[asterisk-dev] Asterisk 13: Media improvements update

Olle E. Johansson oej at edvina.net
Mon Feb 24 07:13:13 CST 2014

On 24 Feb 2014, at 13:59, Matthew Jordan <mjordan at digium.com> wrote:

> (2) I think it's worth thinking about how such a module would be
> written to be flexible enough for everyone's business needs. What
> kinds of things should it watch for, and with what tolerances? Does it
> defer any part of its processing to an external source? For example,
> instead of actually making the decision itself to switch to a lower
> bandwidth codec, it could raise an AMI event that says "hey, this
> channel looks like its dropping a lot of packets". It could register a
> new AMI action ("ChangeTheCodecs") that would allow an external source
> to initiate the change.

Yes, there needs to be a configurable way for this. You might want to fail over to a different SIP trunk
instead of switching to another codec. I would prefer having one or two schemes inside

For Opus, you may want to keep the codec, but change properties.


More information about the asterisk-dev mailing list