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

WEBRTC to SIP client and server

How to setup Kamailio + RTPEngine + TURN server to enable calling between WebRTC client and legacy SIP clients. This config is IPv6 enabled by default. This setup will bridge SRTP --> RTP and ICE --> nonICE to make a WebRTC client (sip.js) be able to call legacy SIP clients. The WebRTC client can be found here.

This setup is for Debian 9 Stretch.

This setup is configured to run with the following services:

  • Kamailio + RTPEngine + Nginx (proxy + WebRTC client) + coturn

The configuration is setup to always bridge via RTPEngine. To change the behavior, take a look in the NATMANAGE route.

Architecture

WebRTC - SIP architecture

Get certificates

For the certificates you need, a simple solution is Let's Encrypt certificates. They will work for both Kamailio TLS, Nginx TLS and TURN TLS. Run the following (you must stop services running on port 443 during certificate request/renewal):

apt-get install certbot
certbot certonly --standalone -d YOUR-DOMAIN

You will then find the certificates under:

/etc/letsencrypt/live/YOUR-DOMAIN/privkey.pem
/etc/letsencrypt/live/YOUR-DOMAIN/fullchain.pem

Get configuration files

All files needed to setup all components on Debian 9 Stretch.

git clone https://github.com/havfo/WEBRTC-to-SIP.git
cd WEBRTC-to-SIP
find . -type f -print0 | xargs -0 sed -i 's/XXXXXX-XXXXXX/PUT-IPV6-OF-YOUR-SIP-SERVER-HERE/g'
find . -type f -print0 | xargs -0 sed -i 's/XXXXX-XXXXX/PUT-IPV4-OF-YOUR-SIP-SERVER-HERE/g'
find . -type f -print0 | xargs -0 sed -i 's/XXXX-XXXX/PUT-DOMAIN-OF-YOUR-SIP-SERVER-HERE/g'

Install RTPEngine

This will do the SRTP-RTP bridging needed to make WebRTC clients talk to legacy SIP server/clients. You can find the latest build instructions in their readme.

The easiest way of installing is to get it from Sipwise repository:

echo 'deb http://deb.sipwise.com/spce/mr7.1.1/ stretch main' > /etc/apt/sources.list.d/sipwise.list
echo 'deb-src http://deb.sipwise.com/spce/mr7.1.1/ stretch main' >> /etc/apt/sources.list.d/sipwise.list
apt-get update
apt-get install -y --allow-unauthenticated ngcp-keyring
apt-get update
apt-get install -y ngcp-rtpengine

After you have successfully installed RTPEngine, copy the configuration from this repository.

cd WEBRTC-to-SIP
cp etc/default/ngcp-rtpengine-daemon /etc/default/
cp etc/rtpengine/rtpengine.conf /etc/rtpengine/
/etc/init.d/ngcp-rtpengine-daemon restart

Install IPTables firewall (optional)

RTPEngine handles the chain for itself, but make sure to not block the RTP-ports it is using. Take a look in iptables.sh for details, and apply it by doing the following. This will persist after reboot. You can run the iptables.sh script at any time after it is set up.

cd WEBRTC-to-SIP
chmod +x iptables.sh
cp etc/network/if-up.d/iptables /etc/network/if-up.d/
chmod +x /etc/network/if-up.d/iptables
touch /etc/iptables/firewall.conf
touch /etc/iptables/firewall6.conf
./iptables.sh

Install Kamailio

apt-get install kamailio kamailio-websocket-modules kamailio-mysql-modules kamailio-tls-modules kamailio-presence-modules mysql-server
cd WEBRTC-to-SIP
cp etc/kamailio/* /etc/kamailio/
kamdbctl create

Select yes (Y) to all options.

kamctl add websip websip
service kamailio restart

Install WebRTC client

This will install the client that can be found here.

To be able to support running HTTP(S) and TURN on the same port (443), we need a newer version of nginx that supports streams the way we need. Get it from official repo:

echo 'deb http://nginx.org/packages/mainline/debian/ stretch nginx' > /etc/apt/sources.list.d/nginx.list
curl -fsSL https://nginx.org/keys/nginx_signing.key | apt-key add -
apt-get update
apt-get install nginx
cd WEBRTC-to-SIP
cp etc/nginx/nginx.conf /etc/nginx/
cp etc/nginx/conf.d/default.conf /etc/nginx/conf.d/
cp -r client/* /var/www/html/
service nginx restart

Install TURN server

apt-get install coturn
cp etc/default/coturn /etc/default/
cp etc/turnserver.conf /etc/
service coturn restart

Testing

You should now be able to go to https://XXXX-XXXX/ and call legacy SIP clients. Click the account icon in the top right corner and add the following settings:

  • Display name: Whatever
  • SIP URI: websip@XXXX-XXXX
  • Password: websip
  • Outbound Proxy: wss://XXXX-XXXX/ws

To manually configure other TURN servers, change the config in client/config.js.

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