Virtual miniSIPServer accepts WebRTC calls

Virtual miniSIPServer accepts WebRTC calls

The new version of miniSIPServer allows users to make calls from browsers that support WebRTC, such as Chrome, Firefox and Edge. The slight inconvenience is that users must deploy their own TLS/SSL certificates to enable miniSIPServer to launch the WebSocket secure service and receive calls from browsers.

We have recently ported this feature to the cloud-based virtual miniSIPServer. More conveniently, users do not need to worry about certificates, as we automatically deploy TLS/SSL certificates for the virtual miniSIPServer. Users can simply initiate calls directly from their browsers. For example, a URL similar to the following can be used in a browser to place a call to your own cloud-based miniSIPServer:

https://www.myvoipapp.com/miniwebphone2/lite.html?server=15000.s2.minisipserver.com&clr=100&pwd=100&cld=101

The server parameter specifies the address of the virtual miniSIPServer. Currently, only virtual servers with the suffix s2.minisipserver.com are supported. If your server uses the old suffix s1.minisipserver.com, you must migrate to a new server to enable WebRTC calling.

For the other parameters: clr, pwd and cld, please refer to the previous documentation. The parameters are identical for both the local miniSIPServer and the cloud‑hosted virtual miniSIPServer.

Likewise, the virtual miniSIPServer currently only accepts incoming calls from WebRTC, and does not support initiating calls from the SIP domain to browsers or WebRTC clients.

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