[SR-Users] Kamailio as front proxy for multiple sip servers

[Yuriy Gorlichenko]

If you looking for examples: you can use this one
https://github.com/havfo/WEBRTC-to-SIP as starting point

anyway, the Path mentioned by Alex is the best approach.

чт, 6 мая 2021 г. в 10:34, Wojtko, Daniel <daniel.wojtko at rittec.de>:

> ​Hi,
>
> afaik rtpproxy doesn't support WebRTC but rtpengine does
>
>
> Regards
>
>
> Daniel
> ------------------------------
> *Von:* sr-users <sr-users-bounces at lists.kamailio.org> im Auftrag von
> Eliphas Levy Theodoro <eliphas at gmail.com>
> *Gesendet:* Mittwoch, 5. Mai 2021 23:44
> *An:* sr-users at lists.kamailio.org
> *Betreff:* [SR-Users] Kamailio as front proxy for multiple sip servers
>
> Hello!
>
> I am trying to config one kamailio as reverse proxy for a bunch of
> internal (no internet address) separate asterisk sip instances (per
> domain). The kamailio server would be the only with the valid IP address,
> so would use rtpengine to force to be in the media path.
>
> Like this scenario:
> https://opensips.org/pipermail/users/2020-August/043610.html
>
> I have used as starting point this very basic config:
>
> https://blog.voipxswitch.com/2015/03/27/kamailio-basic-sip-proxy-all-requests-setup/
>
> Basically just added rtpproxy support, and voilà, inter-SIP is
> working, media always passing into the proxy.
>
> The problem: I would have WebRTC phones connecting too. I tried setting
> WSS up in kamailio, and asterisk (pjsip) wouldn't know how to send the
> message to the proxy: on register it has trasnport=wss in the contact:
> header, looks like it is confusing the asterisk.
>
> So, I resort for the wisdom of the list :) What would be the
> good-best-path to take here, hack the header, or put the webphones
> registering directly on the asterisks (with a nginx reverse proxy maybe)?
>
> Someone must have already made a blog with such setup, but I could not
> google-unearth it at least until now.
>
> Regards,
> --
> Eliphas
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