[Live-devel] Buffering incoming data with openRTSP?

[Utkarsh Nimesh]

Thanks for your inputs.

The problem I am seeing in my use case is at 1.5 Mbps the video of the stored file gets "bursty" some times and is ok when bit rate is 500Kbps. So I am trying to check on source side to rule out any n/w loss or b/w issues. Decoder seems to be catching up as the data is always arriving late.

The network has enough b/w (does it matter if it's TCP or UDP?) in my case.




-----Original Message-----
From: live-devel-bounces at ns.live555.com [mailto:live-devel-bounces at ns.live555.com] On Behalf Of Ross Finlayson
Sent: Monday, January 03, 2011 10:29 AM
To: LIVE555 Streaming Media - development & use
Subject: Re: [Live-devel] Buffering incoming data with openRTSP?

>After looking at the source code, playCommon.cpp
>
>Function:-
>void continueAfterDESCRIBE(RTSPClient*, int resultCode, char* resultString)
>
>unsigned const thresh = 1000000; // 1 second
>subsession->rtpSource()->setPacketReorderingThresholdTime(thresh);

Note that this code affects *only* the application's  behavior when 
packets happen to be reordered in the network - a situation that 
rarely happens.  It is something that you almost certainly don't need 
to change.


Perhaps you should start by explaining *specifically* what problem(s) 
you think you are seeing with the current code, and what you want to 
do?
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
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