[asterisk-dev] Adding new codec passthrough support

[Alex Massover]

Hello,

I'm trying to evaluate how complex is to add new codec to Asterisk. I read about media architecture in Asterisk 10 and took a look on how Kevin Fleming added support for G.722.1 (Siren) - https://reviewboard.asterisk.org/r/158/diff/
The codec we plan to add is iSac, it supports multiple sampling rate, like SILK.

The very basic support we need is passthrough and record (playback is nice to have, but not a must). Recording should be in native isac format. No need for translation, re-negotiation etc, as well we don't want to implement the codec itself.
Af far as I understood it's about defining isac format and format attribute interface - more or less taking look on support of SILK and Siren and doing the same.

Did I get it correctly, or it's much more complex?


--
Best Regards,
Alex Massover

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