That would be great! I've been thinking about this and what other RTP functionality might be nice to have.
> -----Original Message----- > From: [email protected] [mailto:asterisk-video- > [email protected]] 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: [email protected] [mailto:asterisk- > video- > >> [email protected]] 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 > >> _______________________________________________ > >> --Bandwidth and Colocation Provided by http://www.api-digital.com-- > >> > >> asterisk-video mailing list > >> To UNSUBSCRIBE or update options visit: > >> http://lists.digium.com/mailman/listinfo/asterisk-video > > > > _______________________________________________ > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > > > asterisk-video mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-video > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-video mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-video _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-video mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-video
