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

Matthew Jordan mjordan at digium.com
Mon Jun 29 07:32:18 CDT 2015


On Mon, Jun 29, 2015 at 4:36 AM, Kelvin Chua <kelchy at gmail.com> wrote:
> Guys,
>
> just tried asterisk13 and added seanbrights' patch for opus.
>
> incoming INVITE has fmtp ------>
> maxplaybackrate=8000;sprop-maxcapturerate=8000
> but INVITE to my registered peer is ---------->
> maxplaybackrate=48000;sprop-maxcapturerate=48000
>
> it should not even have to load up the opus patch because it is just a
> passthrough
> have you changed anything to chan_sip.c to make this work?
>

Do you have res_format_attr_opus loaded?

-- 
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org



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