[vlc-devel] Re: lip sych problem
vato at wnet.ua
Fri Nov 25 09:25:11 CET 2005
serkan bozkurt wrote:
> Thanks for your replay Viktor,
> I have some questions ,
> First; you are saying that network jitter may cause
> lip synch problem.
> Jitter means the timestamp frequency difference
> between 2 adjacent
> rtp packets in the receiver side.
Should add here - RTP streams are actually 2 (at least) different streams
with different payloads, that are not syncronised well (look at live555
mailing list). I.e - system will "sync on average bitrate", not on each
video packet. It means, that LiveMedia free library do not contain
mechanisms to make A/V sync/resync. Try to add 2 seconds delay in stream -
and all will be OK. I've made so, trying to connect VLC to
> if network goes down then timestamp difference between
> adjacent rtp packets will be bigger than the previous
> This may cause starvation on the receiver side buffers
> because datas are late!
> But after network is okey from the RTCP SR
> timestamps(they are sent periodically), lip synch must
> be ok isn't it?
No. RTP streams with different payloads have different sync rate. The
streams may contain real timecode, but (IMHO) the timecode is replaced with
"timestamps" to avoid complexity, that causes different timestamps for
different RTP substreams.
> Second problem:Why bid rtp packets may result in
> buffer starvation
> on the reiver side can you please explain.
Large packets are sended in parts (fragments). In some situations these
fragments can reach receiver side with jitter, that can break packet
assemble. This packet will be dropped. Most common reccomendation - reduce
packet size to MTU munus 50-60 bytes, to avoid packet fragmentation. Or
check, (in GigEthernet environment) if your network cleanly support
> Third one :
> For the time being live55.com is inactive but I know 2
> days ago it was okey.
> Do you have any idea about this.
No idea :(
Wnet project manager
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