[asterisk-dev] ast_rtp_engine api

[Joshua Colp]

On Wed, Feb 1, 2017, at 03:15 PM, Michael Blake wrote:
> Hello asterisk-dev list,
> 
> 
> 
> I am working on a rtp proxy that essentially takes a mp4 video stream and
> converts it into a sip endpoint.
> 
> 
> 
> To start I hacked up ekiga to use a text file with a gstreamer pipeline
> defined as a video and audio source, demuxing the video and audio and
> feeding it into the sip call.
> 
> 
> 
> I then modified chan_rtp.c to send both the video and audio streams -
> which
> is currently working.  I can use gstreamer to receive the udp streams and
> play back the audio and video.
> 
> 
> 
> Now I want to get rid of ekiga and make chan_rtp also listen for an audio
> and video incoming udp stream to feed into the call.  I have tried adding
> the source ports to the channel, but the sockets don't actually get
> opened
> and listen.
> 
> 
> 
> Looking in the other channels I see where sockets are manually opened,
> but
> I would rather use the rtp engine.
> 
> 
> 
> Could someone point me in the direction where a channel defines a rtp
> address/port using the ast_rtp_engine and opens the listening socket, or
> some guidance to at least identify the api calls to make that happen?
> 
> 
> 
> I think I am close, but I am missing something.

The UnicastRTP channel driver should already allow the audio portion of
this. It puts the local address information on the channel as dialplan
variables, but it can be used as a basis. Is that what you modified?

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org