8 Oct
2018
8 Oct
'18
5:50 p.m.
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:
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/
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.