ConfBridge User Joining Not Getting Video

Home » Asterisk Users » ConfBridge User Joining Not Getting Video
Asterisk Users 6 Comments

I am running 18.8.0 – videosupport is enabled. I get video calls no problem.

However when I make a call file to a soft phone and include:
Codecs: ulaw,h264
in the call file…

sip show channels – shows:
4013c15f1f4cdff (ulaw|h264) No Tx: ACK
so clearly the caller has h264.

Then when I “automatically” request another softphone to join my conf bridge… the soft phone rings, and answers – all I get is audio and sip show channels for that device:
5c77cf1455e4afc (ulaw) No Tx: ACK

How do I get Video in the confbridge ?

Thanks

Jerry

6 thoughts on - ConfBridge User Joining Not Getting Video

  • hi Josh,

    here is the sip debug… It shows the the first call negotiate video – but the second call to bring the end video device into the conf – no video negotitation.

    Audio is at 15542
    Adding codec ulaw to SDP
    Adding codec alaw to SDP
    Adding codec gsm to SDP

    Thanks,

    Jerry

    Asterisk 18.8.0, Copyright (C) 1999 – 2021, Sangoma Technologies Corporation and others. Created by Mark Spencer
    Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type ‘core show license’ for details.
    ========================================================================Running as user ‘silentm’
    Running under group ‘silentm’
    Connected to Asterisk 18.8.0 currently running on DevKaufer (pid = 597669)
    Really destroying SIP dialog ‘c24843e8-d7f1-0740-08dd-8b79fe39a15a’ Method:
    REGISTER

    <--- SIP read from UDP:192.168.2.22:5060 --->

    <------------->
    == Using SIP VIDEO CoS mark 6
    == Using SIP RTP CoS mark 5
    Audio is at 17816
    Video is at 192.168.1.6:10746
    Adding video codec vp8 to SDP
    Adding codec ulaw to SDP
    Adding codec opus to SDP
    Reliably Transmitting (NAT) to 192.168.1.6:48124:
    INVITE sips:mason.kaufer.visualcampus@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss SIP/2.0
    Via: SIP/2.0/WS 192.168.1.6:5060;branch=z9hG4bK73b689b9;rport Max-Forwards: 70
    From: “Mason Kaufer 34” ;tag=as101db932
    To:
    Contact:
    Call-ID: 4c4ab96b31b9fa52416eacf563ae2bf1@192.168.1.6:5060
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX 18.8.0
    Date: Thu, 13 Jan 2022 13:46:18 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer Content-Type: application/sdp Content-Length: 1106

    v=0
    o=root 1174630673 1174630673 IN IP4 192.168.1.6
    s=Asterisk PBX 18.8.0
    c=IN IP4 192.168.1.6
    b=CT:5120
    t=0 0
    m=audio 17816 UDP/TLS/RTP/SAVPF 0 107
    a=rtpmap:0 PCMU/8000
    a=rtpmap:107 opus/48000/2
    a=maxptime:60
    a=ice-ufrag:4ff9bfd157a3896a6bc7f86d312dde00
    a=ice-pwd:2c8b8f052875a1cd7096d71478ff3567
    a

  • chan_sip did not add a video stream. What is the actual configuration for it? What is the actual call file used for it?

  • sip.conf has videosupport in the general section.

    I did find that where I am “joining” the person in the conference I did not have the Codecs: set. I added that – doing better – its negotiating video now – but still not showing me video for a conference.

    Jerry

  • Can you make a 1-to-1 video call between two of the devices (which I assume does give you video?), and then get just those two to join a conference, and see the difference in SDP?

    Antony.

    —
    The first fifty percent of an engineering project takes ninety percent of the time, and the remaining fifty percent takes another ninety percent of the time.

    Please reply to the list;
    please *don’t* CC me.

    —