Re: [rtcweb] Fwd: Last Call: <draft-ietf-rtcweb-audio-10.txt> (WebRTC Audio Codec and Processing Requirements) to Proposed Standard

["Asveren, Tolga" <[e-mail]>]

i- I think w3org should have followed the lead of IETF in this issue rather than the other way around, i.e. the values recommended by the IETF specification should have been cited in the w3org document IMHO.

ii- The reasonable value range could depend on the negotiated codec and that would be known at the time of interesting the digits; so anything with MUST strength is too restrictive IMHO.

iii- The presence of transcoding/interworking (between different forms of digit transfer) devices (they will be there, whether we like it or not, for certain scenarios) makes it even less desirable to have MUST strength mandates.

iv- I think adding some text regarding gap/duration of digit packets could be fine but I rather would prefer it with "recommend" (even not RECOMMEND) (and providing some values only as examples).

Thanks,
Tolga

From: Harald Alvestrand [mailto:[e-mail]]
Sent: Friday, February 26, 2016 2:38 AM
To: Roman Shpount <[e-mail]>; Asveren, Tolga <[e-mail]>
Cc: [e-mail]
Subject: Re: [rtcweb] Fwd: Last Call: <draft-ietf-rtcweb-audio-10.txt> (WebRTC Audio Codec and Processing Requirements) to Proposed Standard

Roman, you're the expert here - googling for the text unearthed this message from you in 2013:

https://lists.w3.org/Archives/Public/public-webrtc/2013Apr/0030.html

Reviewiing that archive might save you the effort of repeating the arguments again; at the moment, I have no idea where the upper limit came from.


On 02/25/2016 09:58 PM, Roman Shpount wrote:
Tolga,

In the proposed text I have copied the requirements from  https://www.w3.org/TR/webrtc/#rtcdtmfsender to make sure specifications are consistent with each other. Here is the text present in WebRTC TR:

The duration parameter indicates the duration in ms to use for each character passed in the tones parameters. The duration cannot be more than 6000 ms or less than 40 ms. The default duration is 100 ms for each tone.

The interToneGap parameter indicates the gap between tones. It must be at least 30 ms. The default value is 70 ms.

I believe the limits in WebRTC TR are based on ITU-T Q.24, which are, in turn, based on physical limitation imposed by transmitting DTMF tones using 8Khz audio. It is physically impossible to design an inband DTMF detector which can detect DTMF digits of duration less then 40 ms with gaps less then 30 ms and still be capable of handling talk off. This requirement is the MUST level to provide minimal support necessary for PSTN interop.

Regards,
_____________
Roman Shpount

On Thu, Feb 25, 2016 at 2:44 PM, Asveren, Tolga <[e-mail]<mailto:[e-mail]>> wrote:

Roman,



What is the rationale of having this "MUST" strength mandate?



Thanks,

Tolga

________________________________
From: rtcweb <[e-mail]<mailto:[e-mail]>> on behalf of Ted Hardie <[e-mail]<mailto:[e-mail]>>
Sent: Thursday, February 25, 2016 1:07 PM
To: [e-mail]<mailto:[e-mail]>
Subject: [rtcweb] Fwd: Last Call: <draft-ietf-rtcweb-audio-10.txt> (WebRTC Audio Codec and Processing Requirements) to Proposed Standard

Mailman claims this was auto-discarded and it doesn't show in the archives, so I'm forwarding.  I will separately check on why it got chucked.
regards,
Ted
---------- Forwarded message ----------
From: Jean-Marc Valin <[e-mail]<mailto:[e-mail]>>
Date: Wed, Feb 24, 2016 at 2:21 PM
Subject: Re: [rtcweb] Last Call: <draft-ietf-rtcweb-audio-10.txt> (WebRTC Audio Codec and Processing Requirements) to Proposed Standard
To: Roman Shpount <[e-mail]<mailto:[e-mail]>>, [e-mail]<mailto:[e-mail]>, [e-mail]<mailto:[e-mail]>, "[e-mail]<mailto:[e-mail]>" <[e-mail]<mailto:[e-mail]>>, [e-mail]<mailto:[e-mail]>


On 02/24/2016 04:42 PM, Roman Shpount wrote:
> Generated events MUST have duration of no more than 6000 ms and no
> less than 40 ms with the recommended default duration of 100 ms for each
> tone. The gap between events MUST be no less then 30 ms with the
> recommended default duration of 70 ms.

OK, I missed that part, but I'm fine with it. Unless anyone objects, I
can add it to the draft.

        Jean-Marc


> Regards,
> _____________
> Roman Shpount
>
> On Wed, Feb 24, 2016 at 4:31 PM, The IESG <[e-mail]<mailto:[e-mail]>
> <mailto:[e-mail]<mailto:[e-mail]>>> wrote:
>
>
>     The IESG has received a request from the Real-Time Communication in
>     WEB-browsers WG (rtcweb) to consider the following document:
>     - 'WebRTC Audio Codec and Processing Requirements'
>       <draft-ietf-rtcweb-audio-10.txt> as Proposed Standard
>
>     The IESG plans to make a decision in the next few weeks, and solicits
>     final comments on this action. Please send substantive comments to the
>     [e-mail]<mailto:[e-mail]> <mailto:[e-mail]<mailto:[e-mail]>> mailing lists by 2016-03-09.
>     Exceptionally, comments may be
>     sent to [e-mail]<mailto:[e-mail]> <mailto:[e-mail]<mailto:[e-mail]>> instead. In either
>     case, please retain the
>     beginning of the Subject line to allow automated sorting.
>
>     Abstract
>
>
>        This document outlines the audio codec and processing requirements
>        for WebRTC endpoints.
>
>
>
>
>     The file can be obtained via
>     https://datatracker.ietf.org/doc/draft-ietf-rtcweb-audio/
>
>     IESG discussion can be tracked via
>     https://datatracker.ietf.org/doc/draft-ietf-rtcweb-audio/ballot/
>
>
>     No IPR declarations have been submitted directly on this I-D.
>
>
>     _______________________________________________
>     rtcweb mailing list
>     [e-mail]<mailto:[e-mail]> <mailto:[e-mail]<mailto:[e-mail]>>
>     https://www.ietf.org/mailman/listinfo/rtcweb
>
>


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