[SR-Users] Kamailio with Asterisk as Media Relay

[David Villasmil]

Also asterisk is a B2BUA, so if calls are delivered  to the B side by
asterisk, the B side will see call coming from the Asterisks, not from the
Kamailio.

On Mon, Oct 8, 2018, 15:54 Alex Balashov <abalashov at evaristesys.com> wrote:

> 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
>
> Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
>
> _______________________________________________
> Kamailio (SER) - Users Mailing List
> sr-users at lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20181008/887d1497/attachment.html>