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

[Jean-Marc Valin <[e-mail]>]

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Hi Roman,

Just as a clarification, are we talking about the duration of RFC4733
tone events or the duration of audio tones (or both)? I agree that
going below 40 ms is just not possible. As for the upper bound, more
than 6000 ms is definitely insane, but I have no precise opinion on
the exact threshold that should be used, or why anyone would want such
long tones in the first place.

Cheers,

	Jean-Marc

On 02/25/2016 03:58 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
> 
> 
> 
> 
> _______________________________________________ rtcweb mailing
> list [e-mail] https://www.ietf.org/mailman/listinfo/rtcweb
> 
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