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How to Integrate WebRTC and SIP with JavaScript?

Discover how to integrate WebRTC and SIP for seamless real-time communication. Learn about the basics, advanced features, and troubleshooting tips to enhance your web applications.

What is WebRTC SIP?

WebRTC (Web Real-Time Communication) and SIP (Session Initiation Protocol) are two pivotal technologies in the realm of online communication. WebRTC enables real-time audio, video, and data sharing directly between browsers without needing plugins, while SIP is a protocol used to initiate, maintain, and terminate real-time sessions that include voice, video, and messaging applications. The integration of WebRTC and SIP, known as WebRTC SIP, combines the best of both worlds, allowing seamless real-time communication over web interfaces with robust session management.

Importance and Applications of WebRTC SIP

WebRTC SIP is integral to various modern web applications. It is widely used in web conferencing solutions, VoIP (Voice over Internet Protocol) services, customer support systems, and collaborative tools. This integration leverages the ease of access provided by web applications and the comprehensive session control and signaling capabilities of SIP, resulting in secure, efficient, and high-quality communication experiences.

Core Concepts of WebRTC

WebRTC facilitates real-time communication directly in web browsers by establishing peer-to-peer connections for audio, video, and data streams. It eliminates the need for external plugins, ensuring a seamless user experience. The fundamental components of WebRTC include MediaStream (to capture audio and video), RTCPeerConnection (to handle network communication), and RTCDataChannel (for peer-to-peer data exchange).

WebRTC Architecture

WebRTC operates using several key components:
  • ICE (Interactive Connectivity Establishment): Ensures reliable connection establishment between peers.
  • STUN (Session Traversal Utilities for NAT): Helps discover the public IP address and port mappings of peers.
  • TURN (Traversal Using Relays around NAT): Provides a fallback mechanism if direct peer-to-peer connections fail.
  • Signaling: A separate mechanism required to exchange information needed to establish the connection. WebRTC does not define a signaling protocol, so developers can use any method, such as WebSockets or SIP, to handle this aspect.
The signaling process involves exchanging session control messages like SDP (Session Description Protocol), which describes multimedia communication sessions to establish a connection.

Core Concepts of SIP

SIP is a protocol used to initiate, manage, and terminate multimedia communication sessions. It operates in a client-server architecture, where clients, also known as User Agents, can initiate or terminate calls, while servers help route and manage these calls. SIP messages are categorized into requests (like INVITE, BYE, REGISTER) and responses (like 200 OK, 404 Not Found).

SIP Architecture

The SIP architecture comprises several components:
  • User Agents (UA): These are the endpoints in a SIP communication session, capable of initiating and receiving calls.
  • Proxy Servers: Act as intermediaries that route SIP requests to the recipient.
  • Registrar Servers: Handle SIP REGISTER requests, updating location information for User Agents.
A typical SIP call flow includes registering a user with a registrar, initiating a call with an INVITE request, establishing a session, and finally terminating the call with a BYE request.

Foundational Code Examples and Basic Integration

Setting Up WebRTC


Before diving into WebRTC, ensure you have a basic understanding of HTML, JavaScript, and web server setup. You'll need:
  • A text editor (e.g., VS Code)
  • A web server to serve your HTML files (e.g., local server using Node.js or Python's SimpleHTTPServer)
  • A modern web browser (e.g., Chrome, Firefox)

Basic WebRTC Example

Let's create a simple example where we establish a peer-to-peer connection and stream video between two peers.


1<!DOCTYPE html>
2<html lang="en">
4    <meta charset="UTF-8">
5    <title>WebRTC Example</title>
8    <h1>WebRTC Video Stream</h1>
9    <video id="localVideo" autoplay playsinline></video>
10    <video id="remoteVideo" autoplay playsinline></video>
11    <script>
12        const localVideo = document.getElementById('localVideo');
13        const remoteVideo = document.getElementById('remoteVideo');
15        // Get user media
16        navigator.mediaDevices.getUserMedia({ video: true, audio: true })
17            .then(stream => {
18                localVideo.srcObject = stream;
19                // Create peer connection
20                const peerConnection = new RTCPeerConnection();
21                stream.getTracks().forEach(track => peerConnection.addTrack(track, stream));
23                // Handle ICE candidates
24                peerConnection.onicecandidate = event => {
25                    if (event.candidate) {
26                        // Send candidate to the remote peer
27                    }
28                };
30                // Handle remote stream
31                peerConnection.ontrack = event => {
32                    remoteVideo.srcObject = event.streams[0];
33                };
35                // Create offer
36                peerConnection.createOffer()
37                    .then(offer => peerConnection.setLocalDescription(offer))
38                    .then(() => {
39                        // Send the offer to the remote peer
40                    });
42                // Handle answer from remote peer
43                // peerConnection.setRemoteDescription(answer);
44            })
45            .catch(error => console.error('Error accessing media devices.', error));
46    </script>

Explanation of the Code

  1. We access the user's media devices (camera and microphone) using navigator.mediaDevices.getUserMedia.
  2. The local video stream is displayed in the localVideo element.
  3. We create an RTCPeerConnection instance and add the local tracks to it.
  4. ICE candidates are handled to establish a connection.
  5. When the peer connection receives a remote stream, it is displayed in the remoteVideo element.
  6. An SDP offer is created and set as the local description, which would be sent to the remote peer for establishing the connection.

Setting Up SIP


To set up a basic SIP environment, you need:
  • A text editor (e.g., VS Code)
  • Node.js for running server-side code
  • JsSIP library for handling SIP in JavaScript

Basic SIP Example

Let's create a simple SIP user agent using the JsSIP library.


1<!DOCTYPE html>
2<html lang="en">
4    <meta charset="UTF-8">
5    <title>SIP Example</title>
6    <script src="https://cdn.jsdelivr.net/npm/jssip/dist/jssip.min.js"></script>
9    <h1>SIP User Agent</h1>
10    <script>
11        // Configuration
12        const socket = new JsSIP.WebSocketInterface('wss://sip.example.com');
13        const configuration = {
14            sockets: [socket],
15            uri: 'sip:username@example.com',
16            password: 'password'
17        };
19        // Create User Agent
20        const userAgent = new JsSIP.UA(configuration);
22        // Start the User Agent
23        userAgent.start();
25        // Register event listeners
26        userAgent.on('registered', () => {
27            console.log('Registered successfully.');
28        });
30        userAgent.on('registrationFailed', (e) => {
31            console.error('Registration failed:', e.cause);
32        });
34        userAgent.on('newRTCSession', (e) => {
35            const session = e.session;
37            session.on('confirmed', () => {
38                console.log('Call confirmed.');
39            });
41            session.on('ended', () => {
42                console.log('Call ended.');
43            });
45            session.on('failed', (e) => {
46                console.error('Call failed:', e.cause);
47            });
48        });
49    </script>

Explanation of the Code

  1. We include the JsSIP library for handling SIP functionality.
  2. A WebSocket interface is created to connect to the SIP server.
  3. Configuration settings, including the SIP URI and password, are defined.
  4. A JsSIP.UA (User Agent) instance is created and started.
  5. Event listeners are registered to handle successful registration, registration failures, and new SIP sessions.
  6. Within a new SIP session, further event listeners manage call confirmation, call ending, and call failures.

Integrating WebRTC with SIP

Signaling between WebRTC and SIP

Signaling is crucial for establishing WebRTC connections, and SIP can serve as the signaling protocol. This integration allows WebRTC to leverage SIP's robust session management capabilities.

Code Example: WebRTC SIP Integration

Let's integrate WebRTC with SIP using JsSIP for signaling.


1<!DOCTYPE html>
2<html lang="en">
4    <meta charset="UTF-8">
5    <title>WebRTC SIP Integration</title>
6    <script src="https://cdn.jsdelivr.net/npm/jssip/dist/jssip.min.js"></script>
9    <h1>WebRTC SIP Integration</h1>
10    <video id="localVideo" autoplay playsinline></video>
11    <video id="remoteVideo" autoplay playsinline></video>
12    <script>
13        const localVideo = document.getElementById('localVideo');
14        const remoteVideo = document.getElementById('remoteVideo');
16        // Get user media
17        navigator.mediaDevices.getUserMedia({ video: true, audio: true })
18            .then(stream => {
19                localVideo.srcObject = stream;
20                const peerConnection = new RTCPeerConnection();
22                stream.getTracks().forEach(track => peerConnection.addTrack(track, stream));
24                peerConnection.onicecandidate = event => {
25                    if (event.candidate) {
26                        // Send
28 candidate to the remote peer via SIP
29                    }
30                };
32                peerConnection.ontrack = event => {
33                    remoteVideo.srcObject = event.streams[0];
34                };
36                // Configuration for JsSIP
37                const socket = new JsSIP.WebSocketInterface('wss://sip.example.com');
38                const configuration = {
39                    sockets: [socket],
40                    uri: 'sip:username@example.com',
41                    password: 'password'
42                };
44                const userAgent = new JsSIP.UA(configuration);
45                userAgent.start();
47                userAgent.on('newRTCSession', (e) => {
48                    const session = e.session;
50                    session.on('peerconnection', (data) => {
51                        const rtcSession = data.peerconnection;
52                        rtcSession.addTrack(stream.getTracks()[0], stream);
53                    });
55                    session.on('sdp', (data) => {
56                        const sdp = data.sdp;
57                        peerConnection.setRemoteDescription(new RTCSessionDescription(sdp));
58                    });
60                    peerConnection.createOffer()
61                        .then(offer => peerConnection.setLocalDescription(offer))
62                        .then(() => {
63                            session.send('sdp', peerConnection.localDescription.sdp);
64                        });
65                });
66            })
67            .catch(error => console.error('Error accessing media devices.', error));
68    </script>

Explanation of the Code

  1. We set up media devices and display the local stream in the localVideo element.
  2. A WebRTC RTCPeerConnection is created, and local tracks are added.
  3. ICE candidates are managed to facilitate peer connection.
  4. JsSIP is configured and a user agent is created to handle SIP signaling.
  5. Event listeners handle new SIP sessions and peer connections, integrating SDP (Session Description Protocol) exchange to establish the WebRTC connection.
By following these steps, you can successfully integrate WebRTC with SIP, enabling seamless real-time communication in web applications.

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Advanced Applications of WebRTC SIP

DTLS and SRTP for Security

One of the critical aspects of WebRTC SIP integration is ensuring secure communication. WebRTC employs DTLS (Datagram Transport Layer Security) to encrypt data channels and SRTP (Secure Real-time Transport Protocol) to encrypt media streams. This dual-layer encryption ensures that both signaling and media data are protected from eavesdropping and tampering.

Implementation of DTLS and SRTP

Implementing DTLS and SRTP is seamless in WebRTC, as these security protocols are enabled by default. When you create a peer connection in WebRTC, it automatically uses DTLS for securing data channels and SRTP for securing audio and video streams. Below is an example of establishing a secure connection:


1const configuration = {
2  iceServers: [{ urls: 'stun:stun.l.google.com:19302' }],
3  mandatory: {
4    'DtlsSrtpKeyAgreement': true
5  }
8const peerConnection = new RTCPeerConnection(configuration);
10// Add tracks, handle ICE candidates, and create offers/answers as usual

NAT Traversal Techniques

NAT (Network Address Translation) traversal is a crucial aspect of establishing peer-to-peer connections. WebRTC uses ICE (Interactive Connectivity Establishment) to determine the best path to connect peers, often requiring the use of STUN (Session Traversal Utilities for NAT) and TURN (Traversal Using Relays around NAT) servers. STUN helps discover the public IP addresses of peers, while TURN relays the data when direct connections fail.

Practical Examples


1const iceServers = {
2  iceServers: [
3    { urls: 'stun:stun.l.google.com:19302' },
4    {
5      urls: 'turn:turn.example.com',
6      credential: 'password',
7      username: 'user'
8    }
9  ]
12const peerConnection = new RTCPeerConnection(iceServers);
This configuration helps ensure that connections can be established even in challenging network environments.

Troubleshooting Common Issues

Common WebRTC SIP Integration Problems

  1. Connection Failures: Often caused by misconfigured STUN/TURN servers or firewall restrictions.
  2. Audio/Video Issues: These can stem from codec mismatches, network bandwidth limitations, or device compatibility problems.

Debugging Tips

  1. Check ICE Candidate Exchange: Ensure that ICE candidates are being exchanged correctly between peers. Use browser developer tools to inspect the ICE candidate gathering and connection state.
  2. Verify Media Stream Setup: Ensure that media streams are correctly set up and added to the peer connection. Check if the local and remote streams are being received as expected.
  3. Network Configuration: Ensure that your network configuration allows WebRTC traffic. This may involve configuring firewalls and routers to permit WebRTC's UDP traffic.

Practical Advice and Solutions

  1. Use Logging: Enable detailed logging in your WebRTC and SIP implementations to track down issues.
  2. Test in Different Environments: Test your WebRTC SIP integration in various network conditions to identify potential issues.
  3. Update Regularly: Keep your WebRTC and SIP libraries up-to-date to benefit from the latest fixes and improvements.
By addressing these common issues and following best practices, you can ensure a robust and reliable WebRTC SIP integration in your web applications.


In conclusion, integrating WebRTC with SIP harnesses the strengths of both technologies to create a robust framework for real-time communication. This combination allows seamless audio, video, and data exchanges directly within web browsers, leveraging SIP’s comprehensive session management. By understanding the core concepts, implementing essential components, and addressing common challenges, developers can effectively deploy WebRTC SIP integrations in various applications, from web conferencing to VoIP services, ensuring secure, high-quality communication experiences for users.

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