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

Harald Alvestrand <harald@alvestrand.no> Sat, 27 February 2016 13:11 UTC

Return-Path: <harald@alvestrand.no>
X-Original-To: rtcweb@ietfa.amsl.com
Delivered-To: rtcweb@ietfa.amsl.com
Received: from localhost (ietfa.amsl.com [127.0.0.1]) by ietfa.amsl.com (Postfix) with ESMTP id D35F81AC416 for <rtcweb@ietfa.amsl.com>; Sat, 27 Feb 2016 05:11:35 -0800 (PST)
X-Virus-Scanned: amavisd-new at amsl.com
X-Spam-Flag: NO
X-Spam-Score: -4.206
X-Spam-Level:
X-Spam-Status: No, score=-4.206 tagged_above=-999 required=5 tests=[BAYES_00=-1.9, RCVD_IN_DNSWL_MED=-2.3, RP_MATCHES_RCVD=-0.006] autolearn=ham
Received: from mail.ietf.org ([4.31.198.44]) by localhost (ietfa.amsl.com [127.0.0.1]) (amavisd-new, port 10024) with ESMTP id n-IHKQv7geqp for <rtcweb@ietfa.amsl.com>; Sat, 27 Feb 2016 05:11:33 -0800 (PST)
Received: from mork.alvestrand.no (mork.alvestrand.no [158.38.152.117]) (using TLSv1.2 with cipher AECDH-AES256-SHA (256/256 bits)) (No client certificate requested) by ietfa.amsl.com (Postfix) with ESMTPS id 0BB051A1B69 for <rtcweb@ietf.org>; Sat, 27 Feb 2016 05:11:33 -0800 (PST)
Received: from localhost (localhost [127.0.0.1]) by mork.alvestrand.no (Postfix) with ESMTP id 080477C675E; Sat, 27 Feb 2016 14:11:31 +0100 (CET)
X-Virus-Scanned: Debian amavisd-new at alvestrand.no
Received: from mork.alvestrand.no ([127.0.0.1]) by localhost (mork.alvestrand.no [127.0.0.1]) (amavisd-new, port 10024) with ESMTP id U6Z1jiyByF5a; Sat, 27 Feb 2016 14:11:29 +0100 (CET)
Received: from [IPv6:2001:470:de0a:1:9c91:e863:f283:7d70] (unknown [IPv6:2001:470:de0a:1:9c91:e863:f283:7d70]) by mork.alvestrand.no (Postfix) with ESMTPSA id A664D7C6750; Sat, 27 Feb 2016 14:11:29 +0100 (CET)
To: "Asveren, Tolga" <tasveren@sonusnet.com>, Roman Shpount <roman@telurix.com>
References: <20160224213121.376.85278.idtracker@ietfa.amsl.com> <CAD5OKxsa9cwYOLqkHDVjoe2vr8NoOsPYO7jD_4TPNSnxU7u53Q@mail.gmail.com> <56CE2CF4.70001@jmvalin.ca> <CA+9kkMAqNZiHX7asFZnNgMnJw3G2bPBB7zXfLex3xdkfcW2tQQ@mail.gmail.com> <SN1PR0301MB15510A18734956A22BD5FB5AB2A60@SN1PR0301MB1551.namprd03.prod.outlook.com> <CAD5OKxu3HSKDNMNhEWHgoBrHj4zOvjwbGFQSyLmBgLo6cL2Lhg@mail.gmail.com> <56D000EF.9010004@alvestrand.no> <SN1PR0301MB15518B65A2E7D2ACFE2663B4B2A70@SN1PR0301MB1551.namprd03.prod.outlook.com> <CAD5OKxuQT2hdDHWdVxHGEcC3PuMMDjpaBpfAygRBa7-kdv79Rg@mail.gmail.com> <SN1PR0301MB15519E82B0384EF6EC348B72B2B80@SN1PR0301MB1551.namprd03.prod.outlook.com>
From: Harald Alvestrand <harald@alvestrand.no>
Message-ID: <56D1A080.7050901@alvestrand.no>
Date: Sat, 27 Feb 2016 14:11:28 +0100
User-Agent: Mozilla/5.0 (X11; Linux x86_64; rv:38.0) Gecko/20100101 Thunderbird/38.5.1
MIME-Version: 1.0
In-Reply-To: <SN1PR0301MB15519E82B0384EF6EC348B72B2B80@SN1PR0301MB1551.namprd03.prod.outlook.com>
Content-Type: text/plain; charset="utf-8"
Content-Transfer-Encoding: 8bit
Archived-At: <http://mailarchive.ietf.org/arch/msg/rtcweb/rvltruZUQk3846FI-KPM7_m-lHw>
Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Fwd: Last Call: <draft-ietf-rtcweb-audio-10.txt> (WebRTC Audio Codec and Processing Requirements) to Proposed Standard
X-BeenThere: rtcweb@ietf.org
X-Mailman-Version: 2.1.15
Precedence: list
List-Id: Real-Time Communication in WEB-browsers working group list <rtcweb.ietf.org>
List-Unsubscribe: <https://www.ietf.org/mailman/options/rtcweb>, <mailto:rtcweb-request@ietf.org?subject=unsubscribe>
List-Archive: <https://mailarchive.ietf.org/arch/browse/rtcweb/>
List-Post: <mailto:rtcweb@ietf.org>
List-Help: <mailto:rtcweb-request@ietf.org?subject=help>
List-Subscribe: <https://www.ietf.org/mailman/listinfo/rtcweb>, <mailto:rtcweb-request@ietf.org?subject=subscribe>
X-List-Received-Date: Sat, 27 Feb 2016 13:11:36 -0000

Den 27. feb. 2016 11:45, skrev Asveren, Tolga:
> If I don’t mis/over-interpret Roman’s answer, it seems like at least
> some people who really care/have practical experience about this issue,
> e.g. Roman and myself, are in favor of not mandating any values and
> suggesting that w3org specification is updated accordingly. I personally
> would prefer nothing more than a (or two) sentence as a warning without
> using any keywords in rtcweb-audio. Does this sound a reasonable choice
> to other folks?

At the WEBRTC API, this *will* lead to noninteroperable implementations,
since some browsers will enforce different limits from other browsers.

It's all coming back now - we decided to go with fixed limits in the
spec because it was inconcievable that implementations wouldn't impose
*some* limits, and the idea of adding API surface for probing what the
limits were was just too gross for such a low-value (relatively
speaking) feature.


> 
>  
> 
> Thanks,
> 
> Tolga
> 
>  
> 
> *From:*Roman Shpount [mailto:roman@telurix.com]
> *Sent:* Friday, February 26, 2016 4:56 PM
> *To:* Asveren, Tolga <tasveren@sonusnet.com>
> *Cc:* Harald Alvestrand <harald@alvestrand.no>; rtcweb@ietf.org
> *Subject:* Re: [rtcweb] Fwd: Last Call: <draft-ietf-rtcweb-audio-10.txt>
> (WebRTC Audio Codec and Processing Requirements) to Proposed Standard
> 
>  
> 
> On Fri, Feb 26, 2016 at 6:19 AM, Asveren, Tolga <tasveren@sonusnet.com
> <mailto:tasveren@sonusnet.com>> wrote:
> 
>     i- I think w3org should have followed the lead of IETF in this issue
>     rather than the other way around, i.e. the values recommended by the
>     IETF specification should have been cited in the w3org document IMHO.
> 
>  
> 
> I agree completely. I am not aware of any IETF document that defines
> DTMF or RFC 4733 tone duration limits, so I proposed to add these limits
> to draft-ietf-rtcweb-audio. Most importantly I wanted the text from W3C
> reviewed in IETF since it was clearly a network related. Furthermore,
> anybody implementing WebRTC compatible RTP audio interface should not
> need to read the API document to find the network specific limits.
> 
>  
> 
>     ii- The reasonable value range could depend on the negotiated codec
>     and that would be known at the time of interesting the digits; so
>     anything with MUST strength is too restrictive IMHO.
> 
>  
> 
> We know that RFC 4733 would be used to transmit DTMF tones from WebRTC
> endpoints. RFC 4733 has no upper or lower limits on tone duration, so
> technically these can be set to anything or not set at all. Some people
> argue that we should limit number of foot guns for future API users, so
> they wanted to have reasonable tone duration limits.
> 
>  
> 
>     iii- The presence of transcoding/interworking (between different
>     forms of digit transfer) devices (they will be there, whether we
>     like it or not, for certain scenarios) makes it even less desirable
>     to have MUST strength mandates.
> 
>  
> 
> Unfortunately I spend a significant amount of my time dealing with
> transcoding elements (SBCs) dealing with RFC 4733 tones. Sending tones
> which are too short or sent at high rates make such transcoding elements
> generate unexpected or broken DTMF sequences. Reordered or interleaved
> tones are commonly generated in response to such sequences. Extremely
> long duration DTMF digits typically break into several digits. There is
> danger in not having reasonable limits. The decision if API users should
> be protected from this danger is up to this group.
> 
>  
> 
>     iv- I think adding some text regarding gap/duration of digit packets
>     could be fine but I rather would prefer it with “recommend” (even
>     not RECOMMEND) (and providing some values only as examples).
> 
>  
> 
> I agree that having reasonable recommended values should be sufficient
> for most cases. The group has to decide if it wants to protect the
> developers from themselves and set MUST level limits on tone and gap
> duration.
> 
> _____________
> Roman Shpount
> 
>  
>