webrtc support and interop with asterisk

Hello,

First of all thanks for this project and all the hard work you are putting into maintaining this project.

I have started using baresip recently to connect to our asterisk server. We are now using it over TLS protocol, but would like to connect over webrtc. With TLS or SIP I need to allow port 5060 or 5061 in firewall, which gets scanned and attacked quite quickly. I can use a higher port, but its just a matter of time before it is found out.

I can avoid this with Webrtc/Websocket connection of asterisk with an nginx in front.

I have configured baresip with proxy as sip:myserver.io:443;transport=wss, and in nginx logs I am seeing: "GET / HTTP/1.1" 404 240 "-" "-" and baresip throws Protocol Error

What does baresip expect when it sends the HTTP request? Any one with a successful connection between asterik and baresip using WSS or webrtc?

Thanks!

[juha-h]

I configured my Apache web server to act as Websocket proxy like this:

<VirtualHost _default_:443>
...
         SSLProxyEngine on
	<LocationMatch />
		ProxyPass ws://192.168.206.95:5063/
	</LocationMatch>	
        SSLEngine on
...
</VirtualHost>

My SIP server is configure to listen for SIP requests over WS at at 192.168.206.95:5063.

Then I configured baresip’s outbound proxy like this:

sip:<web server address>:443;transport=wss

after which baresip was able to register with my SIP server via the Apache Websocket proxy.

[xuoguoto]

There is an update:

When I originally compiled libbaresip-android package pkg-config was missing. After installing pkg-config and recompiling, the crackling noise has disappeared. Now everything is working fine.

Thank you for helping me with this!