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RE: Media Stastics in API

From: Romascanu, Dan (Dan) <dromasca@avaya.com>
Date: Wed, 31 Aug 2011 11:49:27 +0200
Message-ID: <EDC652A26FB23C4EB6384A4584434A04038D6B23@307622ANEX5.global.avaya.com>
To: "Cullen Jennings" <fluffy@cisco.com>, <public-webrtc@w3.org>
Hi Cullen,

You would probably like to add packet delay in each direction.

You also need to provide these measurements for each individual stream
which is part of the session (maybe this is implicit in your proposal). 



> -----Original Message-----
> From: public-webrtc-request@w3.org [mailto:public-webrtc-
> request@w3.org] On Behalf Of Cullen Jennings
> Sent: Tuesday, August 30, 2011 9:34 PM
> To: public-webrtc@w3.org
> Subject: Media Stastics in API
> When the browser sends and receives audio and video, all kinds of
> statistics and reporting information is possible. We get information
> from IP, ICMP, RTP, RTCP, codecs and more. I feel that applications
> really need access to some of this information and would like to start
> a thread on what JS applications need. What I really really really
> don't want to do is enumerate everything that is possible to generate
> then expose all that via an API. I'd rather use our experience with
> existing voice and video to figure out what turns out to be useful and
> report that.
> Heres is a few things I think might be useful as a starting point
> Packet loss rate in each direction
> Round trip time and jitter
> Current total bitrate each direction
> These three are the most basic bits of information need to detect and
> understand why an application may be experiencing poor quality media.
> Thoughts? What else do we need? If you say all of RTCP-XR I am going
> release the flying monkeys.
> I'm also interested in a simple API proposal for this that has the
> right sort of extensibility properties as more things get added over
> time.
Received on Wednesday, 31 August 2011 09:49:57 UTC

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