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

[Michael Welzl <[e-mail]>]

> On 22 Mar 2016, at 16:14, Harald Alvestrand <[e-mail]> wrote:
> 
> Thank you - yes, it was lost.
> 
> I've filed this suggestion as
> https://github.com/rtcweb-wg/rtcweb-transport/issues/16

Thanks!


> My queries are of course:
> 
> - Is the reference to [coupled] normative or informative?

Seeing that you made I-D.ietf-tsvwg-rtcweb-qos a normative reference, I'd say this one should be normative too. For streams that are known to share a bottleneck (e.g. between the same hosts and multiplexed), this *always* works, not only when routers on your path happen to support it.


> - What is the expected timeline for emission of [coupled]?

I think we're quite close to the finish line. (I'll follow up with a private email)


> 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.

Well, I find it strange for I-D.ietf-tsvwg-rtcweb-qos to be normative and [coupled] to be informative.

Cheers,
Michael



> 
> 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
>>