Asterisk 13.26.0 WebRTC: Asterisk Not Passing Along Video

Home » Asterisk Users » Asterisk 13.26.0 WebRTC: Asterisk Not Passing Along Video
Asterisk Users 3 Comments

Hello

I am trying to set up webRTC video calls from my Chrome webbrowser
(Fedora) to my Chrome webbrowser (Windows 10).

There is local video input (I can see myself), but never video on the receiving side.

This is the case in both directions (so it makes no difference which peer is calling which peer).

Both webRTC SIP peers have opus and H264 codec in their peer definition :

  Video Support: Yes
  Prim.Transp. : WS
  Allowed.Trsp : WSS
  SIP Options  : (none)
  Codecs       : (opus|h264)
  Status       : OK (75 ms)
  Useragent    : SIP.js/0.12.0
  Reg. Contact : sip:llghjqha@192.0.2.239;transport=wss
  RTP Engine   : asterisk
  Encryption   : Yes
  RTCP Mux     : Yes

  Video Support: Yes
  Prim.Transp. : WS
  Allowed.Trsp : WSS
  SIP Options  : (none)
  Codecs       : (opus|h264)
  Status       : OK (47 ms)
  Useragent    : SIP.js/0.12.0
  Reg. Contact : sip:6ltm4mqe@192.0.2.7;transport=wss
  RTP Engine   : asterisk
  Encryption   : Yes
  RTCP Mux     : Yes

In general sip.conf I have :

videosupport=yes disallow=all allow=alaw allow=opus allow=h264

When one peer makes a SIP INVITE for a video call, it is clear to me that the necessary codec information is present (this all looks fine to me) :

(calling webRTC client)

SIP Debugging Enabled for IP: 99.99.255.55
[May 10 10:45:24]
[May 10 10:45:24] <--- SIP read from WS:99.99.255.55:47732 --->
[May 10 10:45:24] INVITE sip:17@wss.mydomain.tld SIP/2.0
[May 10 10:45:24] Via: SIP/2.0/WSS 192.0.2.7;branch=z9hG4bK9220692
[May 10 10:45:24] Max-Forwards: 70
[May 10 10:45:24] To:
[May 10 10:45:24] From: “WC User Chrome”
;tag=sdmbqkquhe
[May 10 10:45:24] Call-ID: 3g51uvbnnioje6riokqu
[May 10 10:45:24] CSeq: 4132 INVITE
[May 10 10:45:24] Contact:
[May 10 10:45:24] Allow:
ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
[May 10 10:45:24] Supported: outbound
[May 10 10:45:24] User-Agent: SIP.js/0.12.0
[May 10 10:45:24] Content-Type: application/sdp
[May 10 10:45:24] Content-Length: 5098
[May 10 10:45:24]
[May 10 10:45:24] v=0
[May 10 10:45:24] o=- 6075323372920596423 2 IN IP4 127.0.0.1
[May 10 10:45:24] s=-
[May 10 10:45:24] t=0 0
[May 10 10:45:24] a=group:BUNDLE audio video
[May 10 10:45:24] a=msid-semantic: WMS I46iog3EpKvlzvX9g0MMsh3ON7hT9qtZwZ4E
[May 10 10:45:24] m=audio 34197 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106
105 13 110 112 113 126
[May 10 10:45:24] c=IN IP4 99.99.255.55
[May 10 10:45:24] a=rtcp:9 IN IP4 0.0.0.0
[May 10 10:45:24] a

3 thoughts on - Asterisk 13.26.0 WebRTC: Asterisk Not Passing Along Video

  • Hello

    is this mailing list still active ?

    Op 10-05-19 om 14:10 schreef Jonas Kellens:

    …

  • It is still active. Video under chan_sip, however, is not something many do and in particular it is possible with WebRTC that something has changed and caused problems or there is a bug in a case. The chan_sip module is community supported so it does not see a lot of change.

    The chan_pjsip module is maintained and in regards to video is something that the team at Sangoma who work on Asterisk daily use for video meetings.

    —
    Joshua C. Colp Digium – A Sangoma Company | Senior Software Developer
    445 Jan Davis Drive NW – Huntsville, AL 35806 – US
    Check us out at: http://www.digium.com & http://www.asterisk.org

    —

  • Hello

    is this mailing list still active ?

    Op 10-05-19 om 14:10 schreef Jonas Kellens: