Re: [rtcweb] On ptimes and maxptimes
Cullen Jennings <fluffy@iii.ca> Sun, 23 March 2014 01:39 UTC
Return-Path: <fluffy@iii.ca>
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 2F9711A092A for <rtcweb@ietfa.amsl.com>; Sat, 22 Mar 2014 18:39:37 -0700 (PDT)
X-Virus-Scanned: amavisd-new at amsl.com
X-Spam-Flag: NO
X-Spam-Score: 2.842
X-Spam-Level: **
X-Spam-Status: No, score=2.842 tagged_above=-999 required=5 tests=[BAYES_50=0.8, DATE_IN_PAST_06_12=1.543, J_CHICKENPOX_15=0.6, J_CHICKENPOX_18=0.6, RCVD_IN_DNSWL_LOW=-0.7, SPF_HELO_PASS=-0.001] autolearn=no
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 5sfbww2YM1ox for <rtcweb@ietfa.amsl.com>; Sat, 22 Mar 2014 18:39:35 -0700 (PDT)
Received: from mxout-07.mxes.net (mxout-07.mxes.net [216.86.168.182]) by ietfa.amsl.com (Postfix) with ESMTP id 3EB321A0762 for <rtcweb@ietf.org>; Sat, 22 Mar 2014 18:39:35 -0700 (PDT)
Received: from [10.21.74.45] (unknown [128.107.239.233]) (using TLSv1 with cipher AES128-SHA (128/128 bits)) (No client certificate requested) by smtp.mxes.net (Postfix) with ESMTPSA id 8EDFC22E255; Sat, 22 Mar 2014 21:39:31 -0400 (EDT)
Content-Type: text/plain; charset="us-ascii"
Mime-Version: 1.0 (Mac OS X Mail 7.2 \(1874\))
From: Cullen Jennings <fluffy@iii.ca>
In-Reply-To: <532BF5F4.90901@ericsson.com>
Date: Sat, 22 Mar 2014 12:39:23 -0600
Content-Transfer-Encoding: quoted-printable
Message-Id: <2ED6AAC7-5D6F-48C1-BCAE-EA161C6BC0CF@iii.ca>
References: <CAOJ7v-1e+FXQZrvh8Q549kwyhf8JX2BA_0q=+6mjha9+o-Dgpw@mail.gmail.com> <531DE104.2010908@ericsson.com> <CAOJ7v-31-xHGq1ERYYx5mvZuqTor0Seod_ChNT2AhY2BLaJi-Q@mail.gmail.com> <532AC890.7000602@ericsson.com> <CAOJ7v-2=p8LBjVK289usieiDFsmf6m3N8rYMNpBFPhDzdb_61Q@mail.gmail.com> <532BF5F4.90901@ericsson.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
X-Mailer: Apple Mail (2.1874)
Archived-At: http://mailarchive.ietf.org/arch/msg/rtcweb/BHkqb3_gYAKq53kACyGL0fDin84
Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] On ptimes and maxptimes
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: <http://www.ietf.org/mail-archive/web/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: Sun, 23 Mar 2014 01:39:37 -0000
On Mar 21, 2014, at 2:19 AM, Magnus Westerlund <magnus.westerlund@ericsson.com> wrote: > On 2014-03-20 17:39, Justin Uberti wrote: >> >> >> >> On Thu, Mar 20, 2014 at 3:53 AM, Magnus Westerlund >> <magnus.westerlund@ericsson.com <mailto:magnus.westerlund@ericsson.com>> >> wrote: >> >> On 2014-03-20 01:27, Justin Uberti wrote: >>> >>> >>> >>> On Mon, Mar 10, 2014 at 8:57 AM, Magnus Westerlund >>> <magnus.westerlund@ericsson.com >> <mailto:magnus.westerlund@ericsson.com> >> <mailto:magnus.westerlund@ericsson.com >> <mailto:magnus.westerlund@ericsson.com>>> >>> wrote: >>> >>> Hi, >>> >>> (As individual) >>> >>> On 2014-03-05 17:26, Justin Uberti wrote: >>>> From the session this morning, I recall this consensus: >>>> >>>> * We will add recommendations on the frame sizes to use for >>> PCMU/A to >>>> the rtcweb audio codec I-D (20, 30, 60 ms frames). >>> >>> The above applies to sending payloads >>> >>> >>> Right, thanks for pointing that out. >>> >>> >>> Yes, but that was given that a receiver will be capable of >> receiving any >>> valid number of frames, or samples within a reasonable >> packetization, >>> and I think that should be 120 ms. >>> >>>> * We will consider adding an a=maxptime attribute that >>> represents the >>>> "minimum maxptime" of all the offered codecs. That is, >> if both >>> Opus >>>> (maxptime of 120ms) and PCMU (maxptime of 60) are offered, a >>>> maxptime=60 SHOULD be included. >>>> o Since maxptime spans all codecs, this means that >> some modes >>>> (e.g. Opus 120ms) will be unavailable, unless only >> Opus is >>> offered. >>> >>> I think you base the assumption on maxptime based on the sending >>> recommendations, not what is capable of receiving. If one is >> capable of >>> receiving 200 ms of a-law and Opus, then you would need to say >> 120 ms as >>> Opus will limit you. >>> >>> >>> I agree that Opus' maxptime is 120, but should that prevent >> sending PCMU >>> with 200 ms? That is, since sending Opus with 200 ms is not legal, is >>> there any point in forcing WebRTC impls to indicate this in SDP? >> >> What I wished we fixed this issue with maxptime a long time ago. This is >> a well known problem that it really has PT scope, but are signalled on >> m= line scope. >> >>> >>> >>> >>>> * We will consider adding an a=ptime attribute, set to the >>> receiver's >>>> preferred frame size to receive. Again, this value spans all >>> codecs, >>>> so the specified value may not be viable for all codecs >> (e.g. >>> 30 ms >>>> works for PCMU but not for Opus) >>>> >>>> After thinking about this some, I'm not sure the >> maxptime/ptime values >>>> will lead to any different behavior than if they were not >> specified. >>>> Since there is a complexity cost (especially if the values >> need to >>>> change based on which codecs are offered), is there a clear >> upside >>> here? >>> >>> The maxptime values may in some cases be limited downwards for >>> interoperability, for example if one like to gateway AMR into a CS >>> network then the gateway might indicate a maxptime that is >> less, more >>> like 20 or 40 ms. >>> >>> >>> Sure - WebRTC impls need to respect any maxptime value that they >>> receive, but it's not clear to me that they need to include one in the >>> SDP they send. >> >> Ok, but then we are back to my previous argument for writing down what >> receiver requirement in regards to each payload type one is required to >> support in the audio draft for the ones specified there. But what about >> the other codecs that aren't included in the audio draft? >> >> How can a peer know if there are limitations? I wouldn't attempt to >> solve the general problem of the scope issue with maxptime. Simply state >> that you should include this. It is going to be limited but still >> provide information in many cases that is quite useful for the sender. >> >> >> Just to be sure I understand, your position is that WebRTC impls SHOULD >> include an a=maxptime attribute indicating the "minimum maximum" frame >> size they can receive, across all codecs? > > Yes, as long as we lack payload type specific, I think that is the best > we can do. >> >> Do you think anything needs to be included regarding a=ptime? > > I think it MAY be included to indicate an initial preference for > packetization interval. In cases that is not common across the set of > payload types for that m= section, then one should skip it. > > Cheers > > Magnus Westerlund It seems like from interoperability point of view, we also need, if a browser receives a maxptime or prime, it MUST be able to understand that.
- [rtcweb] On ptimes and maxptimes Justin Uberti
- Re: [rtcweb] On ptimes and maxptimes Magnus Westerlund
- Re: [rtcweb] On ptimes and maxptimes Justin Uberti
- Re: [rtcweb] On ptimes and maxptimes Magnus Westerlund
- Re: [rtcweb] On ptimes and maxptimes Justin Uberti
- Re: [rtcweb] On ptimes and maxptimes Magnus Westerlund
- Re: [rtcweb] On ptimes and maxptimes Cullen Jennings
- Re: [rtcweb] On ptimes and maxptimes Randell Jesup
- Re: [rtcweb] On ptimes and maxptimes Magnus Westerlund