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[General] PAP2 v2 and syslog »
« [VoiceStick] SIP Ports  
AuthorAll Replies

ctylor

join:2006-03-15
Vancouver, BC

reply to DracoFelis
Re: [Equipment] Useful Sipura tricks...

said by DracoFelis See Profile :

said by ctylor :

While ctylor found a "Network Jitter Level: low" worked well for him, my tests (with my ISP) showed differently. While it is true that a setting of "low" results in the lowest possible latency/echo (which is "a good thing"), I started having problems with "stuttering sounds" (fraction of a second drop-outs of sound) when I used the "low" setting. So I boosted my setting to "medium" which helped (but didn't eliminate the effect), and eventually put it back to the default of "high" (which works reasonably well with my ISP). Granted, I lost the lower latency/echo that I gained by setting it to "low", but using a setting of "high" also caused me to lose the constant "stuttering" on my VoIP line! So lowering the "Network Jitter Level:" setting is clearly a YMMV setting. It's worthwhile trying, because it will lower latency/echo, but be prepared to set it back up if/when you start getting "stuttering" on the line...

NOTE: Yes, that means that I'm currently using trick A (more bandwidth used, by lowering my RTP packet size), but not trick B (because, in my case, lowering my jitter buffer caused problems with packet loss, which was a worse problem for our family than the latency/echo caused by the default jitter buffer size)...
Hi Draco, were you testing this setting using 'real world' conditions of actual phone calls with people, or mostly with IVR systems and echo tests (like FWD's)? I have difficulty replicating your stuttering/packet discard problem when the Jitter Buffer setting is on "Low" on echo tests, IVRs, or any of the phone calls I've had in the last week.

(Well not entirely true: I had a distinct problem last night when bit torrent was up and running--the echo test was almost comical! The end to end latency increased from very slight to nearly a second and half, and every fourth word (when it would finally arrive two seconds later!) would have a syllable or two get 'dropped'. That is the only stutter I've been experiencing since I've made this change on my SPA-2100.)

So I think with this one, Draco is definitely right: YMMV--your mileage may vary. I would have to guess there are two, potentially three main factors that determine whether you naturally experience a lot of network jitter with your phone calls: 1) Your ISP, 2) Your VOIP Provider and distance in ms to your sip proxy. Both 1 and 2 differ between Draco and me. Also thirdly, I am using a SPA-2100, while he is using a SPA-3000. So our test conditions do differ. But I encourage people to try it and see if they get Draco's problem of uncorrected jitter on their connection. I don't find I do on mine.
Forums » VOIP etc » Voice Over IP - VOIP » VOIP Tech Chat[General] PAP2 v2 and syslog »
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