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Linksys SPA942 Jitter Adjustments

Posted on 2010-11-11
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I am looking for some tips on adjusting jitter and any other audio quality control on Linksys SPA942 phones.

Mainly users experience a "garbled" or "warbling" effect randomly when talking.  Also sometimes an echo.

These phones are connected over the WAN to an asterisk box, and calls come in and out through wholesale SIP carriers.

I am using G729a for all calls.

Thanks!
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Question by:jkockler
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by:SkykingOH
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All of the jitter buffering in the world can't make up for a poor network environment.

You don't "adjust" jitter.  Jitter is a measurement of LAN impairment.

Is your WAN oversubscribed?  

g.729a is great at reducing bandwidth however if you are transcoding that could cause additional issues.  Are you using a soft switch between your phones and providers or are the phones directly registering to the providers?

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by:jkockler
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I have the phones connecting  to an asterisk box, over the WAN.. Then asterisk connects to the providers..

Linksys SPA942 --> Internet ---> asterisk ---> Internet --> Providers

There should be plenty of available bandwidth.

The latency between asterisk and the phones is generally between 65ms and 89ms..

So far in the Linksys phones, I have reduced the RTP packet size from .030 down to 0.020

I have also reduced the network jitter level settings on the phones from high to low.
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The jitter adjustment on the phone is the size of the jitter buffer.  Setting it to low will make things worse because you are telling the phone you have low jitter and to buffer less.

You must have some type of jitter and packet loss or the phones would sound fine.

Once the packet hits the Internet you have no control.

Have you watched the actual path with a tool such as MTR that gives you real time jitter and packet loss?

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by:jkockler
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Thanks

I have not tried MTR yet.

I am going to try a new provider because I do not have the issue when calling between phones connected to asterisk, in other words when I eliminate the provider leg of the calls, there are no audio problems.  However I have seen audio issues on 2 separate termination providers thus far, Gafachi, and Simwood..  Not so much on inbound calls though through Voip.MS
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by:jkockler
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Maybe if I push that network jitter level the other direction to very high, I might see an improvement.  I am just trying to avoid creating other issues such as a delay or an echo, which I have read increasing that buffer will cause.
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by:SkykingOH
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Increasing the buffer increases the packet size so any packet loss would be magnified.

That is why I suggested running MTR instead of guessing.  

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by:jkockler
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alright I will try MTR

I have run pings to all hosts involved, and had pretty much no packet loss, but it looks like MTR gives some more in depth options.

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by:jkockler
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wow this MTR is friggin sweet.

Looks like I am getting between 2-3% loss on one of the hops between me and my termination provider Simwood...
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by:jkockler
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Looks like I am also seeing between 2-4% loss on at least 3 hops between asterisk and the phones.
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by:jkockler
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sorry for being manic with my posts here..

As the amount of packets sent grows, the loss percentage between asterisk and the phones is dropping to less than 1% on most of the hops, however a loss exists in almost every hop.

The loss percentage between asterisk and the provider seems to be leveling out at about 1.1% on about 2/3 of the hops.
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by:jkockler
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Round trip times averages etc:

Simwood  average of 129.3, and worst so far on that route is 577, out of 1100 packets

gafachi average of 73.5, and worst at 512, out of 600 packets

flowroute average of 115.3, and worst at 582, out of 860 packets

between phones and asterisk, average of 64.4, and worst hop of 871
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by:jkockler
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screen shot of the different routes
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by:rivira
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Hi Jkockler,

I have experience in IP Telephony as I worked for a Wholesale Carrier in the US.  Wholesale carriers are where many of the International routes are sent through as they are more competitive in price and therefore increase profits for resellers or other ITSPs.  This offering is possible to achieve by mixing routes of different qualities (for example premium, standard and so forth)

If you are calling to a number outside the US it is possible that the "quality" of the route you hit when calling from your asterisk system varies from time to time and thus you will have a different experience from call to call.

So my recommendations are as follows:

1) Check that you have enough bandwidth on your WAN for the number of concurrent calls that you need to place at any given time.  
If using G.729 codec use N Calls x 32kbps (up/down bandwidth) to calculate required bandwidth. Make sure you have QoS properly implemented at your edge router so you can give priority to SIP and RTP traffic.

2) Check the over subscription of your WAN.  Perhaps you are experiencing network congestion from your Internet provider when this happens.

3) Check your termination provider.   Ask them if they have several quality plans and if so, on which one you are on.


One final question, are your IP phones all behind the same location or at different locations?


Example:

IP Phones/SPAs ------WAN---- internet ---- asterisk  ?

or.

IP Phones/SPAs ------WAN1--------internet-----
                                                                                ------------ asterisk ?
IP Phones/SPAs-------WAN2--------internet-----

If all of your phones are in the same place and only the asterisk server is on the Internet, most likely your RTP (Voice) traffic is going through your LAN which it would not impact voice quality.  If this is the case and you are not dialing internationally I would pay close attention to recommendations 1 and 2 before number 3.


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by:jkockler
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Thanks for the feedback!!

I ended up changing termination providers and have no issues since.  Not holding my breath though.. : )
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