[Asterisk-video] video RTP question

Gallmeier, Jonathan Jonathan.Gallmeier at polycom.com
Mon Aug 24 10:35:53 CDT 2009


That would be great! I've been thinking about this and what other RTP
functionality might be nice to have. 

> -----Original Message-----
> From: asterisk-video-bounces at lists.digium.com [mailto:asterisk-video-
> bounces at lists.digium.com] On Behalf Of Klaus Darilion
> Sent: Monday, August 24, 2009 9:29 AM
> To: Development discussion of video media support in Asterisk
> Subject: Re: [Asterisk-video] video RTP question
> 
> Maybe ast_frame should have a member *rtp which points to the RTP
> header
> (if exists)
> 
> klaus
> 
> Gallmeier, Jonathan schrieb:
> > Thanks. I appreciate your comment.
> >
> > I spent a little time reading the RTP code in rtp.c. As it turns
out,
> > the 12 bytes prior to the frame data contains the RTP header. The
> header
> > appears to be copied when a copy of the frame is made. All I needed
> to
> > do to recover the header is index _backwards_ 12 bytes from the ptr
> > field in the frame structure.
> >
> > Even though Asterisk is a multi-protocol PBX, RTP is commonly used
to
> > handle real-time video transport. Both H.323 and SIP encapsulate
RTP.
> I
> > don't know what the overall roadmap looks like for Asterisk video
> > support, but handling RTP headers should be pretty high on the list.
> The
> > simple reason is that compressed video frames tend to span multiple
> RTP
> > packets and there are rules on how the video bitstream can be split
> up.
> > Stripping the RTP headers early means that someone else down the
line
> > would need to partially decode the video bitstream to properly re-
> create
> > the RTP header in order to bridge to another protocol. The compute
> > required for this is not insignificant when you consider scaling the
> PBX
> > to a large number of users. I don't know if this makes sense, but I
> > think that it is important to consider if video is to be a major
> feature
> > of Asterisk. At the very least, ensure that protocols do not break
> the
> > existing behavior.
> >
> > Jonathan
> >
> >
> >> -----Original Message-----
> >> From: asterisk-video-bounces at lists.digium.com [mailto:asterisk-
> video-
> >> bounces at lists.digium.com] On Behalf Of Olle E. Johansson
> >> Sent: Saturday, August 22, 2009 3:38 AM
> >> To: Development discussion of video media support in Asterisk
> >> Subject: Re: [Asterisk-video] video RTP question
> >>
> >>
> >> 20 aug 2009 kl. 16.35 skrev Gallmeier, Jonathan:
> >>
> >>> I have a very basic question regarding how the RTP headers are
> >>> handled within the Asterisk video channels. I discovered that the
> >>> RTP headers are stripped off for the audio channels, leaving the
> >>> compressed audio bitstream. Adding them back in is reasonably
> >>> trivial in my soft phone sip bridge application that I'm writing.
> >>>
> >>> For video, is the same thing done? Are the RTP headers stripped
> off,
> >>> leaving only the compressed video bitstream? Is there an easy way
> to
> >>> tell Asterisk channels to leave the RTP headers intact within an
> >>> Asterisk channel so that I don't have to partially parse the
> >>> bitstream to determine RTP header parameters when the video frame
> is
> >>> broken between multiple RTP packets? Unfortunately, I have to give
> >>> my smart phone RTP packets with headers. It won't accept simple
> >>> video bistreams.
> >>>
> >> Asterisk is a multiprotocol PBX and onl a limited set of the
> channels
> >> actually use RTP. So internally, Asterisk doesn't use RTP framing
at
> >> all.
> >>
> >> /O
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