Best sounding codec for FreePBX/Asterisk


I am currently run FreePBX12/Asterisk13. We are using free-pbx as a “telephone-board” for a non-profit, all volunteer internet radio station. I use a confbridge and in-studio softphone to bridge any phone callers tot he live studio sound board. This works pretty good, but because of the double encoding/decoding using basic G711u codec (one from gvoice motif to PBX and one from PBX to the softphone) I think the sound quality from callers and to the callers has plenty room for improvement.

I have tried G722 for my softphone and I can not hear any difference from G711u. I have three questions

I can not control Gvoice… it is what it is, but if I use a higher fidelity codec from PBX to my softphone, will that make any difference, or will that sound the same as if I just use G711u as I am using now?

If there is audible improvement using higher fidelity codec, how can I add that to the FreePBX/Asterisk (it appears that there is no Asterisk source is included)?

Which high-fidelity codec is available to me? I know that Opus is not a kosher deal as of now (correct me if I am wrong), perhaps there is something else (Speex, Silk, something I do not know about)?


Basically the best you can use is the lowest common factor between all your endpoints.

The “best” codec that works over most internet connections is g711, very few VSP’s support g722, if you are lucky to have one then prefer it, it is “better” . BUT you are limited to what the far-end connection can offer and very few of them do better than g711, and only on SIP connected device, it just won’t work over the PSTN.

If you are in the professional broadcast industry, then we often use bonded ISDN channels to get 44Kbs from the “talent” but that is for outbound FM transmission and use PSTN “hybrids” like Zenith to mix the various audio streams back into the broadcast stream, you just can’t polish a turd from the PSTN system (and google voice is even lower quality) unless you are Japanese (dorodango)

I mean I agree with dicko about lossless connetions but none of the typical codecs deal with jitter or packet loss from individual phones, we’re currently struggling to get even a receive jitter buffer working with PJSIP.

Can you enlighten me on the double encoding reference? G711 ulaw…there is no transcoding…its the native codec. it should sound exactly the same what the provider sends to you and what your softphone receives.
LOL…3 years later.


I do realize this but we’re trying to achieve google voice like quality over data / wifi. We network link conditioner for Mac OS and see.

G711 cannot achieve this in the slightest… So we tried iLBC and the results were better but then there is no receive buffer for pjsip channels which all our extensions are in since it’s the newer way to go.

Tried a chan sip with jitter buffer which works, but then there is some strange bug in asterisk where ilbc doesn’t work with chan SIP.

FEC on opus doesn’t seem to work either, so I mean all these wonderful technologies to have a great call even with packet loss and none of it works today :frowning:

but 711 is basically the quality that all mobiles and landlines use. How do you expect to be better when the source is already limited?

Almost all cellphones actually use gsm


Verizon for one can do g722 end to end over their own network if both are set to use their “hi-def calling”

but there will intrinsically be a time domain lag as the signal traverses the network, and that is almost always greater than Landlines, but jitter is not related to lag , it tries to reorder and compensate for errant packets, as such it will always increase lag and decrease “mos”


Yes GSM (may bad) …which is arguably an even worse sounding codec.

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