whisk / WEBRTC-to-SIP

Setup for a WEBRTC client and Kamailio server to call SIP clients

WebRTC to SIP

Automated demo configuration to enable calls between WebRTC and legacy SIP clients. This setup will allow:

• WebRTC -> SIP
• SIP -> WebRTC
• WebRTC <-> WebRTC
• SIP <-> SIP

This configuration provide 3 virtual servers: SIP server with RTP proxy, TURN server and a web-server. Used components:

• Kamailio 4.4 (SIP and WebRTC server)
• RTPEngine (RTP proxy)
• coturn (TURN server)
• sip.js (WebRTC client)
• nginx (web-server)
• Ubuntu 16.04 LTS
• Vagrant (for sandboxing)

This configuration was tested with Chrome browser and CSipSimple for Android.

Installation

Hostnames and IPs

NOTE: By default all virtual servers are bound to 192.168.56.0/24 network and use websip.dev domain. If you'd like to use different addresses, review contents of /files/etc directory and apply changes as necessary.

Add to your /etc/hosts:

192.168.56.10 kamailio.websip.dev 192.168.56.11 turn.websip.dev 192.168.56.12 client.websip.dev

Certificates

As WebRTC requires TLS, you need to have TLS keys and certificates. You may go with pre-generated ones or create your own self-signed bundle:

cd certs
./generate.sh

NOTE: previously generated keys and certs will be overwritted.

You may optionally add ca.pfx (or ca.crt) to your system trusted CA list.

Provisioning

• Install Vagrant. You may also need to install VirtualBox.
• Launch boxes by typing:

  vagrant up kamailio
  vagrant up turn
  vagrant up client

It could take a few minutes and about 8GB of disk space.

If you'd like to provision boxes again, type:

  vagrant provision <box name>

See vagrant help for more info.

Usage

With a SIP client connect to our SIP server:

• Username: oldsip
• Password: oldsip
• Server: kamailio.websip.dev

Open https://kamailio.websip.dev to confirm security exception if you have not added self-generated CA certificate to trusted list.

Open https://client.websip.dev in Chrome browser. This page implements simple WebRTC client with built-in user credentials (websip:websip). Ented oldsip to call to.

You should receive a call from websip on your legacy SIP client.

Try to do it vice-versa etc.

More info

Please refer to the following files:

• Configuration files under files/etc
• WebRTC sample page under files/var/
• Boxes provision scripts in Vagrantfile

License

All used components are distributed under their respective licenses.