[Live-devel] Live555 MPEG2 Transport Stream packet loss

[xcsmith at rockwellcollins.com]

Hi all,

We've eliminated the NAT/ATM from the equation, and tried streaming the
MPEG-2 TS over UDP (connecting the viewing PC directly to an ADSL line),
and it did improve the sittuation dramatically, however, we sill have
around 0.3 - 0.7% packet loss over the WAN which results in video smearing
here and there, and audio glitches (which are worse than video smearing
because they are easily identified by ear and are more disturbing).

We are wondering if either TS or RTP provide some means of forward error
correction, since this rate of lost packets is normal over the public
Internet, and from what we can tell the player (VLC) can not deal with
those errors (and conceal them from the viewer).

We are trying to find some information regarding embedding FEC information
into the transport stream (as far as we understand it provides at least a
container for error correction data), yet with no success.

Any ideas?

Many thanks,
Ilya

Re: Are you watching the video in real time or just recording it?  you
could use openRTSP to record the video stream and see if it helps at all to
do this.  in openRTSP you can adjust the amount of time to wait for a late
packet.  I'm not sure how much difference it would make.  Have you tried
going back to TCP streaming?  If a packet is too late, it will be
discarded, but if you are only doing a recording and not trying to watch
the video real time, then you can wait a little longer for late packets.

I don't think the TS format provides any sort of other error correction,
aside from having very small packets to stop packet loss from having a
dramatic effect on the system.  I do not believe there is any way to
recalculate the missing data from nearby data, if that's what you are
asking.