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<p class=MsoNormal>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.<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>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.<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>Jonathan<o:p></o:p></p>
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