[Asterisk-video] video RTP question

[Klaus Darilion]

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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