  YE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE..Server: SIPPER for PhonerLite..C
  ontent-Length: 0  
#
U 192.168.10.82:5060 - 192.168.10.62:5060
  SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 192.168.10.62:5060;branch=z9hG4bKa9a4
  .7d75f496.0;i=2..Via: SIP/2.0/WS fvnm7l6n6841.invalid;received=192.168.10.3
  6;branch=z9hG4bK7756386..Record-Route: sip:192.168.10.62;r2=on;lr..Record
  -Route: sip:192.168.10.62:8091;transport=ws;r2=on;lr..From: 22 sip
  : 22@192.168.10.62 ;tag=pisvn8sql7..To: sip: 11@192.168.10.62 
;tag=8
  062fe20a945e51183b7e539fe35f40c..Call-ID: 48akg6r345bbtdiofjcg..CSeq: 3313
  INVITE..Contact: sip:11@192.168.10.82:5060..Allow: INVITE, OPTIONS, A
  CK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE..Server: SIPPER for PhonerLi
  te..Content-Length: 0 
#
U 192.168.10.82:5060 - 192.168.10.62:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.10.62:5060;branch=z9hG4bKa9a4.7d75
  f496.0;i=2..Via: SIP/2.0/WS fvnm7l6n6841.invalid;received=192.168.10.36;bra
  nch=z9hG4bK7756386..Record-Route: sip:192.168.10.62;r2=on;lr..Record-Rout
  e: sip:192.168.10.62:8091;transport=ws;r2=on;lr..From: 22 sip:
   22@192.168.10.62 ;tag=pisvn8sql7..To: sip: 11@192.168.10.62 
;tag=8062fe
  20a945e51183b7e539fe35f40c..Call-ID: 48akg6r345bbtdiofjcg..CSeq: 3313 INVIT
  E..Contact: sip:11@192.168.10.82:5060..Content-Type: application/sdp.
  .Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE..S
  ession-Expires: 90;refresher=uac..Supported: replaces, timer, from-change..
  Server: SIPPER for PhonerLite..Content-Length:   289v=0..o=- 1976025328
   1 IN IP4 192.168.10.82..s=SIPPER for PhonerLite..c=IN IP4 192.168.10.82..t
  =0 0..m=audio 5062 RTP/AVP 107 8 0 9..a=rtpmap:107 opus/48000..a=rtpmap:9 G
  722/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtcp-mux..a=ssrc:19
  05944947..a=sendrecv..m=video 0 RTP/AVP 120.. 
#
U 192.168.10.62:5060 - 192.168.10.82:5060
  ACK sip:11@192.168.10.82:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.10.62:5
  060;branch=z9hG4bKa9a4.7d75f496.2;i=2..Via: SIP/2.0/WS fvnm7l6n6841.invalid
  ;received=192.168.10.36;branch=z9hG4bK8369992..Max-Forwards: 68..To: sip:1
   1@192.168.10.62 ;tag=8062fe20a945e51183b7e539fe35f40c..From: 22 
  sip: 22@192.168.10.62 ;tag=pisvn8sql7..Call-ID: 48akg6r345bbtdiofjcg..CS
  eq: 3313 ACK..Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER..Su
  pported: outbound..User-Agent: JsSIP 0.7.4..Content-Length: 0 
#
U 192.168.10.62:5060 - 192.168.10.82:5060
  BYE sip:11@192.168.10.82:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.10.62:5
  060;branch=z9hG4bK79a4.0ef211d.0;i=2..Via: SIP/2.0/WS fvnm7l6n6841.invalid;
  

[OpenSIPS-Devel] WebRtc and opensips

2015-08-20 Thread Герман Чичикин
 Hi, All!

Having trouble with testing webrtc with a rtpengin. I use opensips.cfg from 
opensips.org (Documentation - Tutorials - WebSocket Transport using 
OpenSIPS), opensips v2.1, clients - sipml5, jssip demo webphones or my own 
script in various combinations. Clients register without problems. When a 
client calls another, the callee joyfully calls. But when I pick up the phone 
call is dropped, session ends and no any media. The caller receives a bye 
from opensips. 
Anybody can tell in what direction to dig?

Best regards, Herman.
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