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

Re: Min DTMF Gap

From: Barry Dingle <btdingle@gmail.com>
Date: Mon, 20 Jan 2014 22:17:14 +1100
Message-ID: <CAN=GVAvRhjW9UEKxCMi8o+YhRLTYZ4mKC0TnitVPT_TKD27auQ@mail.gmail.com>
To: Gunnar Hellstrom <gunnar.hellstrom@omnitor.se>
Cc: Roman Shpount <roman@telurix.com>, "Cullen Jennings (fluffy)" <fluffy@cisco.com>, Mike Johns <m.johns@commsalliance.com.au>, "public-webrtc@w3.org" <public-webrtc@w3.org>

I agree with you and Roman. What you suggest appears to cover the
Australian DTMF case. My only hesitation is that I cannot reply on behalf
of Australian Industry. (Everyone 'in authority' is on Summer Leave at the
moment !!)

/Barry Dingle

 Gunnar Hellstrom <gunnar.hellstrom@omnitor.se> wrote:

>  On 2014-01-19 20:06, Roman Shpount wrote:
>  On Fri, Jan 17, 2014 at 12:44 PM, Roman Shpount <roman@telurix.com>wrote:
>>   On Fri, Jan 17, 2014 at 12:35 PM, Cullen Jennings (fluffy) <
>> fluffy@cisco.com> wrote:
>>> I’m 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
> 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 11:18:04 UTC

This archive was generated by hypermail 2.4.0 : Friday, 17 January 2020 19:17:54 UTC