On Thu, Jun 8, 2017, at 11:35 AM, Stephen Davies wrote:
> Hi,
>
> Reviewing rtp_engine.c it appears that we only support telephone-event
> rtp
> with a sample rate of 8000?
>
> JSSIP using Opus offers Opus as "opus/48000/2" and then (by necessity, I
> think), telephone-event/48000.
>
>
> In
Hi,
Reviewing rtp_engine.c it appears that we only support telephone-event rtp
with a sample rate of 8000?
JSSIP using Opus offers Opus as "opus/48000/2" and then (by necessity, I
think), telephone-event/48000.
EG (this is a JSSIP using WebRTC behind a Freeswitch system):
v=0
