GrimmKull / Reticulum

WebRTC Webphone with SIP Proxy implemented on Raspberry Pi platform.

Reticulum

Reticulum (lat. dim. m.) - Little net (a network; any reticulated system or structure.)

WebRTC Webphone with SIP Proxy implemented on Raspberry Pi platform.

Status



Try it out at reticulum.outbox.systems or at reticulum.link.

Browser support

Currently Reticulum supports only the latest Chrome browser on desktop and mobile.

Please note that mobile version still has same issues with website responsive layout and might be difficult to use.

Firefox and Opera support coming soon.

Launching SIP Proxy and Webphone static file hosting

To start the Reticulum server use:

./start.sh

To start the Reticulum server as daemon use:

./run.sh

To stop the Reticulum server use:

./stop.sh

When running on Windows make sure that the ruby gems have been installed with openssl support.

For detailed deployment and setup instructions take a look at SETUP.md file.

Using the webphone

Navigate to: https://$IP_ADDRESS:7000/index.html
where $IP_ADDRESS is the address of the machine where Reticulum Proxy is hosted. Click on Connect to make a WebSocket connection to Reticulum Proxy and to get the Media from your microphone and webcam. Once the video from webcam shows up in the lower right corner click Register. Congratulations you can now make a call to another registered Reticulum user.

Admin page

Navigate to: https://$IP_ADDRESS:7000/admin.html
in order to access the Admin page. This page allows you to view SIP message flow, view all transactions and their states on Proxy and create new users.

Creating new users

Reticulum has several users created by default to be used for testing. Navigate to users tab on Admin page and click Register to add another user.

Default test users for Reticulum

username	password
ana	apass
bob	bpass

User DB

Create the user DB storage:

CREATE DATABASE reticulum;

CREATE TABLE user (
    id INT AUTO_INCREMENT,
    name VARCHAR(100) NOT NULL,
    email VARCHAR(100) NOT NULL,
    password VARCHAR(250) NOT NULL,
    PRIMARY KEY(id)
);

Building single JavaScript webphone file

To combine all Webphone source files use gulp.

Install gulp

Inside the builds folder run the following commands:

npm install gulp
npm install gulp-concat gulp-rename gulp-uglify

To start the default gulp task and concatenate all JavaScript files call:

gulp

This will create a file reticulum_phone.js in the builds/build folder. To use this file copy it to the client folder, comment all other source files in index.html and uncomment the following line:

<script type="text/javascript" src="reticulum_phone.js"></script>

To create a single uglified Javascript file with null media call:

gulp node

This will create a file reticulum_node_phone.min.js in the builds/build folder.

NOTE: Null media build can be used for testing the SIP stack since it will not reference the WebRTC and media components nor will it add SDP payload to SIP messages.

Using NGINX as reverse proxy

If you checkout the code for Reticulum to /var/www/ folder on your server you can use the following configuration to use NGINX as the proxy. NGINX will host your static files and proxy all WebSocket requests on https://your.comain.com/ws to Reticulum. To see an example of this configuration in action go to reticulum.outbox.systems or reticulum.link.

upstream reticulum {
    server 0.0.0.0:7000;
}

server {
    listen 443 ssl;

    # The host name to respond to
    server_name your.domain.com;

    ssl_certificate /home/user/path_to_certs/certificates/your.domain.crt;
    ssl_certificate_key /home/user/path_to_certs/certificates/your.domain.key;

    # Logs
    access_log /var/log/your.domain.log;

    # Path for static files
    root /var/www/reticulum/client;
    index index.html;

    #Specify a charset
    charset utf-8;

    # WS conf
    location /ws {
        proxy_pass https://reticulum/ws;
        proxy_http_version 1.1;

        add_header Access-Control-Allow-Origin *;

        proxy_set_header X-Real-IP $remote_addr;
        proxy_set_header Upgrade websocket;
        proxy_set_header Connection Upgrade;
        proxy_set_header X-Forwarded-For $proxy_add_x_forwarded_for;
        proxy_set_header Sec-WebSocket-Protocol $http_sec_websocket_protocol;

        proxy_read_timeout 86400;
        proxy_redirect off;
        break;
    }
}

Resources

• Faye WS
• P2P SIP
• SIP.js
• SIP
• libre
• baresip
• sipml5
• JsSIP

JsSIP support and interoperability

To test the JsSIP support go to tryit.jssip.net

WARNING: Handshake errors with WebSocket connection in JsSIP you may encounter are probably caused by the fact you are using a different, possibly unsigned, certificate with your Reticulum proxy then the one on tryit.jssip.net. This error usually appears on OSX.

Reticulum supports JsSIP as a client. You can use it to communicate between different instances of JsSIP client and with Reticulum webphone. Use the example bellow

as a configuration template.

sipml5 support and interoperability

To test the sipml5 support go to sipml5 call

WARNING: Handshake errors with WebSocket connection in sipml5 you may encounter are probably caused by the fact you are using a different, possibly unsigned, certificate with your Reticulum proxy then the one on sipml5 call. This error usually appears on OSX.

Reticulum supports sipml5 as a client. You can use it to communicate between different instances of sipml5 client and with Reticulum webphone. Use the examples bellow as a configuration template.

To configure sipml5 to use Websocket connection to an instance of Reticulum proxy click on the Expert mode? button and use the following example configuration:

STUN server issues

Trickle example

To test total time for ICE candidates look at the link above.

Usually getting the candidates takes about 9 seconds regardless of the configuration parameters on Windows machine.

On OSX machine getting the candidates is much faster.

Admin page diagrams example

There might be performance issues when using the Admin after many calls have been made. Rendering diagrams in such a case can be extremely resource consuming.

Please use the admin page only for registering new users and looking at the SIP Call flow diagram for a single call, and not for multiple calls on a busy Reticulum system.

Diagrams

Title: Call 1
A->>Proxy: Invite
Proxy->>A: 100 Trying
Proxy->>B: Invite

B->>Proxy: 180 Ringing
Proxy->>A: 180 Ringing

B->>Proxy: 200 OK
Proxy->>A: 200 OK

A-->>B: ACK

Running WS stress test

NOTE: This guide is deprecated. A new custom testing tool will be available soon.

npm install -g thor

Add support for SIP messages to Thor by adding the protocol: "sip" line to mjolnir.js:

var socket = new Socket(task.url, {
	protocolVersion: protocol,
	protocol: "sip"
});

Run Thor with the following command:

thor --amount 1000 --messages 1 --masked wss://$IP_ADDRESS:7000

Make sure to add a unique SIP Register message generator to generator.js