[Live-devel] Droping packets

CodeJones codesjones at gmail.com
Fri Jul 29 11:55:34 PDT 2005


Thanks for the reply.

I am using the recorder with multicast data.  

I think that my problem was that the mrouted was in a bad state and the
multicast joins were taking a really long time.  I fixed this and now I
am not dropping any packs.  

I am recoding it as an MP4 file, but have noted that quicktime complains
that the file is corrupt.  

Codejones

On Thu, 2005-07-28 at 13:15 -0700, Ross Finlayson wrote:

> >The problem seems to be that  I am losing packets on the the first 
> >socket.  When I run the application it opens the two sockets one for 
> >video and one for audio.  The time that it takes to open the second 
> >is to slow to get back
> 
> This seems unlikely.  The input sockets  - for both audio and video - 
> are opened (synchronously, by "MediaSubsession::initiate()"), before 
> any RTSP "PLAY" command is sent to the server.  For unicast streams, 
> no data will start arriving until after the "PLAY" command is 
> sent.  (For multicast streams, however, I suppose it's conceivable 
> that one socket might start buffering incoming data before the other 
> has been initialized, but that shouldn't be a problem (see below).)
> 
> >  and packets are lost and the recoded stream is then out of sync.
> 
> A/V sync is handled by RTCP.  It doesn't matter if some audio or 
> video packets get lost, or audio and video is delayed with respect to 
> one another.  See 
> <http://www.live.com/liveMedia/faq.html#separate-rtp-streams>.
> 
> How are you recording your stream?  If you're recording a ".mov" or 
> ".mp4"-format file (using "-q" or "-4"), then try adding the "-y" 
> option, to generate time-synchronized tracks (see 
> <http://www.live.com/openRTSP/#quicktime>).
> 
> 
> 	Ross Finlayson
> 	LIVE.COM
> 	<http://www.live.com/>
> 
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live.com
> http://lists.live.com/mailman/listinfo/live-devel
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