[Live-devel] live555 question

[McLachlan, Donald (IC)]

Hi Ross,

>From that link:

> Outputting QOS statistics
> 
> Use the "-Q" option to output QOS ("quality of service") statistics about the data stream (when the program exits). These statistics
> include the (minimum, average, maximum) bitrate, packet loss rate, and inter-packet gap.

Thanks, that looks promising. I'll look into the example code later today.

"inter-packet gap" is an excellent start. Ultimately I think we'd like to measure end-to-end latency (aka one-way delay), and from that learn the min/max latency. I saw in the FAQ about the presentation time "changing" once the first Sender Report is received. Therefore I'm hoping that after reception of the first SR  one can then compare 'the RTP packet timestamp + SR timestamp correction factor'  to the receiver system time (yes both ends with be synchronized to UTC) and thus obtain the packet latency.

Given that the data flow is across a one-way link (and there is no return path) the sender will not receive Receiver Reports from the receiver.  Off the top of your head, do you know whether this will cause any problems?

Thanks again,
Don


-----Original Message-----
From: live-devel [mailto:live-devel-bounces at ns.live555.com] On Behalf Of Ross Finlayson
Sent: April-25-17 2:56 PM
To: LIVE555 Streaming Media - development & use
Subject: Re: [Live-devel] live555 question


> We are wanting to send video across a unidirectional link. 
> I am hoping that the media server includes timestamps and sequence numbers into the stream such that at the client end one can compute and report (or better yet if existing software will compute and report) stats like:
> -          Number of lost packets.
> -          Min/avg/max length of packet loss events, and the number of packet loss events.
> -          Min/avg/max latency.
>  
> (I think these could be done by inspecting RTP timestamps and sequence 
> numbers) Are these things that can be done with live555?

Yes, our software implements the IETF’s RTP/RTCP standard (including the RTP payload formats for several common video and audio codecs).

We have several demo applications (in source code form) that you can try, as well as a pre-built RTSP server application.  See http://live555.com/liveMedia/#testProgs

(In particular, you can run our “openRTSP” command-line client application with the “-Q” option, to report ‘quality of service’ stats; see http://live555.com/openRTSP/#other-options



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


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