[SR-Users] Kamailio with Asterisk as Media Relay

[Alex Balashov]

On Mon, Oct 08, 2018 at 02:35:54PM +0200, Daniel Tryba wrote:

> On Mon, Oct 08, 2018 at 07:16:43AM -0400, Alex Balashov wrote:
> > The SDP-bearing INVITE and response are simply passed along as-is by
> > Kamailio, and it is the SDP which specifies where the media goes. So, if
> > endpoint A calls through Kamailio proxy B to Asterisk server C via SIP,
> > A and C will negotiate media amongst themselves without any intervention
> > or special measures on your part whatsoever.
> 
> In theory, but with Asterisk in the middle be prepared to have this fail
> since it initially is in the loop regarding RTP and can negotiate
> incompatible RTP legs between AB and BC which will not be fixed when
> Asterisk leaves the RTP path. Mainly I experience this with
> dtmf/telephone-events mapping, e.g.: a=rtpmap:101 telephone-event/8000
> If a and c have different values, dtmf will fail.

Well, yes, all kinds of interesting things can happen in the bridging
process. But in principle, at least, it is possible to bridge RTP across
two call legs without such issues. :-)

-- 
Alex Balashov | Principal | Evariste Systems LLC

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