WEBRTC to SIP client and server

How to setup Kamailio + RTPEngine + TURN server to enable calling between WEBRTC client and legacy SIP clients. This setup will bridge SRTP --> RTP and ICE --> nonICE to make a WEBRTC client (SIPJs) be able to call legacy SIP clients.

This setup is for Debian 9 Stretch for all servers.

This setup is configured to run with the following servers:

Server - Kamailio + RTPEngine + Nginx (WEBRTC client)
Server - TURN

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

Architecture

Get certificates

For the certificates you need a simple solution is Let's Encrypt certificates. They will work for both Kamailio TLS and Nginx TLS. On the servers you need certificates, 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/XXXXX-XXXXX/PUT-IP-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'
find . -type f -print0 | xargs -0 sed -i 's/XXX-XXX/PUT-DOMAIN-OF-YOUR-TURN-SERVER-HERE/g'

Install RTPEngine

This will do the SRTP-RTP bridging needed to make WEBRTC clients talk to legacy SIP server/clients.

apt-get install build-essential dpkg-dev debhelper iptables-dev libcurl4-openssl-dev libglib2.0-dev libhiredis-dev libpcre3-dev libssl-dev markdown zlib1g-dev libxmlrpc-core-c3-dev dkms linux-headers-`uname -r` default-libmysqlclient-dev libavcodec-dev libavfilter-dev libavformat-dev libavresample-dev libavutil-dev libevent-dev libjson-glib-dev libpcap-dev
git clone https://github.com/sipwise/rtpengine.git
cd rtpengine
./debian/flavors/no_ngcp
dpkg-buildpackage
cd ..
dpkg -i ngcp-rtpengine-daemon_*.deb ngcp-rtpengine-iptables_*.deb ngcp-rtpengine-kernel-dkms_*.deb
cd WEBRTC-to-SIP
cp etc/default/ngcp-rtpengine-daemon /etc/default/
/etc/init.d/ngcp-rtpengine-daemon restart

Install IPTables firewall

This is required by RTPEngine for setting up the IPTables chain, and 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
/etc/init.d/kamailio restart

Install WEBRTC client

apt-get install nginx
cd WEBRTC-to-SIP
cp etc/nginx/sites-available/default /etc/nginx/sites-available/
cp -r client/* /var/www/html/

Install TURN server

apt-get install coturn
cp etc/default/coturn /etc/default/
cp etc/turn* /etc/
/etc/init.d/coturn restart

Testing

You should now be able to go to https://webrtcnginxserver/ and call legacy SIP clients.

ihusejnovic/WEBRTC-to-SIP