W3C home > Mailing lists > Public > public-webrtc@w3.org > January 2014

Re: Min DTMF Gap

From: Gunnar Hellstrom <gunnar.hellstrom@omnitor.se>
Date: Mon, 20 Jan 2014 08:00:31 +0100
Message-ID: <52DCC98F.6050306@omnitor.se>
To: Roman Shpount <roman@telurix.com>, "Cullen Jennings (fluffy)" <fluffy@cisco.com>
CC: Barry Dingle <btdingle@gmail.com>, Mike Johns <m.johns@commsalliance.com.au>, "public-webrtc@w3.org" <public-webrtc@w3.org>
On 2014-01-19 20:06, Roman Shpount wrote:
> On Fri, Jan 17, 2014 at 12:44 PM, Roman Shpount <roman@telurix.com 
> <mailto:roman@telurix.com>> wrote:
>
>     On Fri, Jan 17, 2014 at 12:35 PM, Cullen Jennings (fluffy)
>     <fluffy@cisco.com <mailto:fluffy@cisco.com>> wrote:
>
>
>         Iím fine with lower limits allowing people to shoot themselves
>         in the feet but I want the defaults to be safe for most cases.
>
>         So the way I think we should set this is to set the default to
>         be "safe" for all major deployments world wide.  And have the
>         minimum values allow you set it to be as low as is usable in
>         any any major deployment world wide. With that strategy, and
>         the information folks provided in this email thread, I think
>         we get to the following.
>
>         How about this for a proposed change:
>
>         We change the min tone time to 40 ms.
>
>         We change the min gap time to 30 ms.
>
>         We change the default gap to 70 ms (this meets Australia AS/CA
>         S0020)
>
>         We leave the default tone duration at 100 ms.
>
>         Does that change look OK to folks?
>
>
>     This looks perfect to me.
>
>
> After I thought about this a little bit more I would suggest that tone 
> and gap generation should take the packet ptime into account. The 
> reason is that WebRTC will switch between RFC 4733 tone and audio 
> codec during the pause. This means for 70 ms pauses and most common 
> ptime of 20 ms we will end up with a 10 ms audio packet. Packets of 
> duration other then ptime have a tendency to cause interop issues. In 
> our implementations of similar DTMF generation code we would round up 
> gap and tone duration to the next ptime interval, so that with 20 ms 
> ptime you can set the gap to 40, 60, 80, 100 ms (etc) and tone 
> duration to 40, 60, 80, 100 ms (etc). With 30 ms ptime it would be 30, 
> 60, 90 ms (etc) for the gap and 60, 90, 120 ms (etc) for the tone.
Yes, this seems right.
The WebRTC API spec should then just mention the rounding of the 
durations and alignment with the ptime, and the IETF RTCWEB audio spec 
should be specific about how to do the interleaving and relative timing 
of audio and DTMF.

So, for the API spec:
Default duration is 100 ms and default gap is 70 ms. The transmission 
timing and duration used will be aligned by extension if needed to use 
the same packet transmission time and interval as the audio stream.

And the details in an RTCWEB spec.

/Gunnar
> _____________
> Roman Shpount
>
Received on Monday, 20 January 2014 07:01:18 UTC

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