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

[Harald Alvestrand <[e-mail]>]

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
>     >
>     >
>
>
>     _______________________________________________
>     rtcweb mailing list
>     [e-mail] <mailto:[e-mail]>
>     https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
>
> _______________________________________________
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> https://www.ietf.org/mailman/listinfo/rtcweb