Re: [AVTCORE] WGLC review on draft-ietf-avtcore-multi-media-rtp-session-09

Colin Perkins <csp@csperkins.org> Mon, 07 September 2015 11:35 UTC

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From: Colin Perkins <csp@csperkins.org>
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Date: Mon, 7 Sep 2015 12:35:02 +0100
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Cc: Ali Begen <abegen@cisco.com>, "avt@ietf.org" <avt@ietf.org>
Subject: Re: [AVTCORE] WGLC review on draft-ietf-avtcore-multi-media-rtp-session-09
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Hi,

Thanks for the review; some responses inline.

On 6 Sep 2015, at 14:24, Asveren, Tolga <tasveren@sonusnet.com> wrote:
> I did not monitor the progress of this draft over its lifetime, so this is more of an "outsider's review" (which probably is useful in its own right)
> 
> i- 2. Terminology
> "Media Type:  The general type of media data used by a real-time
>      application.  The media type corresponds to the value used in the
>      <media> field of an SDP m= line.  The media types defined at the
>      time of this writing are "audio", "video", "text", "application",
>      and "message"."
> 
> - It could be good to provide a reference to the corresponding IANA registry.
> - I know it is not very common but still defined/possibly used, is there a specific reason not to include "image" (e.g., image/t38, image/g3fax) in this list?

I copied the list from draft-ietf-mmusic-rfc4566bis-15, which doesn't mention "image". I agree we should mention it, as should 4566bis. (I've cc'd Ali, for comment).

> ii- 3.  Background and Motivation
> In general, this section seems trying to do a "hard sell", which IMHO can be softened up a little bit.
> 
> - "However, we note that many RTP-using application don't use network
>   QoS features, and don't expect or desire any separation in network
>   treatment of their media packets, independent of whether they are
>   audio, video or text.  When an application has no such desire, it
>   doesn't need to provide a transport flow structure that simplifies
>   flow based QoS."
> This really depends on the environment and deployment model. In addition to that, providing QoS with DiffServ is still possible when multiple types of media are sent in a single RTP session as explained in draft-ietf-dart-dscp-rtp (which is already referenced by this draft)

It's not clear what change you want here. Do you have a suggestion for how to update the text?

> -  "1.  increased delay to establish a complete session, since each of
>       the transport layer flows needs to be negotiated and established;"
> At least negotiation part can happen in parallel. And when multiple streams use the same RTP session, isn't there still a need to negotiate different SSRC values, one per stream? A reference to RFC5576 could be useful as well.

Negotiation can only happen in parallel to a certain extent, before the amount of STUN traffic causes congestion. The SSRCs also don't necessarily have to be negotiated in advance for either model. 

> iii- 5.3.  Per-SSRC Media Type Restrictions 
> This section provides the justification why different SSRC but be used for audio and video streams but not really why a SSRC can't be re-used for streams of different type. It would be good to cover that aspect as well.

Sorry, I'm not sure what you're asking here. Can you rephrase?

> iv- 4.  Applicability
> "Compatible RTCP Behaviour:  The RTCP timing rules enforce a single
>      RTCP reporting interval for all participants in an RTP session.
>      Flows with very different media sending rate or RTCP feedback
>      requirements cannot be multiplexed together, since this leads to
>      either excessive or insufficient RTCP for some flows, depending
>      how the RTCP session bandwidth, and hence reporting interval, is
>      configured."
> This part IMHO requires more justification. Why is it not possible to send RTCP in different rates for different flows? Some RTCP packets could have reports pertaining to a single stream whereas the other ones (when timing for reports for different streams overlaps or is sufficiently close enough) for multiple streams. Aggregate RTCP bandwidth need shouldn't be a constraining factor just because a single transport flow is used AFAICS. If that is not the case, it really could be useful to explain why.

The RTCP timing rules in RFC 3550 don't support different reporting intervals for different SSRCs (other than a sender/receiver split), and doing so runs a risk of causing inappropriate RTCP timeouts. We could add a reference to RFC 3550 here, if it helps?

> v- 7.  Signalling
> "When using SDP signalling, the BUNDLE extension
> [I-D.ietf-mmusic-sdp-bundle-negotiation] is used to signal RTP
> sessions containing multiple media types."
> It could be a good idea to reference relevant sections as the bundle-draft has some irrelevant aspects (arguably which even could be confusing in the context of this specification).

Can you be specific on how BUNDLE should be referenced?

Cheers,
Colin




-- 
Colin Perkins
https://csperkins.org/