I would like the Asterisk SIP channel to be able to handle 60ms of voice per packet instead of the default 20ms. It would be very nice to have this as a config option or at least to be able to hardcode it in the source files. Most SIP gateways allow you to do this and it would be great if Asterisk could do it as well. I would of course be willing to pay for this development and if anybody else thinks this is useful and wants to contribute $$$ to make this happen then please let me know.
Any developer willing to do this can contact me offlist and let me know what it would cost.
Thanks,
-- Andres
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