Typically video is synced by using the SSRC of the two streams and the RTCP reports that link a timestamp and a NTP time. From this, you can figure out which packets corresponded to the same moment of time in each stream. In practice though, usually it's possible to sync the video and the audio for a real-time VoIP call without a great deal of effort, just by playing them when they are received. This isn't always the case though, and the proper way to do it is the way I mentioned above.
<br><br>Also, depending on how the jitter buffer is written, it may not work for video. I.e., the marker bit has a different meaning in video, so that may break a jitter buffer written primarily for audio.<br><br>Duane<br>
<br><div><span class="gmail_quote">On 5/18/06, <b class="gmail_sendername">Olle E Johansson</b> <<a href="mailto:oej@edvina.net">oej@edvina.net</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<br>18 maj 2006 kl. 19.29 skrev Zoa:<br><br>><br>> I never actually tried that, but if those are normal RTP packets it<br>> should not be a difference with audio packets i think ?<br>> Didnt you know somebody who claimed it broke video ? Could this
<br>> person tell us how he is using the video with asterisk ?<br><br>I don't remember, but we need to make sure that the jitterbuffer is<br>applied to both the audio rtp and the video rtp<br>streams. Those are different in
rtp.c.<br><br>Any of you implementors out there on the list that can provide any<br>advice here? Do we need to sync with rtcp?<br>Any specific requirements for video jitterbuffers?<br><br>/Olle<br>_______________________________________________
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