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

Tim Panton <tim@phonefromhere.com> Mon, 07 March 2016 11:10 UTC

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From: Tim Panton <tim@phonefromhere.com>
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Subject: Re: [rtcweb] Last Call: <draft-ietf-rtcweb-audio-10.txt> (WebRTC Audio Codec and Processing Requirements) to Proposed Standard
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> On 7 Mar 2016, at 10:47, Tim Panton <tim@phonefromhere.com> wrote:
> 
> 
>> On 4 Mar 2016, at 19:55, Roman Shpount <roman@telurix.com> wrote:
>> 
>> On Fri, Mar 4, 2016 at 5:34 AM, Asveren, Tolga <tasveren@sonusnet.com> wrote:
>> i- I don't think satisfactory technical explanation is provided why "enforcing a range" is superior to a "default range". I will leave this here and obviously this is just my personal -but honest- opinion.
>> 
> …. snip…. 
>> It probably makes sense to note, that support for such application is impossible from the current WebRTC API which only allows to play DTMF tone of predefined length. To implement this application you need start play tone/stop play tone API which is not provided.
>> 
>> - Remote valve/door/lock control
>> Authorized person makes a call and then continuously presses "6" to keep the valve/door/lock open for that period.
>> 
>> Same not as above applies here.
>> 
>> - Panic button
>> An application for women who are mistreated by their husbands. Police/Social service workers rush to the registered address if she presses "7" continuously for longer than 10 seconds.
>> 
>> The "long press" for DTMF one is typically 1-4 sec. Most of current PSTN networks will not pass a tone longer then 8 sec as a single tone due to RFC 2833 tone duration limitation.
>> 
>> I am just trying to show that there are many different/imaginative ways using the "digit UI".
>> 
>> Unfortunately application you are referring to are unimplementable using WebRTC.
> 
> Actually they are all implementable with webRTC - if combined with webAudio.
> You just generate the DTMF tones with webAudio, mix (as appropriate)  with the mic signal and then
> inject the result into the outgoing using 
> AudioContext.createMediaStreamDestination()
> and friends.
> 
> We could add a caveat that all the DTMF apis are optional if the browser also implements
> webAudio. As it is possible to write a a javascript polyfill that implements the current api in 
> terms of webaudio, which should be adequate for the legacy compatibility needs.

And as proof by example : https://hacks.mozilla.org/2015/11/webrtc-sending-dtmf-in-firefox/
(thanks to Adam Roach for the code and Philipp Hancke for pointing it out)- 
I’d done a simpler version some years ago in phonoSDK - 
https://github.com/phono/PhonoSDK/commit/d6bb7ca54402aba3c0dd34dbd1912731f5af111b


> 
> Also we should add a note that _all_ these sorts of usages would be better served by
> carrying the data over the DataChannel, DTMF is for legacy interop only.
> 
> Tim.