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WebRTC Network with SIP.js and Routr

This is a simple example of a WebRTC network using SIP.js and Routr. The network consists of a SIP proxy (Routr) and two WebRTC clients (John and Jane). The clients are able to make and receive calls from each other.

Prerequisites

Starting the network

To start the network, run the following command:

DOCKER_HOST_ADDRESS=192.168.1.7 docker compose up

Be sure to change the DOCKER_HOST_ADDRESS environment variable to the IP address of the host machine. This is necessary for both clients to be able to connect to the SIP proxy.

In addition to Routr, the previous command will start two WebRTC client at http://localhost:8080. You must open two different browsers to simulate two different clients.

Setting the clients

When ask for credentials, use the following:

For John Doe:

  • Username: 1001
  • Password: 1234
  • Domain: sip.local
  • Signal Server: ws://localhost:5062

For Jane Doe:

  • Username: 1002
  • Password: 1234
  • Domain: sip.local
  • Signal Server: ws://localhost:5062

Making a call

To make a call please use the Target AOR field. For example, to call Jane, enter sip:1002@sip.local and click the call button.

Getting help

If you have a bug report, please file an issue in this repository's Issue Tracker. For questions or feedback, please join Fonoster's community on Discord.