[Live-devel] Calculating rtptime for RTSP/PLAY-Response "RTP-Info" header

[Chris Burns]

The crux of my question concerns the *delta* that a reference
implementation (Helix USM) applies to one of the rtptime=X values that
is reported in the RTP-Info header. All of our testing (with ~30
different brand/model combinations) has shown that this is critical for
many mobile phones to correctly apply A/V sync.
 
The live555 code (the 2008.07.25 release) does not adjust (calculate a
delta) the RTP-Info header in this way. It simply reports the seqNum &
rtptime of the first packet on each stream.
 
And while you may think that RTSP/RTP/RTCP clients should not use the
RTP timestamps, these phones are doing just that. Wouldn't it be nice if
we could play nicely with them too?
 
Cheers & thanks for all your hard work and excellent code.
 
ChrisB

Chris Burns
M: +64 21 391 286
chris.burns at abdeus.com

<http://www.abdeus.com/> 

 


________________________________

	From: live-devel-bounces at ns.live555.com
[mailto:live-devel-bounces at ns.live555.com] On Behalf Of Ross Finlayson
	Sent: Friday, 08 August 2008 12:02
	To: LIVE555 Streaming Media - development & use
	Subject: Re: [Live-devel] Calculating rtptime for
RTSP/PLAY-Response "RTP-Info" header
	
	

		I'm specifically trying to work out how to create the
"RTP-Info" header in a RTSP/PLAY-Response.


	Our server implementation does this automatically (see
"RTSPServer.cpp", and search for "rtpInfo").  You shouldn't have to do
anything yourself to generate this.

	Similarly, at the client end, the information is filled in
automatically - if you wish, you can just access the
"MediaSubsession::rtpInfo" structure.  Most clients, however, will not
need to access this structure directly.  Instead, if they wish to get
the current 'normal play time' (NPT) for the stream, they can just call
"MediaSubsession::getNormalPlayTime()", which uses the "rtpInfo"
structure.

	Also (as I have explained several times before), RTSP/RTP/RTCP
clients (receivers) should rarely, if ever, need to look at RTP
timestamps, sequence numbers, or RTCP packet data.  Our receiving code
automatically uses this information itself, to give you a
properly-synchronized presentation time for each incoming frame of data.
	-- 

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

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