node.js-sip

node.js-sip is a comprehensive VoIP framework for Node.js. Despite its name, this library goes beyond SIP (Session Initiation Protocol) and offers a full-fledged toolkit for building robust VoIP applications. Leverage its extensive capabilities for SIP transport, registration, messaging, call handling, and more.

Features

• SIP Transport: Built-in transport layer supporting UDP.
• Client and Server Modes: Easily switch between User Agent Client (UAC) and User Agent Server (UAS).
• Call Handling: Initiate, accept, reject, and terminate calls.
• Message Handling: Send and receive SIP messages.
• Customizable: Built with modularity and extensibility in mind.

Installation

npm install node.js-sip

Getting Started

Basic Usage

Here's a quick example to demonstrate how to set up a client and make a SIP call.

const VOIP = require('node.js-sip');

const voipClient = new VOIP({
    type: 'client',
    transport: {
        type: 'UDP',
        port: 5060,
    },
    username: '1000',
    register_password: 'yourPassword',
    register_ip: '192.168.1.2',
    register_port: 5060,
}, (response) => {
    if (response.type === 'REGISTERED') {
        console.log('Successfully registered with the SIP server.');
        v.call({
            to:'14444444444',
            ip:'192.168.1.12',
            port:'5060',
            callId:'1234567890',
            username:'1001',
            client_callback:(m) => {
                console.log(`client_callback`)
                console.log(m)
            }
        })
    }
});

Setting Up a SIP Server

//Nik Hendricks 10/13/23
const SIP = require('../SIP')
const VOIP = require('../')

const USERS = {
    '1000':{
        password:'rootPassword',
        name:'test testerson',
        ip:undefined,
        port:undefined,
        registered:false,
        call_id:undefined,  
        extension: '1000'
    },
    '1001':{
        password:'rootPassword',
        name:'Bill Billerson',
        ip:undefined,
        port:undefined,
        registered:false,
        call_id:undefined,  
        extension: '1001'
    }
}


var server = new VOIP({
    type:'server',
    transport:{
        type: 'UDP',
        port: 5060,
    },
},
(d) => {
    if(d.type == 'UAS_READY'){
    }else if(d.message !== undefined){
        let parsed_headers = SIP.Parser.ParseHeaders(d.message.headers);
        if(d.type == 'REGISTER'){
            server.uas_handle_registration(d.message, USERS, (response) => {
                console.log('response')
                console.log(response)
            })
        }else if(d.type == 'INVITE'){
            server.uas_handle_invite(d.message, USERS, (response) => {
                console.log('response')
                console.log(response)
            })
            
        }
    }
})

server.TrunkManager.addTrunk({
    name:'trunk1',
    type:'SIP',
    username:'1001',
    password:'rootPassword',
    ip:'192.168.1.2',
    port:5060,
    callId:'1234567890'
})

server.Router.addRoute({
    name: 'Main Trunk Route',
    type: 'trunk',
    match: '^[0-9]{11}$',
    endpoint: 'trunk:trunk1', //will decide if i want the prefix or to use the type property
})

setTimeout(() => {
    server.TrunkManager.trunks['trunk1'].register();
}, 1000)

API Reference

Constructor

new VOIP(props, callback)

• props: Configuration object for the VOIP instance.
  • type: 'client' or 'server'.
  • transport: Transport configuration.
    • type: 'UDP' (currently supported).
    • ip: Local IP address for the transport.
    • port: Port for the transport.
  • username: SIP username (for client).
  • register_password: SIP password (for client).
  • register_ip: Registrar IP (for client).
  • register_port: Registrar port (default: 5060).
• callback(response): Callback invoked on significant events.

Methods

call(props)

Initiates a SIP call.

• props: Call properties.
  • to: Destination phone number.
  • ip: Destination IP address.
  • port: Destination port.
  • callId: Call ID.
  • username: SIP username.
  • client_callback: Callback for client events.

accept(message)

Accepts an incoming SIP INVITE message.

reject(message)

Rejects an incoming SIP INVITE message.

bye(message)

Terminates an ongoing call.

message(extension, body)

Sends a SIP MESSAGE to the specified extension.

• extension: The SIP extension to message.
• body: The message body.

TODO / Known Issues

• Further orginization of the codebase is needed.
• Finish RTP implementation.
• Finish SDPParser implementation.
• When making a call from a UAS in a b2bua configuration bye messages recived from the UAS Trunk UAC are not being associated with the originating messages.
• UAS UAC Registrations does not actually check if the password is correct.

🔆 Activity

🌟 Star History

License

MIT License. See the LICENSE file for more details.

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For detailed usage and advanced configurations, refer to the full documentation (coming soon).