Hello
I would like to ask if there is any available informations on how to use Kamailio as a SIP Proxy but letting (multiple) Asterisk Instances do the Media Relaying. Is this scenario supported and if yes are there any tutorials or documentation available for it?
I know the Kamailio Dispatcher Module, but this does as far as I know a dispatching of also the SIP Traffic. Not just the Media Traffic.
Hi Benjamin,
No positive action on your part is required of you or Kamailio to make this scenario work; it's simply the default.
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.
-- Alex
On Mon, Oct 08, 2018 at 01:12:56PM +0200, Benjamin Marty wrote:
I would like to ask if there is any available informations on how to use Kamailio as a SIP Proxy but letting (multiple) Asterisk Instances do the Media Relaying. Is this scenario supported and if yes are there any tutorials or documentation available for it?
I know the Kamailio Dispatcher Module, but this does as far as I know a dispatching of also the SIP Traffic. Not just the Media Traffic.
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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.
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. :-)
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@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@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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David Villasmil