Re: [rtcweb] I-D Action: draft-ietf-rtcweb-transports-12.txt

[Harald Alvestrand <[e-mail]>]

Thank you - yes, it was lost.

I've filed this suggestion as
https://github.com/rtcweb-wg/rtcweb-transport/issues/16

My queries are of course:

- Is the reference to [coupled] normative or informative?
- What is the expected timeline for emission of [coupled]?

I see that RFC 7657 got published with [coupled] as an informative
reference.
The "e.g." in your first suggestion might be loose enough to warrant an
informative reference.

Den 22. mars 2016 15:45, skrev Michael Welzl:
> Hi,
> 
> On 26 February, I sent an email to rtcweb in which I made some suggestions to this document. I see that these have not been incorporated, and my email has also never been answered (except that I-D.ietf-dart-dscp-rtp was replaced by RFC 7657, but that may not have been due to my email). I can understand that: probably my prior email just drowned in the WebRTC Audio Codec related thread. However I do think that these comments would be good to address, so I'm copying in the email again below.
> 
> Cheers,
> Michael
> 
> ----
> 
> Hi,
> 
> Overall, I like this document a lot - it makes for a very good read!
> 
> - but I think it would make sense for section 4.1 to explicitly point to draft-ietf-rmcat-coupled-cc (the next version of which is going to explain how weights much be set to adhere to the priority levels that are described in this section; it's easy, we just didn't have this text in there yet).
> 
> 
> To be concrete, I suggest the following two changes:
> 
> ***
>   When an WebRTC implementation has packets to send on multiple streams
>   that are congestion-controlled under the same congestion controller,
>   the WebRTC implementation SHOULD cause data to be emitted in such a
>   way that each stream at each level of priority is being given
>   approximately twice the transmission capacity (measured in payload
>   bytes) of the level below.
> ***
> 
> should be:
> 
> ***
>   When a WebRTC implementation has packets to send on multiple streams
>   that are congestion-controlled under the same congestion controller
>   or multiple coupled congestion controllers (e.g. using the mechanism in
>   [draft-ietf-rmcat-coupled-cc]),
>   the WebRTC implementation SHOULD cause data to be emitted in such a
>   way that each stream at each level of priority is being given
>   approximately twice the transmission capacity (measured in payload
>   bytes) of the level below.
> ***
> 
> (note a fixed nit in there: the second word is "a" instead of "an")
> 
> 
> and, perhaps even more importantly, a small change in section 4.2:
> 
> ***
>   More advice on the use of DSCP code points with RTP is given in
>   [I-D.ietf-dart-dscp-rtp].
> ***
> 
> should be:
> 
> ***
>   More advice on the use of DSCP code points with RTP as well as coupled
>   congestion control is given in [I-D.ietf-dart-dscp-rtp].
> ***
> 
> and in fact I-D.ietf-dart-dscp-rtp should now be RFC 7657.
> 
> 
> Cheers,
> Michael
>