Conversation without transcoding


#1

Hello,

We have FreePBX behind NAT and number 8-800 over sip trunk from provider.
All internal users have sip-phones.
All external users have 3G/4G mobile phones.
Is it possible in this case to provide end-to-end conversation without transcoding?
As far as I know, mobile phones use AMR-WB codec https://en.wikipedia.org/wiki/Adaptive_Multi-Rate_Wideband? Why FreePBX do not support this codec?


(David55) #2

On the first line of the Wikipedia page you reference, it says the algorithm is patented. That is why Asterisk doesn’t support it; it is not legally possible to obtain an GPL licensed and free of charge implementation. The second reason is probably because it is not widely used on SIP phones, so there is no incentive to provide even pass through support for it.

Also, I believe that a lot of mobile phones and mobile [phone networks, still use GSM based codecs, so any statement about the use of G.722.2 needs to be geographically qualified.

To use it you would need a direct mobile network gateway, as the PSTN is mainly G.711, with some G.722, which is not the same sort of technology as G.722.2, so there would be transcoding at any PSTN gateway.

A pass through implementation would require all audio files to be pre-converted to G.711, and wouldn’t allow features like whisper, conferences, answering machine detection, and some or all types of call recording, etc. I’m not sure if FreePBX uses PlayTones, but that wouldn’t work. People have used Asterisk with pass through for G.729, to get round the past patenting, and the lack of properly licensed free implementations, but that was for bandwidth minimisation, not quality.


(David55) #3

I’m also not sure how the multi-rate aspect would be handled by the networks. To me, that suggests that it is only used on individual air interfaces, where the rate can be negotiated, and the network probably transcodes it internally. I can’t see a network that is bandwidth starved honouring the high rate stream that is produced by a SIP phone seeking maximum quality.

Even if rate changes are negotiated end to end, Asterisk could no longer treat the stream as pass through, as it would need to understand the rate change signalling.


(system) closed #4

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