[Live-devel] Droping packets

[Ross Finlayson]

>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/>