[asterisk-dev] storing INVITE fmtp and use it to send relay

[Kelvin Chua]

oh that figures. i am working on ast12 and never bothered trying 13. i'm
playing around with opus and so frustrated that it does not adapt to the
incoming sdp. even if after parse_sdp, it just resets everything on
generate_sdp. i'll try 13 later, thanks for the tip
On Jun 26, 2015 5:30 PM, "Alexander Traud" <pabstraud at compuserve.com> wrote:

> > If I receive an INVITE with fmtp from a peer, it won't be used to build
> the
> > INVITE to the egress right?
>
> With Asterisk 13/chan_sip, it is possible to copy over the fmtp - even 1:1
> -
> I do this here with AMR-WB. I created a res/res_format_attr_ and adopted
> format_parse_sdp_fmtp/format_generate_sdp_fmtp, just like the Opus sample.
> Thanks to Asterisk 13, the selected codec of the ingress gets the first
> priority for the egress.
>
> > Is there any function [to] act like a proxy?
>
> Mhm. I was not able to do that because here, Asterisk removed unknown
> codecs
> and adds its allowed ones after the one selected for the ingress. So the
> lines m=, a/v/t=, and their order are going to be different. But as stated,
> copying over (one!) fmtp per format is possible. By the way, which
> codec/format are you about?
>
>
>
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