[Bloat] [Cerowrt-devel] FQ_Codel lwn draft article review

[Simon Barber]

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:
>
> 	
> 		
> 		
> 	
> 	
> 		
> 			
> 				
> 					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:
> 	
> 		
> 		
> 	
> 	
> 		
> 			
> 				
> 					R= 94.2 - 0.024*Latency - 0.11*max(0,Latency-177.3 ms) - 30*log(1 +
> 15*Loss).
>
> 				
> 			
> 		
> 	
> 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.
>
>
> 				
> 			
> 		
> 	
> 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 Høiland-Jørgensen" <toke at toke.dk> wrote:
>
>> Oliver Hohlfeld
>> <oliver at net.t-labs.tu-berlin.de> 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 Høiland-Jørgensen
>> toke at toke.dk
>
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