WebSIP

µURU has an experimental WebSIP feature, this makes it possible to call phones directly from a browser.

It works by creating temporary extensions ad-hoc when someone wants to call an extension from the browser, the range for those extensions must be reserved so that no other extensions can be registers in this range.

Once the extension was created signaling to the asterisk server is done via websockets and media is transported via webrtc (using SIP.js). At the end of the call the extension gets deleted again so that the range will hopefully never be exhausted (of course depending on the size of the range).

To prevent problems with clients that do not unregister, µURU checks regulary for orphaned/unused extensions. To keep extensions alive a client must send at least every 3 minutes a keep alive request. All that is handled in the frontend.

Please note

All modern browsers require webrtc to be transported securely, so you need to use a reverse proxy with according certificates to enable secure transport for at least the asterisk server.

Configuration

Key	Description
UURU_ENABLE_WEBSIP	Enable websip (default is 1)
UURU_WEBSIP_PUBLIC	If true (default) yo don't need an account to use websip
UURU_WEBSIP_WS_HOST	External url where the asterisk websocket is reachable
UURU_WEBSIP_EXTENSION_RANGE	Tuple of start and end of the range e.G. [9900, 9999]