That example is not about SIP over WebRTC. Well another way which you can try is to perform an HTTP 301 redirect to the address:port you want to reach, however I still think putting Kamailio instead of nginx will let you do a lot more flexible stuff with SIP not changing the client part. ________________________________ Alexandru Covalschi VoIP Engineer and System Administrator tel: +373 673 98 493
3 янв. 2017 г., в 23:51, Sergey Safarov <s.safarov@gmail.com> написал(а):
ср, 4 янв. 2017 г. в 0:03, Alexandru Covalschi <568691@gmail.com <mailto:568691@gmail.com>>:
But FreeSwitch wants SIP headers "X-AUTH-IP" and "X-AUTH-PORT" in every SIP message not only connection establishing First of all Freeswitch doesn't require those headers. This header required to auth particular SIP/WebRTC clients by IP addresses via proxy daemons.
Well for that you should use STUN and set that in your clients application. Another way is to use Kamailio, yes, it can add those on the fly, but again - you must put it instead of the proxy server (which you may actually want to do as it is a SIP proxy by-desing and it works with websockets pretty fine). If i place Kamailio instead of http proxy server, then i must have two entry points: 1) for http trafic - http server; 2) web-socket trafic - Kamailio.
But i wants avoid configuration with more than one entry point.
Sergey
________________________________ Alexandru Covalschi VoIP Engineer and System Administrator tel: +373 673 98 493 <tel:+373%20673%2098%20493>
3 янв. 2017 г., в 22:47, Sergey Safarov <s.safarov@gmail.com <mailto:s.safarov@gmail.com>> написал(а):
Then you will try cofigure Kamailio between NGINX and FreeSwitch. In this case Kamailio can parce http headers and add requred SIP header. Are you can suggest other way to publish FreeSwitch socket on same port with http server?
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