Re: [rtcweb] WG Last Call: draft-ietf-rtcweb-use-cases-and-requirements-10

"Eggert, Lars" <lars@netapp.com> Fri, 18 January 2013 16:19 UTC

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From: "Eggert, Lars" <lars@netapp.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Thread-Topic: [rtcweb] WG Last Call: draft-ietf-rtcweb-use-cases-and-requirements-10
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Date: Fri, 18 Jan 2013 16:19:31 +0000
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] WG Last Call: draft-ietf-rtcweb-use-cases-and-requirements-10
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Hi,

a few quick comments:

>    F3              Transmitted streams and data MUST be rate
>                    controlled.

Would like to add something along the lines of "and the send rate MUST be reduced when network congestion is detected." I guess that's implied but it doesn't hurt to make it explicit. RMCAT will take care of this eventually, but until then, it's the browser's job.

>    F5              The browser MUST be able to render good quality
>                    audio and video even in the presence of
>                    reasonable levels of jitter and packet losses.
> 
>                    TBD: What is a reasonable level?

I'd suggest "a few percent." Internet protocols in general don't do well when loss is higher than 10% or so.

>    F23             The browser must be able to send short
>                    latency unreliable datagram traffic to a
>                    peer browser.

It may be a good idea to point to RFC5404 here, which has further guidelines about such traffic.

Lars