Welcome to the SIPSorcery demonstration for the signalrtc open source real-time communications VoIP and WebRTC signaling server.


The purpose of this demonstration is to allow SIP and WebRTC experimentation and testing. It should not be used for production purposes.

If you have a GitHub account click Login to create your free SIP Account. Note that you don't need a SIP account unless you'd like to test SIP registrations or authenticated SIP calls . The destinations in the SIP dial plan are all publicly accessible.

SIP Tests

The server supports the following functions for SIP clients:

The call destinations supported are as below. The dial plan is stored in a database and is a compiled on demand using the Roslyn .NET compiler.

The entries in the dialplan below can be called using the destination on each line and a host of sipsorcery.cloud. Examples:

If you have a publicly accessible SIP end point that you'd like to add please create an issue with the URI details.

WebRTC Tests

This server also supports some WebRTC demos and echo tests for a number of different WebRTC implementations.

Browser WebRTC + HTTP REST Signaling source
Browser WebRTC + HTTP REST Signaling source
Browser WebRTC + SIPSorcery WebRTC (echo server source) + HTTP POST request for Signaling
Browser WebRTC + Janus WebRTC (echo server reference) + HTTP POST request for Signaling
Browser WebRTC + Python aiortc WebRTC (echo server source) + HTTP POST request for Signaling

3rd Party Integrations

The signalrtc application is, as the name implies, used for signaling only. To create and consume audio and video streams additional applications or servers are needed. The primary purpose of 3rd party integrations is to check interoperability of the SDP negotiation and the media transport.

For VoIP the goal is to provide 4 interoperability tests per server:

For WebRTC the goal is to provide an echo test per server.

Hosting additional servers can be done but adds to the hosting costs. If you would like to see a particular server integrated for experimentation or testing feel free to open an issue and consider sponsoring the project to help with the costs.

Currently the servers below are hosted and are accessible for the tests listed above via the dial plan or WebRTC REST signaling controller:

Additional Features

This demonstration application is hosted on Azure and is automatically deployed using Azure DevOps. Additional features can be added by submitting pull requests to one of the open source projects below:

Additional servers that may be integrated depending on time, resources and interest: