From mboxrd@z Thu Jan 1 00:00:00 1970 Return-Path: Received: from ondar.cablelabs.com (ondar.cablelabs.com [192.160.73.61]) by huchra.bufferbloat.net (Postfix) with ESMTP id 90AF821F15E for ; Wed, 28 Nov 2012 13:39:59 -0800 (PST) Received: from kyzyl.cablelabs.com (kyzyl [10.253.0.7]) by ondar.cablelabs.com (8.14.5/8.14.5) with ESMTP id qASLdv4c028919; Wed, 28 Nov 2012 14:39:57 -0700 Received: from srvxchg.cablelabs.com (10.5.0.15) by kyzyl.cablelabs.com (F-Secure/fsigk_smtp/407/kyzyl.cablelabs.com); Wed, 28 Nov 2012 14:39:57 -0700 (MST) X-Virus-Status: clean(F-Secure/fsigk_smtp/407/kyzyl.cablelabs.com) Received: from srvxchg.cablelabs.com ([10.5.0.15]) by srvxchg ([10.5.0.15]) with mapi; Wed, 28 Nov 2012 14:39:58 -0700 From: Greg White To: Simon Barber Date: Wed, 28 Nov 2012 14:40:11 -0700 Thread-Topic: [Bloat] [Cerowrt-devel] FQ_Codel lwn draft article review Thread-Index: Ac3NsOnNteXYqJQZSY+s2lVYLfVFWA== Message-ID: In-Reply-To: <50B669E5.8020105@superduper.net> Accept-Language: en-US Content-Language: en-US X-MS-Has-Attach: X-MS-TNEF-Correlator: user-agent: Microsoft-MacOutlook/14.2.5.121010 acceptlanguage: en-US Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: quoted-printable MIME-Version: 1.0 Cc: =?iso-8859-1?Q?Toke_H=F8iland-J=F8rgensen?= , "bloat@lists.bufferbloat.net" Subject: Re: [Bloat] [Cerowrt-devel] FQ_Codel lwn draft article review X-BeenThere: bloat@lists.bufferbloat.net X-Mailman-Version: 2.1.13 Precedence: list List-Id: General list for discussing Bufferbloat List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , X-List-Received-Date: Wed, 28 Nov 2012 21:40:00 -0000 Good point. I wrote that without double checking my code. What I should've said was: Latency is min_latency + 60ms. This is under the assumption of a fixed 60ms dejitter buffer that is prescient as to what the min_latency will be. -Greg On 11/28/12 12:45 PM, "Simon Barber" wrote: >Not sure why you would use mean latency - for VoIP the thing that >matters is maximum latency - if some packets are late, then you don't >hear them. More precisely a small packet loss rate is OK, so you should >use something like the 99th percentile of latency. In my last jitter >buffer implementation I ran an estimator for 99th percentile latency, >and added 5ms. > >That said, most jitter buffers in use are poor implementations, and >introduce more latency than necessary. > >Simon > > >On 11/28/2012 11:21 AM, Greg White wrote: >> Jitter on its own is not useful for estimating VoIP quality. >> >> For (c), I've used: >> >> =09 >> =09 >> =09 >> =09 >> =09 >> =09 >> =09 >> =09 >> Cole, Robert G., and Joshua H. Rosenbluth. "Voice over IP >>performance >> monitoring." ACM SIGCOMM Computer Communication Review 31, no. 2 (2001): >> 9-24. >> >> Which I generally simplify to: >> =09 >> =09 >> =09 >> =09 >> =09 >> =09 >> =09 >> =09 >> R=3D 94.2 - 0.024*Latency - 0.11*max(0,Latency-177.3 ms) - 30*log(1= + >> 15*Loss). >> >> =09 >> =09 >> =09 >> =09 >> where Loss is mean packet loss rate (counting packets with latency > >> (min_latency + 60ms) as lost), Latency is mean latency for packets not >> counted as lost. >> >> >> =09 >> =09 >> =09 >> =09 >> You can then translate R into an estimate of MOS using Eq.1 in the above >> reference. >> >> -Greg >> >> >> On 11/27/12 5:51 PM, "Toke H=F8iland-J=F8rgensen" wrote: >> >>> Oliver Hohlfeld >>> writes: >>> >>>> The jitter measurements you have in mind will give you an idea on the >>>> jitter specific to the chosen traffic scenario, nothing more --- and >>>> in particular not the VoIP quality (although low vs. high jitter could >>>> /indicate/ certain /possible/ quality degradations). >>> >>> Well no, in this sense the only "real" test for voip quality is picking >>> up the (soft)phone and talking to someone. However, since the context >>> here is automated measuring tools (preferably generating solid >>> quantitative, comparable data), that is hardly feasible. >>> >>> I guess the goal of a comprehensive testing suite is to gather as many >>> indicators of quality degradations (in the widest possible sense) as >>> possible and testing for them under a variety of traffic conditions. I >>> am by no means an expert on VoIP, but someone suggested measuring >>>jitter >>> could be useful, and I've proposed a possible way to do that (using >>> iperf udp flows at a low-ish bandwidth). >>> >>> Since for the purpose of this particular discussion I seem to be in the >>> test tool building business (at least for the time being), what I >>>really >>> need before going forward with this is someone to comment on (a) if >>> using iperf udp flows is a valid way to measure jitter, (b) if >>>measuring >>> jitter is actually something someone wants to do and (c) if there are >>> other tests that would be useful for testing VoIP (or general) >>> conditions instead of / in addition to the jitter measurements. >>> >>> So far I don't have an answer to (a), only negative answers to (b) and >>> nothing concrete for (c). So for the time being I'm shelving the idea, >>> and will just note that it seems quite feasible to return to it should >>> someone change their mind on (b) :) >>> >>> >>> -Toke >>> >>> -- >>> Toke H=F8iland-J=F8rgensen >>> toke@toke.dk >> >> _______________________________________________ >> Bloat mailing list >> Bloat@lists.bufferbloat.net >> https://lists.bufferbloat.net/listinfo/bloat >>