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 8 Jessie for all servers.

For the clients to avoid firewalls and to have the best setup, divide the servers like this:

Server - Kamailio + RTPEngine
Server - TURN
Server - WEBRTC client

The default configuration is setup to always bridge via RtpEngine, if you want to bridge on failure, comment out the line #!define WITH_ALWAYS_BRIDGE in etc/kamailio/kamailio.cfg. If you bridge on failure, the proxy receives a 488 Not Acceptable Here from the other side, it will remove SRTP/ICE/RTCP-mux and try again.

Get certificates

For the certificates you need I recommend 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):

git clone https://github.com/letsencrypt/letsencrypt
cd letsencrypt
./letsencrypt 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 8 Jessie.

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 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` 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_*' 'ngcp-rtpengine-iptables_*' 'ngcp-rtpengine-kernel-dkms_*'
cd WEBRTC-to-SIP
cp etc/default/ngcp-rtpengine-daemon /etc/default/
/etc/init.d/ngcp-rtpengine-daemon restart

Install IPTables firewall (optional)

This is optional. If this is installed it will persist after reboot. You can run the iptables.sh 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 to legacy SIP clients.

Insha-A/WEBRTC-to-SIP