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Telechat Review of draft-ietf-rum-rue-09

Request Review of draft-ietf-rum-rue
Requested revision No specific revision (document currently at 11)
Type Telechat Review
Team Transport Area Review Team (tsvart)
Deadline 2021-12-14
Requested 2021-11-19
Authors Brian Rosen
I-D last updated 2021-11-29
Completed reviews Artart Last Call review of -09 by Rich Salz (diff)
Genart Last Call review of -09 by Matt Joras (diff)
Secdir Last Call review of -09 by Russ Mundy (diff)
Tsvart Telechat review of -09 by Dr. Bernard D. Aboba (diff)
Assignment Reviewer Dr. Bernard D. Aboba
State Completed
Request Telechat review on draft-ietf-rum-rue by Transport Area Review Team Assigned
Posted at
Reviewed revision 09 (document currently at 11)
Result Ready w/issues
Completed 2021-11-29
This document has been reviewed as part of the transport area review team's
ongoing effort to review key IETF documents. These comments were written
primarily for the transport area directors, but are copied to the document's
authors and WG to allow them to address any issues raised and also to the IETF
discussion list for information.

When done at the time of IETF Last Call, the authors should consider this
review as part of the last-call comments they receive. Please always CC if you reply to or forward this review.

Reviewer: Bernard Aboba
Document: draft-ietf-rum-rue-09
Result: Ready with Issues

Overall, found several issues with this document:

1. The specification appears to contradict some of its normative references
with respect to support of IPv6.

2. Although the document does include language within the provider
configuration, it does not mention language negotiation [RFC 8373], which
included relay services as a use case.  So I'm curious as to how users with
different communications requirements might be accommodated. For example, a
user who is speech impaired but not hearing impaired, and so would like to
communicate via ASL to the interpreter (or realtime text in English if video
fails) but can receive spoken English.

3. Maintainability. This specification does not appear implementable using the
WebRTC 1.0 API, and as a result, it cannot run in a browser.   Major changes
would be required to WebRTC code bases such as libwebrtc and pion to allow for
native implementation.  The separate fork that would be needed would become
increasingly difficult to maintain over time.

1. IPv6 support

Section 4 says:

"   Implementations MUST support IPv4 and IPv6.  Dual stack support is
   NOT required and provider implementations MAY support separate
   interfaces for IPv4 and IPv6 by having more than one server in the
   appropriate SRV record where there is either an A or AAAA record in
   each server DNS record but not both.  The same version of IP MUST be
   used for both signaling and media of a call unless ICE ([RFC8445]) is
   used, in which case candidates may explicitly offer IPv4, IPv6 or
   both for any media stream."

[BA] This document requires that RFC 8835 be supported, but this paragraph
conflicts with that document in multiple ways. RFC 8835 indicates ICE as both
required to implement and to use.  When it says that "Dual stack support is NOT
required" is it referring to the client or the provider? RFC 8835 seems to
conflict in its IPv6 requirements as well.

3. WebRTC usage

Section 5.5 says:

   Implementations MUST conform to [RFC8835] except for its guidance on
   the WebRTC data channel, which this specification does not use.  See
   Section 6.2 for how RUE supports real-time text without the data

[BA]  The WebRTC 1.0 API does not allow data to be sent except via the data
channel API.  So when it says that the WebRTC data channel is not used, by what
mechanism is real-time text to be sent?  This is not audio/video data so that
it cannot be sent using the RTCRtpSender API.   The only way I can think of
that this might be implemented is to add back support for the RTP data channel
so that real-time text could be implemented on top of the RTCDataChannel send()
method. However that in turn would violate aspects of the WebRTC security
model, such as the deprecation of SDES.

If that is the intent here (it's the only practical way that RTT can be
supported other than via the gateway model), then I would prefer that it be
stated explicitly.

Other places in the document conflict with the RFC 8835 requirement.

For example, RFC 8835 Section 3.4 says:

   The primary mechanism for dealing with middleboxes is ICE, which is
   an appropriate way to deal with NAT boxes and firewalls that accept
   traffic from the inside, but only from the outside if it is in
   response to inside traffic (simple stateful firewalls).

   ICE [RFC8445] MUST be supported.  The implementation MUST be a full
   ICE implementation, not ICE-Lite.  A full ICE implementation allows
   interworking with both ICE and ICE-Lite implementations when they are
   deployed appropriately.

RFC 8835 also has requirements for support of IPv4 and IPv6 that seem to be in
conflict with the statements in Section 4.   For example, RFC 8835 Section 3.2

3.2.  Ability to Use IPv4 and IPv6

   Web applications running in a WebRTC browser MUST be able to utilize
   both IPv4 and IPv6 where available -- that is, when two peers have
   only IPv4 connectivity to each other, or they have only IPv6
   connectivity to each other, applications running in the WebRTC
   browser MUST be able to communicate.

   When TURN is used, and the TURN server has IPv4 or IPv6 connectivity
   to the peer or the peer's TURN server, candidates of the appropriate
   types MUST be supported.  The "Happy Eyeballs" specification for ICE
   [RFC8421] SHOULD be supported.

Section 6 of the document says:

   This specification adopts the media specifications for WebRTC
   ([RFC8825]).  Where WebRTC defines how interactive media
   communications may be established using a browser as a client, this
   specification assumes a normal SIP call.  The RTP, RTCP, SDP and
   specific media requirements specified for WebRTC are adopted for this
   document.  The RUE is a WebRTC "non-browser" endpoint, except as
   noted expressly below.

[BA] Is RUE is really a WebRTC "non-browser" endpoint?  If the goal is to allow
RUE to be easily built on top of native WebRTC libraries such as libwebrtc or
pion, then it should inherit WebRTC requirements such as ICE, dual stack
support, etc.

Section 6.1 says:

   Implementations MUST support [RFC8834] except that MediaStreamTracks
   are not used.  Implementations MUST conform to Section 6.4 of

[BA] Since MediaStreamTracks are how audio/video is obtained from devices (or
rendered), I don't understand how a WebRTC application (browser or non-browser)
can function without them.  Elsewhere, the specification states that the data
channel isn't used, now it seems to say that audio/video isn't used either.

RFC 8827 is the security architecture of WebRTC.  Is it really true that RUE
implementations only need to support Section 6.4??  If so, this allows major
deviations from WebRTC to the point where interoperability could be severely

6.2.  Text-Based Communication

   Implementations MUST support real-time text ([RFC4102] and [RFC4103])
   via T.140 media.  One original and two redundant generations MUST be
   transmitted and supported, with a 300 ms transmission interval.
   Implementations MUST support [RFC9071] especially for emergency
   calls.  Note that RFC4103 is not how real-time text is transmitted in
   WebRTC and some form of transcoder would be required to interwork
   real-time text in the data channel of WebRTC to RFC4103 real-time

   Transport of T.140 real-time text in WebRTC is specified in
   [RFC8865], using the WebRTC data chanel.  RFC 8865 also has some
   advice on how gateways between RFC 4103 and RFC 8865 should operate.
   It is RECOMMENDED that RFC 8865 including multiparty support is used
   for communication with browser-based WebRTC implementations.
   Implementations MUST support [RFC9071].

[BA] The reason why RFC 8865 was developed was that there was no practical way
to support RFC 4103 except via the RTCDataChannel API. Requiring RFC 4103 to be
supported doesn't make the practical problems go away - you need a way to
provide the RTT data to be sent.

When you say that implementations MUST support RFC 9071, how is this supposed
to work in a WebRTC implementation?

6.3.  Video

   Implementations MUST conform to [RFC7742] with following exceptions:
   only H.264, as specified in [RFC7742], is Mandatory to Implement, and
   VP8 support is OPTIONAL at both the device and providers.  This is
   because backwards compatibility is desirable, and older devices do
   not support VP8.

[BA]  The reality is that H.264 is not very widely used in WebRTC applications
(less than 1 percent of calls use it), and as a result, implementations are
quite buggy.  I do not believe that it is serving RUE users well to make VP8
optional.   Even if you're going to stick with H.264, you might consider going
beyond only requiring support only for constrained baseline profile.  Some
implementations now support constrained high profile, for example.

Section 6.8

   For backwards compatibility with calling devices that do not support
   the foregoing methods, implementations MUST implement SIP INFO
   messages to send and receive XML encoded Picture Fast Update messages
   according to [RFC5168].

[BA] Really? Earlier in the document is says that backwards compatibility is
not a strict requirement.  Any widely used WebRTC code base will support NACK,
FIR and PLI.  seems