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" <[email protected]> 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: >> >> >> >> >> >> >> >> >> >> 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" <[email protected]> wrote: >> >>> Oliver Hohlfeld >>> <[email protected]> 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 >>> [email protected] >> >> _______________________________________________ >> Bloat mailing list >> [email protected] >> https://lists.bufferbloat.net/listinfo/bloat >> _______________________________________________ Bloat mailing list [email protected] https://lists.bufferbloat.net/listinfo/bloat
