Changing Audio Codec appears to fix most issues...

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davecool
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Changing Audio Codec appears to fix most issues...

Post by davecool »

I've stumbled onto what appears to be a solution to many of the issues I have experienced.

I logged into the Grandstream HT701 ATA suppplied by Fongo and changed the audio codec from PCM to G729. All the stuttering is gone, as is the "bad cell phone" voice quality. To our ears, and the people we call, it sounds like a regular phone line again. I have even turned off the QoS in my router and calls sound normal even with torrents running.

This hasn't changed the Fongo-side issues we have experienced when calling to our family in the USA, but it certainly improves our experience! Maybe Fongo can recommend this change to others who have experienced the same issues we have had.

David
pedor
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Re: Changing Audio Codec appears to fix most issues...

Post by pedor »

Interesting that a lower quality codec would actually improve your call experience so it seems to imply that your network was having trouble processing the PCM (G711) requirement of 64Kbps while G729 only needs 8Kbps.

Have you tried a VoIP test like this one where you can compare the scores among the codecs?
http://myspeed.visualware.com/servers/n ... alware.com
davecool
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Re: Changing Audio Codec appears to fix most issues...

Post by davecool »

I have tried that test a number of times, it says my DSL should support PCM without trouble, but it just doesn't appear to be the case. G729 is working very well for us! Too bad it took 6 weeks of frustration to figure this out.
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Jake
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Re: Changing Audio Codec appears to fix most issues...

Post by Jake »

Could it be a setting in your router? Maybe it is giving only a very small bandwidth to your ATA, but full speed to your computers. That might explain when you do tests via your browser it seems like it should be OK, but because the ATA is not getting that speed in practice things go wrong.

G711 should be pretty good on most connections, as long as their isn't another bandwidth hog somewhere.
davecool
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Re: Changing Audio Codec appears to fix most issues...

Post by davecool »

I've tested this ATA connected directly to the modem with the router and rest of my network disconnected and PCM still wasn't perfect. I suppose it doesn't matter now that it appears to be working well.
beverly
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Re: Changing Audio Codec appears to fix most issues...

Post by beverly »

I for one has this problem but switched codec. I noticed I have been getting lots of lost packets each time it breaks up. I wonder if that server that caused lost packets before is causing problems again. O I might mention that sometimes the other caller can't hear me but I can hear them. I was think this a port issue but it is not. My set-up hasn't changed for a long time now until today when I changed the codec on the ATA but many reboot to try and fix the problem but no go.
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bridonca
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Re: Changing Audio Codec appears to fix most issues...

Post by bridonca »

Here is some clarifications. G.729 is not a low quality codec. It is a codec that compresses the audio signal, where it is slightly degraded, but not to the point where most people can tell the difference between G.729 and G.711.

There reason G.729 sounds better on VOIP unfriendly internet connections is because it is much more tolerant to poor internet connections than G.711. Not only does G.729 use a lot less bandwidth, it also delays the audio signal slightly to allow for jitter. But if there is too much jitter, no codec can give you a good VOIP experience.

Jitter is when packets are delivered out of order. This causes big issues for VOIP traffic, which has to be delivered close to real time. You can set up a buffer to wait up for the slow packets, but that introduces delay, or latency. Most people notice the latency when it is over 1/3rd of a second or 300 ms. When you also have to account for the time it takes a signal to get from phone to phone, 300ms is not much room to work with when it comes to jitter control.

I should note with Fongo, unless this has been fixed, incoming connection still use the G.711 codec. I would put more effort in consistently pestering the ISP that is providing such a duff internet connection. You are paying good real money for garbage internet, and the ISP should be held into account for a sub standard service. Nowadays, jitter usually comes from the ISP traffic shaping bit torrent packets, which in the process, messes up VOIP packets.
beverly
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Re: Changing Audio Codec appears to fix most issues...

Post by beverly »

OK I changed my mind it doesn't help. Please disregard my other post
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Funkytown
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Re: Changing Audio Codec appears to fix most issues...

Post by Funkytown »

beverly wrote:OK I changed my mind it doesn't help. Please disregard my other post
Are your calls good now?
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