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https://github.com/havfo/WEBRTC-to-SIP

Setup for a WEBRTC client and Kamailio server to call SIP clients
https://github.com/havfo/WEBRTC-to-SIP

bridge kamailio rtpengine sip turn webrtc webrtc-client

Last synced: 3 months ago
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Setup for a WEBRTC client and Kamailio server to call SIP clients

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# 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](https://github.com/havfo/SipCaller).



This setup is for Debian 12 Bookworm.



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](https://raw.githubusercontent.com/havfo/WEBRTC-to-SIP/master/images/webrtc-sip.png "WebRTC to 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):

```bash

apt-get install certbot

certbot certonly --standalone -d YOUR-DOMAIN

```

You will then find the certificates under:

```bash

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

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

```



## Get configuration files

```bash

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](https://github.com/sipwise/rtpengine#on-a-debian-system).



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

```bash

echo 'deb https://deb.sipwise.com/spce/mr11.5.1/ bookworm main' > /etc/apt/sources.list.d/sipwise.list

echo 'deb-src https://deb.sipwise.com/spce/mr11.5.1/ bookworm main' >> /etc/apt/sources.list.d/sipwise.list

wget -q -O - https://deb.sipwise.com/spce/keyring/sipwise-keyring-bootstrap.gpg | apt-key add -

apt-get update

apt-get install -y ngcp-keyring ngcp-rtpengine

```



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

```bash

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.

```bash

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

```bash

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.



```bash

kamctl add websip websip

service kamailio restart

```



## Install WebRTC client

This will install the client that can be found [here](https://github.com/havfo/SipCaller).



Install Nginx:

```sh

apt-get update

apt-get install nginx libnginx-mod-stream

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

```sh

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`.