Re: [Rum] I-D Action: draft-ietf-rum-rue-01.txt

[Richard Shields <[e-mail]>]

OPUS looks like an interesting audio codec. Considering that we support older devices, where video is already pushing the limits of CPU and memory usage, making OPUS MTI raises concerns about how well it would perform on older hardware.

Richard

From: Rum <[e-mail]> On Behalf Of Brian Rosen
Sent: Tuesday, November 5, 2019 2:51 PM
To: Isaac Roach <[e-mail]>
Cc: [e-mail]
Subject: Re: [Rum] I-D Action: draft-ietf-rum-rue-01.txt

[EXTERNAL] In this case I think the IETF considers OPUS to be the best practice audio codec, its free, small, and efficient.  There is only one Wideband codec that is MTI in the WebRTC specs.  I think it would be very hard to get this profile through the IETF without it.

This effort, while some of the motivation is from the US FCC, is broader scope, and thus not limited to FCC interoperability requirements.  One of the things that happens when you take work to the IETF consensus process is that it’s hard to limit to some external organizations requirements.

Brian


On Nov 5, 2019, at 2:22 PM, Isaac Roach <[e-mail]<mailto:[e-mail]>> wrote:

Sorenson has a concern about making OPUS mandatory to implement as part of the RUM.  We don’t necessarily object to improvements that might have benefits for users in the long-run, but making OPUS mandatory specifically at this time in the RUM seems to be beyond what is necessary to implement the FCC’s interoperability requirements.  There’s nothing inherent in establishing a standard interface between devices and providers that requires implementation of OPUS.  This would instead appear to create a new minimum standard for VRS that goes beyond the FCC’s requirements.  In addition, Sorenson is concerned that mandatory implementation of OPUS would be expensive as it would require implementation of the CODEC on endpoints and backend systems when it would likely not be used in the near future for Relay PSTN G.711 calls. It would more likely be used for P2P calls but that is out of scope per the RUM charter.

We’re happy to discuss the merits of these and other long-term goals and improvements for VRS, but this is not the right place to create new minimum requirements that are not necessary for interoperability.

Thanks,

Isaac


From: Rum <[e-mail]<mailto:[e-mail]>> on behalf of Eric Burger <[e-mail]<mailto:[e-mail]>>
Date: Tuesday, November 5, 2019 at 12:09 PM
To: "[e-mail]<mailto:[e-mail]>" <[e-mail]<mailto:[e-mail]>>
Subject: Re: [Rum] I-D Action: draft-ietf-rum-rue-01.txt

The Pulakka paper I referenced for Keith might be why one would transcode from G.711 to Opus. However, I would not mandate it.
https://www.isca-speech.org/archive/archive_papers/interspeech_2006/i06_1245.pdf




On Nov 5, 2019, at 1:44 PM, Paul Kyzivat <[e-mail]<mailto:[e-mail]>> wrote:

The new text in section 6 on Mandatory to Implement is confusing. A G.711 call originating in the PSTN is never going to be *originated* by a RUE.

Calls *originated* by a RUE should *offer* all MTI codecs including OPUS. If the call terminates in the PSTN then OPUS won't be selected in the answer.

OTOH, a VRS provider when relaying a call to a RUE that originated in the PSTN may offer G.711 and not OPUS. In normal use cases such a call will be a VRS call with an interpreter. I *think* in that case audio gets relayed to the RUE, in which case continuing G.711 makes sense. But does audio from interpreter also go to the RUE? If so, transcoding up to OPUS and then mixing in the interpreter might make sense.

So this is heavily entangled in the need for different requirements for the RUE and the VRS Provider.

Thanks,
Paul

On 11/4/19 4:20 PM, [e-mail]<mailto:[e-mail]> wrote:


A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Relay User Machine WG of the IETF.
        Title           : Interoperability Profile for Relay User Equipment
        Author          : Brian Rosen
Filename        : draft-ietf-rum-rue-01.txt
Pages           : 28
Date            : 2019-11-04
Abstract:
   Video Relay Service (VRS) is a term used to describe a method by
   which a hearing persons can communicate with deaf/Hard of Hearing
   user using an interpreter ("Communications Assistant") connected via
   a videophone to the deaf/HoH user and an audio telephone call to the
   hearing user.  The CA interprets using sign language on the
   videophone link and voice on the telephone link.  Often the
   interpreters may be supplied by a company or agency termed a
   "provider" in this document.  The provider also provides a video
   service that allows users to connect video devices to their service,
   and subsequently to CAs and other dead/HoH users.  It is desirable
   that the videophones used by the deaf/HoH/H-I user conform to a
   standard so that any device may be used with any provider and that
   video calls direct between deaf/HoH users work.  This document
   describes the interface between a videophone and a provider.
The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rum-rue/
There are also htmlized versions available at:
https://tools.ietf.org/html/draft-ietf-rum-rue-01
https://datatracker.ietf.org/doc/html/draft-ietf-rum-rue-01
A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-rum-rue-01
Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.
Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/

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