Hello,
if both endpoints can do webrtc srtp, then it works with rtpproxy to do
srtp packet forwarding for nat traversal or networks bridging.
Cheers,
Daniel
On 06.05.21 10:31, Wojtko, Daniel wrote:
Hi,
afaik rtpproxy doesn't support WebRTC but rtpengine does
Regards
Daniel
------------------------------------------------------------------------
*Von:* sr-users <sr-users-bounces(a)lists.kamailio.org> im Auftrag von
Eliphas Levy Theodoro <eliphas(a)gmail.com>
*Gesendet:* Mittwoch, 5. Mai 2021 23:44
*An:* sr-users(a)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
<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-reques…
<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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