On Thu, Jun 29, 2017 at 11:55:51AM -0500, Richard Mudgett wrote:
> > To me this looks like a bug in asterisk. Either asterisk should use the
> > same rtp payloads for telephone-events on both call legs during inital
> > callsetup or asterisk should come to the conclusion there is an
> >
On Thu, Jun 29, 2017 at 8:32 AM, Daniel Tryba wrote:
> While trying to use direct_media I'm seeing RTP payload mismatches after
> succesful reinvites.
>
> Initial INVITE from endpoint A to asterisk has rfc4733 DMTF
> m=audio 35648 RTP/AVP 9 8 111 96
> a=rtpmap:96
While trying to use direct_media I'm seeing RTP payload mismatches after
succesful reinvites.
Initial INVITE from endpoint A to asterisk has rfc4733 DMTF
m=audio 35648 RTP/AVP 9 8 111 96
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
>From asterisk to upstream U:
m=audio 14338 RTP/AVP 9 8 111