Hi Colin,
see below.
Am 04.08.2016 um 12:00 schrieb Colin Perkins <csp(_at_)csperkins(_dot_)org>:
On 3 Aug 2016, at 14:54, Allison Mankin
<allison(_dot_)mankin(_at_)gmail(_dot_)com> wrote:
Hi,
I've reviewed this draft (draft-ietf-rtcpweb-transports-14.txt) as part of
the TSV Area Review Team, paying special attention to transport-related
concerns. Please take these as any other IETF last call comments.
Summary: this draft specifies the mandatory transport protocols (and
transport features) associated with the use of WebRTC media. It does not
appear to pose any transport-related danger, except perhaps that a
reviewer's head aches over the number of RFCs that are needed to get media
bits from point A to point B, but this is not a fault of the draft. The
draft is broadly ready for publication as a PS, however there are a few
issues for the Transport Area.
Section 3.4:
If TCP connections are used, RTP framing according to [RFC4571
] MUST
be used, both for the RTP packets and for the DTLS packets used to
carry data channels.
About the passage above, RFC4571 doesn't talk about DTLS. It looks like
this passage also needs a reference to whatever of the specs defines framing
for DTLS?
Section 4.1 Local Prioritization
This section describes the resource allocations that are expected for
prioritized different streams when there is congestion. There are two
highly relevant congestion control documents that are approved (or nearly
so), and I can't see that the RTCWB WG considered them from my quick review
of mailing list discussions, but it would be a good idea for this draft to
call them out:
draft-ietf-avtcore-rtp-circuit-breakers-17 - this has enough positions to
pass and is waiting for an AD followup (looks like for the IANA re-review
after a version change). It puts some additional considerations on flows
that are likely to be relevant to the flows in the present draft.
This is listed as “MUST implement” in draft-ietf-rtcweb-rtp-usage-26, which
is referenced from Section 3.5 of the rtcweb-transport draft.
Colin
rtcweb-transport says
"For transport of media, secure RTP is used. The details of the profile of RTP
used are described in "RTP Usage“ [I-D.ietf-rtcweb-rtp-usage]."
Given that this doc is called "Transports for WebRTC“, I would appreciate if it
says slightly more about the recommendations given in rtcweb-rtp-usage,
especialy regarding congestion control.
What’s about the following?
"For transport of media, secure RTP is used. The details of the profile of RTP
used are described in "RTP Usage“ [I-D.ietf-rtcweb-rtp-usage], which mandates
the use of a circuit breaker [draft-ietf-avtcore-rtp-circuit-breakers-17] and
congestion control (see [draft-ietf-rmcat-cc-requirements-09] for further
guidance).“
Mirja
draft-ietf-rmcat-cc-requirements-09 - this is in the RFC Editor queue and
seems to be waiting for the rtcweb-overview draft, to which it normatively
refers. I think it would be better if the rmcat draft referenced
rtcweb-transpoarts, and if rtcweb-transports would check on its alignment
with the rmcat requirements in the congestion control remarks in section 4.1.
Section 4.2 Usage of Quality of Service - DSCP and Multiplexing
I will just flag here that I reviewed the mailing list and it seems that
there was a lot of TSV review of the DSCP material here already, and a
consensus reached.
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Colin Perkins
https://csperkins.org/
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