Hi

I'm originating calls from (mod_sofia's) internal to external profile and joining them into a conference.
The purpose is to play multiple video streams in a conference and consume the resulting canvas by a single VERTO RTC caller.
My setup works but there's a LOT useless RTP traffic (lo) coming from the conference to the video playback channels.
Is there any way to disable the audio/video conference feedback to those play-only video channels?
Perhaps an existing channel variable that I've missed, or some way to tweak the SDP and put a sendonly for audio and video?

My setup is (on top of the vanilla config):

sofia status
                     Name          Type                                       Data      State
=================================================================================================
                 external       profile           sip:mod_sofia@192.168.1.110:5080      RUNNING (1)
                 internal       profile           sip:mod_sofia@192.168.1.110:5060      RUNNING (0)
=================================================================================================

dialplan/default.xml
<extension name="rtsp stream from buggy IP camera">
        <condition field="destination_number" expression="3600">
                <action application="answer" />
                <action application="playback" data="av://rtsp://192.168.1.10/user=admin&password=&channel=1&stream=0.sdp?" />
        </condition>
</extension>

<extension name="static mp4 file">
        <condition field="destination_number" expression="3601">
                <action application="answer" />
                <action application="playback" data="av:///var/www/vid/055_640.mp4" />
        </condition>
</extension>

dialplan/public.xml
    <extension name="public_extensions">
      <condition field="destination_number" expression="^(\d{4})$">
        <action application="transfer" data="$1 XML default"/>
      </condition>
    </extension>

CLI (add 2 video streams to an existing conference)
originate sofia/internal/3600@192.168.1.110:5080 &conference(3500-192.168.1.110)
originate sofia/internal/3601@192.168.1.110:5080 &conference(3500-192.168.1.110)

Thanks


Em 14/11/2016 22:08, Abaci B escreveu:
you can have 2 sip profiles and use originate to create the calls going from 1 profile to the other (one can listen o local ip if it helps)

On Mon, Nov 14, 2016 at 6:39 PM, Saumar Hajjar <saumar@uol.com.br> wrote:
Thanks Anthony,

For your second suggestion, is it possible to "loop a call into the conference with SIP on the same box" using just CLI commands - or do I need a external SIP client for this?


Em 14/11/2016 20:11, Anthony Minessale escreveu:
Your only real option is make the calls from another box or loop them into the conference with SIP on the same box.
The conference file playing mechanism is not designed for simultaneous files to play.



On Thu, Nov 10, 2016 at 12:00 PM, Saumar Hajjar <saumar@uol.com.br> wrote:
Hi,

I'm working in a PoC and I'm considering FS.

I'd like to create a conference and have several video files playing
simultaneously in the canvas.
I already tried: conference name play av:///var/www/vid/video.mp4 and it
works great.
But I need multiple files playing and apparently the above command
queues a video file (or freezes if it's a rtsp stream... Later I'll
confirm this and file a Jira)

I also tried creating a loopback leg, joining the conference, and
playing a video just to this particular member. But it fails because the
member doesn't support video.

Basically I'd like to do something like the examples found at the bottom
of
https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video.
Probably I'm missing some trivial stuff... I really appreciate any
advice on this.

Thanks in advance,

Saumar

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--
Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬


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