[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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