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Jitter Question

Posted on 2011-09-06
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Last Modified: 2013-11-12
When I go VoipPhone->ISP->ISP->Asterisk Box->ISP->Voip Provider->PSTN, I get intermittent jitter, BUT...the asterisk is running MixMonitor and the recording is always clear as a bell. The jitter is coming at either PSTN/VoipPhone end, or both.
I guess the fact the recording is always clear means its the downlink, but could be either side. Is this just a fact of life? (The asterisk is behind a fibre broadband connection)
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Question by:Silas2
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Author Comment

by:Silas2
ID: 36488166
PS The asterisk box is very ropey, ie 500MB membery, Pentium II (I think), do you think beefing that up would make a difference?
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Expert Comment

by:tabturn
ID: 36488764
Are these managed circuits w/ QoS?
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Author Comment

by:Silas2
ID: 36489301
err....I'm afraid I don't know what that is, Could you elaborate?
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Expert Comment

by:nauliv
ID: 36520872
Hello Silas,

Have you tried to put a VoIP phone on the same network as AsteriskPBX ? This would take the ISP out of the equation and would tell us whether or not it is related to the VoIP phone or Asterisk itself...

Let us know!
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Author Comment

by:Silas2
ID: 36522181
Thanks for your suggestion, I've beefed up the server, it seems to be better so far. I'm on it all the time, I might try changing the Voip trunk provider. Unfortunately, its intermittant.
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Expert Comment

by:nauliv
ID: 36523199
The interesting part I that the mixmonitor recordings are clean. Are recording both the caller/callee audios, and they are both fine? If this is the case, it would suggest that the problem is with the asterisk>isp>ipphone path, which could be a bandwidth issue. Is there a lot of other (non voip) traffic going out the same path as asterisk->ipphone ? If the answer is 'sometimes', this would explain the intermittent results... And in this case, you could see if the routers on these isp connections allow to prioritize traffic and set priority on traffic from/to the asterisk ip address.

If the internet connections are dedicated to voip and nothing else, then i'd suggest to check which codec you are using, and if it can be replaced with a less bandwidth consuming one (gsm for example).

Let us know how it's going !
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Author Comment

by:Silas2
ID: 36524371
I'm using G729a, which I heard is the best?
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Expert Comment

by:nauliv
ID: 36525772
G729a is indeed a very low bandwidth consuming codec. I'm assuming you didn't purchase the asterisk license for it, so it is used in pass-through mode (straight from the PSTN to the IP-Phone), so the jitter-latency is in effect from end to end.

Could you try using the GSM or uLAW codecs, and ensure your sip.conf has a "canreinvite=no". This way, the audio connections will be from Ip-Phone <-> PBX and PBX <-> PSTN.

Let us know how it goes !
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Author Comment

by:Silas2
ID: 36530349
Ok thanks for that suggestion, do me a favour in that I'm currently trying lots of combinations, ADSL/physical locations/Laptops so I'll get back to you slowly over this.
I think beefing up the hardware running asterisk has made a difference + 1 laptop had a cheap external sound card which was causing a jitter-like effect. I've got loads more combinations though...
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nauliv earned 500 total points
ID: 36531426
Sure. Keep us updated :)
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