Subject: [rtcweb] #1: RFC 4734 citation

From: "rtcweb issue tracker" <[email-masked]>
Date: Mon, 22 Oct 2012 22:36:14 -0000

#1: RFC 4734 citation

 Section 3 states:

    WebRTC clients are REQUIRED to implement the following audio codecs.

    o  Opus [RFC6716], with any ptime value up to 120 ms

    o  G.711 PCMA and PCMU with one channel, a rate of 8000 Hz and a
       ptime of 20 - see section 4.5.14 of [RFC3551]

    o  Telephone Event - [RFC4734]

 [BA] I believe the citation should be to "RTP Payload for DTMF Digits,
 Telephony Tones, and Telephony Signals" [RFC4733], not to "Definition of
 Events for Modem, Fax, and Text Telephony Signals" [RFC4734].

 If the desire is in fact to mandate support for RFC 4734, please explain
 why it is necessary for browsers to include support for digital encoding
 of modem and fax signals.

-- 
--------------------------------+------------------------
 Reporter:  bernard_aboba@…     |      Owner:  jmvalin@…
     Type:  defect              |     Status:  new
 Priority:  critical            |  Milestone:  milestone1
Component:  audio               |    Version:  1.0
 Severity:  Active WG Document  |   Keywords:
--------------------------------+------------------------

Ticket URL: <http://trac.tools.ietf.org/wg/rtcweb/trac/ticket/1>
rtcweb <http://tools.ietf.org/rtcweb/>

---

From: Randell Jesup <[email-masked]>
Date: Tue, 23 Oct 2012 09:37:29 -0400

>   of modem and fax signals.

I'm certain the intent was RFC 4733.

-- 
Randell Jesup
[email-masked]

---

From: Iñaki Baz Castillo <[email-masked]>
Date: Tue, 30 Oct 2012 14:06:33 +0100

2012/10/23 rtcweb issue tracker <[email-masked]>:
>  Telephone Event - [RFC4734]

Could somebody give some points about why DTMFs over media are
required in WWW? Isn't DTMF mechanism OLD enough to incorporate it
into WebRTC?

Autoanswer: interoperability with legacy telephony world.

-- 
Iñaki Baz Castillo
<[email-masked]>

---

From: Matthew Kaufman <[email-masked]>
Date: Tue, 30 Oct 2012 13:19:29 +0000

> From: [email-masked] [mailto:[email-masked]] On
> Behalf Of Iñaki Baz Castillo
>
> required in WWW? Isn't DTMF mechanism OLD enough to incorporate it into
> WebRTC?
> 

The issue is that because the DTMF is on the RTP media stream (and not the signaling channel) then for any interworking with legacy telephony equipment, radio systems, or other things that use DTMF it is necessary for the sending browser to do the insertion of DTMF.

Matthew Kaufman

---

From: "rtcweb issue tracker" <[email-masked]>
Date: Tue, 30 Oct 2012 13:54:07 -0000

#1: RFC 4734 citation

Comment (by fluffy@…):

 This is a bug on draft-ietf-rtcweb-audio-00.txt

-- 
--------------------------------+-------------------------
 Reporter:  bernard_aboba@…     |       Owner:  jmvalin@…
     Type:  defect              |      Status:  new
 Priority:  critical            |   Milestone:  milestone1
Component:  audio               |     Version:  1.0
 Severity:  Active WG Document  |  Resolution:
 Keywords:                      |
--------------------------------+-------------------------

Ticket URL: <http://trac.tools.ietf.org/wg/rtcweb/trac/ticket/1#comment:1>
rtcweb <http://tools.ietf.org/rtcweb/>

---

From: Harald Alvestrand <[email-masked]>
Date: Wed, 31 Oct 2012 16:23:28 +0100

> Autoanswer: interoperability with legacy telephony world.
Autoanswer (and you should know this by heart now, it's been said so 
many times):

Use case 4.3.2 from draft-ietf-rtcweb-use-cases-and-requirements-09:

4.3.2.  Fedex Call

4.3.2.1.  Description

    Alice uses her web browser with a service something like Skype to be
    able to phone PSTN numbers.  Alice calls 1-800-gofedex.  Alice should
    be able to hear the initial prompts from the fedex IVR and when the
    IVR says press 1, there should be a way for Alice to navigate the
    IVR.

If you want to argue for having that use case deleted from the 
requirements, start a debate on that topic. If you have an alternative 
solution, state that in the positive as what should go into the -audio- 
draft instead of RFC 4734.

---

From: Iñaki Baz Castillo <[email-masked]>
Date: Wed, 31 Oct 2012 18:12:43 +0100

2012/10/31 Harald Alvestrand <[email-masked]>

>> 2012/10/23 rtcweb issue tracker <[email-masked].**org<[email-masked]>
>> >:
>>
> Use case 4.3.2 from draft-ietf-rtcweb-use-cases-**and-requirements-09:
>
> 4.3.2.  Fedex Call
>
> 4.3.2.1.  Description
>

Well, as we all assume that media gateways are requried for
interoperability with other VoIP networks, I assume it would be not so hard
that the gateway receives the "DTMF's" via WWW means (HTTP, WebSocket...)
and generates RTP DTMF's in the legacy leg.

Anyhow I expect that it's not so hard to implement RTP DTMF's in a media
stack, but I would not like to hear about "my PBX's does not detect
Chrome's DTMF's". :)

-- 
Iñaki Baz Castillo
<[email-masked]>

---

From: Bernard Aboba <[email-masked]>
Date: Wed, 31 Oct 2012 10:20:46 -0700

RFC 4734 relates to modem and fax events, not DTMF. citation should have been to 4733. 

On Oct 31, 2012, at 8:23, "Harald Alvestrand" <[email-masked]> wrote:

> Autoanswer (and you should know this by heart now, it's been said so many times):
> 
> If you want to argue for having that use case deleted from the requirements, start a debate on that topic. If you have an alternative solution, state that in the positive as what should go into the -audio- draft instead of RFC 4734.
> _______________________________________________
> rtcweb mailing list
> [email-masked]
> https://www.ietf.org/mailman/listinfo/rtcweb

---

From: Roman Shpount <[email-masked]>
Date: Wed, 31 Oct 2012 13:41:46 -0400

> stack, but I would not like to hear about "my PBX's does not detect

We had an extensive discussion few months back on why RFC 4733 tones are
required. The main reason is time synchronization of audio and DTMF. If you
have an application which does something like "record your name and press
pound", and DTMF is delivered over a separate path, you end up with name
either truncated or padded with silence. In any case, I believe, there was
a group consensus that RFC 4733 DTMF tones will be supported.
_____________
Roman Shpount

---

From: Bernard Aboba <[email-masked]>
Date: Wed, 31 Oct 2012 10:48:43 -0700

Right. But somehow RFC 4734 was cited instead.

On Oct 31, 2012, at 10:41 AM, "Roman Shpount" <[email-masked]> wrote:

>> Well, as we all assume that media gateways are requried for interoperability with other VoIP networks, I assume it would be not so hard that the gateway receives the "DTMF's" via WWW means (HTTP, WebSocket...) and generates RTP DTMF's in the legacy leg.
>> 
>> Anyhow I expect that it's not so hard to implement RTP DTMF's in a media stack, but I would not like to hear about "my PBX's does not detect Chrome's DTMF's". :)
> 
> We had an extensive discussion few months back on why RFC 4733 tones are required. The main reason is time synchronization of audio and DTMF. If you have an application which does something like "record your name and press pound", and DTMF is delivered over a separate path, you end up with name either truncated or padded with silence. In any case, I believe, there was a group consensus that RFC 4733 DTMF tones will be supported.
> _____________
> Roman Shpount
>  
>