[Freeswitch-users] Sofia endpoint that only sends video
Saumar Hajjar
saumar at uol.com.br
Sun Nov 20 00:21:54 MSK 2016
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 at 192.168.1.110:5080 RUNNING (1)
internal profile
sip:mod_sofia at 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 at 192.168.1.110:5080
&conference(3500-192.168.1.110)
originate sofia/internal/3601 at 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 at uol.com.br
> <mailto:saumar at 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 at uol.com.br <mailto:saumar at 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
>> <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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>> --
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>>
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