Janus WebRTC is an open-source, general-purpose WebRTC gateway that enables real-time audio, video, and data communication through a modular plugin architecture. Developers use Janus to build video conferencing, live streaming, and SIP integration applications without relying on proprietary platforms. For teams that prefer a fully managed alternative, VideoSDK provides production-ready WebRTC infrastructure with SDKs across ten or more platforms.
Introduction
Open-source WebRTC servers have seen a surge in adoption as developers move away from proprietary communication platforms and toward self-hosted, customizable infrastructure. Among the most prominent options, Janus WebRTC stands out as a mature, general-purpose gateway that has powered production deployments since 2014.
The appeal is straightforward. Janus gives you full control over media routing, signaling, and session management. You choose your plugins, your transport layers, and your scaling strategy. That flexibility comes with operational responsibility, though. Running Janus in production requires understanding its architecture, plugin system, and performance characteristics.
This guide walks through everything you need to evaluate and deploy Janus WebRTC in 2026. You will learn how the core gateway works, which plugins matter most, how to deploy and connect clients, and where Janus fits relative to alternatives like Mediasoup, Jitsi, and managed solutions like VideoSDK.
What Is Janus WebRTC Server?
Janus WebRTC is defined as an open-source, general-purpose WebRTC gateway designed to bridge WebRTC peers with backend media and signaling services. Created by Meetecho and maintained as a community project on GitHub, Janus acts as a middle layer between browsers or mobile clients and your application logic.
Janus works by accepting WebRTC connections from clients, negotiating media flows through its plugin system, and relaying audio, video, and data channels according to each plugin's logic. The server itself does not impose a specific use case. Instead, it provides a framework where modular plugins handle the actual media processing, whether that is video conferencing, live streaming, SIP bridging, or simple echo testing.
This plugin-based design is what separates Janus from monolithic WebRTC solutions. The core handles session management, ICE negotiation, and transport, while plugins define what happens to the media once it arrives. VideoSDK takes a different approach to the same problem by offering a fully managed video calling API that abstracts away the server infrastructure entirely, letting developers focus on application logic rather than gateway operations.
WebRTC Architecture and Core Components
Janus WebRTC is built around a small set of core components that work together to manage sessions, route media, and expose transport interfaces. Understanding these building blocks is essential before deploying or debugging a Janus instance.
The Janus core is the central engine. It handles WebRTC peer connections, DTLS handshakes, ICE candidate gathering, and media pipeline setup according to the W3C WebRTC specification. The core itself does not process media in any application-specific way. It creates the infrastructure for media to flow and then hands that media off to plugins.
Plugins are where the real work happens. Each plugin implements a specific feature set. The VideoRoom plugin manages multi-party video conferences. The AudioBridge plugin mixes audio from multiple participants. The Streaming plugin distributes live media to subscribers. Plugins receive media from the core, process it according to their logic, and send processed media back through the core to connected peers.
Sessions represent a client's connection to the Janus server. When a client connects, Janus creates a session with a unique identifier. Within that session, the client attaches one or more handles to specific plugins. A handle is the plugin-specific context for that client within that session. One session can have multiple handles attached to different plugins simultaneously.
Transport layers define how clients communicate with Janus for signaling. Janus supports multiple transports, including plain HTTP REST APIs, WebSockets, RabbitMQ, and Unix sockets. The transport layer carries JSON messages between clients and the server, while media flows over WebRTC data paths.
This separation of concerns is what makes Janus flexible. You can swap transports without touching plugin logic, or add new plugins without modifying the core. The trade-off is that every layer requires configuration and understanding, which adds operational complexity compared to a managed real-time communication platform.
Key Features and Janus Plugins
Janus WebRTC ships with a set of built-in plugins that cover the most common real-time communication use cases. Each plugin is independently maintained and can be enabled or disabled at compile time or runtime.
The EchoTest plugin is the simplest. It reflects a client's audio and video back to them, which makes it useful for testing WebRTC connectivity and debugging media pipeline issues. The VideoRoom plugin is the most widely used, implementing a multi-party video conferencing room where participants can publish and subscribe to each other's streams. It supports both mixer and forwarder modes, giving developers control over whether media is mixed server-side or forwarded individually.
The AudioBridge plugin provides server-side audio mixing for voice-only scenarios like conference calls and audio rooms. The Streaming plugin enables one-to-many media distribution, where a source feeds RTP packets to Janus and subscribers receive the stream over WebRTC. The SIP plugin bridges WebRTC clients with traditional SIP networks, enabling inbound and outbound calls to phone systems. The DataChannel plugin allows bidirectional data exchange between peers without media, useful for chat applications and real-time data synchronization.
Janus also supports custom plugins. Developers can write plugins in C that integrate with the Janus core through a well-defined plugin API. This extensibility is one of Janus's strongest features, though it requires C development expertise and careful memory management.
Setting Up Janus: Deployment Options
Deploying Janus WebRTC offers three main paths, each with different trade-offs in control, convenience, and maintenance overhead.
Source compilation gives you the most control. You clone the Janus repository from GitHub, install dependencies including libnice, libsrtp, and the Janus build toolchain, then compile and configure the server. This approach lets you enable or disable specific plugins and features at build time. It also requires you to manage dependencies, handle library version conflicts, and keep the build environment consistent across deployments. Source compilation is best for teams that need custom plugin development or want to run Janus on specialized hardware.
Docker containers simplify deployment significantly. The Janus community maintains official Docker images that package the server with all dependencies pre-configured. You pull the image, mount your configuration files, expose the required ports, and the container handles the rest. Docker deployment is ideal for development environments, CI testing, and production deployments where you want reproducible builds. The trade-off is less granular control over compile-time options, though you can build custom Docker images from source if needed.
Package managers offer the fastest path to a running instance. On Debian and Ubuntu systems, Janus is available through apt repositories. On macOS, Homebrew provides a formula. Package installation is convenient for quick evaluation but may lag behind the latest releases and offers limited control over which plugins are compiled in.
Regardless of deployment method, production setups require configuring STUN and TURN servers for NAT traversal, setting up HTTPS for secure WebSocket connections, and configuring firewall rules for the UDP port range that Janus uses for media. WebRTC security also demands proper certificate management and token-based authentication to prevent unauthorized access. These infrastructure concerns are significant, and they are one reason teams consider managed alternatives like VideoSDK, which handles TURN servers, geo-fencing, and network-adaptive streaming as part of the platform.
Connecting Clients: Janus JavaScript Library
The Janus JavaScript library, commonly known as janus.js, provides a client-side abstraction layer that simplifies connecting browsers to a Janus server. Rather than manually implementing WebRTC peer connection logic, signaling message formatting, and ICE negotiation, developers use the library to handle these tasks through a higher-level API.
The library manages the full connection lifecycle. It establishes a transport connection to the Janus server, creates a session, attaches handles to plugins, and handles the WebRTC offer and answer exchange. The library also includes the webrtc-adapter shim, which normalizes WebRTC API differences across browsers. This is important because Chrome, Firefox, and Safari still have subtle differences in how they implement WebRTC APIs, and the adapter prevents browser-specific bugs from reaching your application code.
For mobile applications, the janus.js library is not directly applicable. React Native and Flutter developers typically use platform-specific WebRTC libraries and implement the Janus signaling protocol manually or through community-maintained wrappers. This is an area where managed SDKs like VideoSDK's React Native SDK and Flutter SDK reduce complexity by providing native, platform-optimized WebRTC implementations without requiring developers to bridge signaling protocols themselves.
Using the Janus Streaming Plugin for Live Media
The Janus streaming plugin is one of the server's most powerful components for live media distribution. It enables one-to-many streaming where a single media source feeds multiple subscribers over WebRTC, achieving low latency that traditional HLS cannot match.
The plugin supports three streaming modes. On-demand mode creates a mountpoint that starts streaming only when at least one subscriber is connected and stops when the last subscriber leaves. Live mode keeps a mountpoint active regardless of subscriber count, which is useful for always-on streams like radio or surveillance feeds. RTP mode accepts incoming RTP packets from an external source, such as FFmpeg or GStreamer, and distributes them to WebRTC subscribers.
In a typical RTP streaming setup, an encoding tool like FFmpeg captures a media source, encodes it to a compatible codec, and sends RTP packets to a configured port on the Janus server. The streaming plugin receives these packets, wraps them in WebRTC, and distributes them to connected subscribers. Each subscriber receives the stream as a standard WebRTC media track in their browser.
This architecture achieves sub-second latency for live streaming, which is critical for interactive use cases like live shopping, auctions, and real-time events. The trade-off is scalability. Each subscriber requires a separate WebRTC connection, and the Janus server handles all media forwarding. For large-scale streaming with thousands of viewers, you need horizontal scaling or a CDN-based HLS fallback. Platforms like VideoSDK address this with interactive live streaming that supports sub-second latency for interactive audiences while offering RTMP output for broader CDN distribution.
Managing Sessions via the Janus REST API and WebSocket
Janus WebRTC exposes two primary signaling interfaces: a REST API over HTTP and a WebSocket API. Both carry the same JSON message format, and the choice between them depends on your application architecture and browser compatibility requirements.
The REST API follows a hierarchical endpoint structure. The root endpoint handles server-level requests like ping and info. The session endpoint, created by sending a create request to the root, manages a single client session. The handle endpoint, created by attaching a plugin within a session, manages the plugin-specific interaction. Clients send requests to these endpoints and receive synchronous responses. For asynchronous events like media state changes or participant joins, the REST API uses long-polling, where the client periodically polls the session endpoint for pending events.
The WebSocket API provides a more efficient alternative. A single WebSocket connection carries all messages for a session, eliminating the need for long-polling. Messages include a transaction ID that correlates requests with responses, and the server pushes asynchronous events over the same connection. This reduces latency and overhead compared to REST long-polling.
Both APIs require careful handling of transaction IDs, session timeouts, and error responses. Sessions expire if not kept alive with periodic ping messages, and handles must be detached cleanly to avoid resource leaks. These signaling details are handled automatically by managed platforms, which is one reason teams migrating from self-hosted Janus to VideoSDK's REST API report reduced operational overhead.
Janus Performance Tuning and Scalability
Janus WebRTC performance depends heavily on your workload profile and hardware configuration. Understanding the bottlenecks helps you size instances and plan for growth.
CPU is the primary constraint for media processing. In forwarder mode, where Janus simply relays packets between participants without decoding, CPU usage scales linearly with the number of concurrent media streams. In mixer mode, where the server decodes, mixes, and re-encodes audio or video, CPU usage is significantly higher per participant. For large conferences, forwarder mode is generally more scalable.
Network I/O is the second bottleneck. Each WebRTC connection consumes bandwidth proportional to the media bitrate, and Janus must handle both inbound and outbound traffic for every participant. A single Janus instance handling 100 participants in a video room with 1.5 Mbps per stream needs roughly 150 Mbps of outbound bandwidth, and that number grows quadratically in mesh-like forwarding scenarios.
Horizontal scaling requires running multiple Janus instances behind a load balancer. Because Janus is stateful, the load balancer must use sticky sessions or IP hashing to ensure a client's WebSocket or HTTP connection routes to the same instance. For cross-instance communication, Janus supports a shared session backend using Redis or RabbitMQ, though this adds complexity. TURN server capacity is also critical, as a significant percentage of WebRTC connections in production require TURN relay for NAT traversal.
Janus Troubleshooting: Common Pitfalls and Fixes
Even with a correct deployment, Janus WebRTC can produce issues that are difficult to diagnose without knowing where to look. Here are the most frequent problems and their practical fixes.
Delayed or frozen video in the streaming plugin often results from codec mismatches between the RTP source and the WebRTC subscriber. Ensure your encoding tool outputs a codec that Janus and the browser both support, typically H.264 or VP8 for video and Opus for audio. Check that the payload type in your RTP stream matches the mountpoint configuration.
Muted audio tracks usually stem from incorrect media direction in the SDP offer or answer. Verify that the plugin is configured to send media and that the client is subscribing with the correct media direction. Browser autoplay policies can also mute audio if the user has not interacted with the page before the stream starts.
MTU problems manifest as packet loss and video corruption. WebRTC uses SRTP over UDP, and large packets may be fragmented or dropped by network equipment. If you see consistent corruption on high-bitrate streams, reduce the MTU on the Janus server or configure the encoding tool to limit packet size.
Token and permission errors occur when the Janus token authentication mechanism is enabled but the client does not send a valid token with its requests. Check that your token generation logic matches the shared secret configured on the server, and ensure tokens are passed in the correct message field.
ICE failures, where connections never establish, typically indicate missing or misconfigured STUN and TURN servers. Verify that your STUN and TURN servers are reachable from both the Janus instance and the client network. Test TURN connectivity separately before debugging the full WebRTC flow.
Choosing Janus WebRTC Over Alternatives
Janus WebRTC is one of several open-source WebRTC servers available in 2026. Understanding how it compares to alternatives helps you make the right architectural decision.
Mediasoup is a Node.js-based WebRTC server that focuses on being a lightweight SFU. It does not include built-in plugins for specific use cases like SIP or streaming. Instead, it provides a media pipeline that developers build on top of using their own application logic. Mediasoup generally offers better performance for pure SFU workloads because it is more tightly optimized, but it lacks Janus's out-of-the-box plugin functionality. Teams that need fine-grained control over media routing often prefer Mediasoup, while teams that want pre-built functionality lean toward Janus.
Jitsi Videobridge is the media server behind Jitsi Meet. It is optimized for video conferencing and integrates tightly with the Jitsi client ecosystem. Jitsi is an excellent choice if your primary use case is video conferencing and you want a complete solution including client UI. However, Jitsi is less flexible than Janus for non-conferencing use cases like SIP bridging or custom streaming pipelines.
LiveKit is a newer open-source project that provides a complete WebRTC platform with client SDKs, server APIs, and a built-in SFU. LiveKit offers a more modern developer experience with SDKs for multiple platforms, but it has a smaller community and fewer deployment options than Janus. LiveKit is attractive for teams that want an open-source solution with managed-SDK convenience.
For teams that want to avoid self-hosting entirely, VideoSDK provides a managed WebRTC platform with SDKs for React, React Native, Flutter, Android, iOS, JavaScript, Python, Unity, and more. VideoSDK handles TURN servers, media routing, recording, transcription, and scaling, which eliminates the operational burden of running Janus in production. The Prebuilt UI Kit also offers zero-code embedding for teams that need a working video call interface without custom development.
Definitions Glossary
Janus Core: The central engine of the Janus WebRTC server that handles WebRTC peer connections, ICE negotiation, DTLS handshakes, and media pipeline setup before handing media off to plugins.
Plugin: A modular component in Janus that implements specific media processing logic, such as video conferencing (VideoRoom), audio mixing (AudioBridge), or live streaming (Streaming).
Session: A client connection to the Janus server, identified by a unique session ID, within which one or more plugin handles are created.
Handle: The plugin-specific context for a client within a session, representing the client's attachment to a particular plugin and managing that interaction.
Transport Layer: The signaling interface between clients and Janus, supporting HTTP REST, WebSockets, RabbitMQ, or Unix sockets for carrying JSON messages.
SFU (Selective Forwarding Unit): A media routing architecture where the server forwards individual media streams between participants without decoding or mixing, offering better scalability than MCU-based mixing.
Key Takeaways
- Janus WebRTC is a mature, open-source gateway with a modular plugin architecture that supports video conferencing, live streaming, audio bridging, SIP integration, and data channels.
- The core handles session management and WebRTC negotiation while plugins define media behavior, giving developers flexibility to customize functionality without modifying the server core.
- Deployment options range from source compilation for maximum control to Docker containers for convenience, but all production setups require STUN and TURN server configuration for NAT traversal.
- The streaming plugin enables sub-second latency live media distribution by accepting RTP from external sources like FFmpeg and delivering it to WebRTC subscribers.
- Horizontal scaling requires sticky sessions and shared state backends, making operational complexity a significant factor for large deployments.
- Teams seeking a managed alternative can use VideoSDK to avoid the infrastructure burden of self-hosted WebRTC servers while gaining SDKs across ten or more platforms.
Conclusion
Janus WebRTC remains a strong choice for developers who need a customizable, open-source WebRTC gateway with proven production reliability. Its plugin architecture gives you control over media processing that managed platforms cannot always match, and the community around Janus continues to maintain and extend the project.
That said, running Janus in production is not trivial. TURN server management, horizontal scaling, session state synchronization, and client SDK maintenance across platforms all require dedicated engineering effort. For teams that would rather focus on application features than WebRTC infrastructure, VideoSDK offers a managed platform with multi-platform SDKs, built-in recording and transcription, and a Prebuilt UI Kit for zero-code embedding. You can explore the REST API reference or sign up at app.videosdk.live/login to get started with free credits.
What are you building with WebRTC? Drop a comment below, I would love to hear whether you are self-hosting Janus or considering a managed platform for your real-time communication use case.
Step 5: Implement Participant View
The participant view is a crucial part of any WebRTC application, as it displays the video and audio streams from all participants in the session. In this section, we'll extend our HTML and JavaScript code to handle multiple participant streams, ensuring they are displayed correctly on the user interface.
Creating the Participant View
First, update your
index.html file to include a container for participant videos:HTML
1<!DOCTYPE html>
2<html>
3<head>
4 <title>Janus WebRTC Demo</title>
5 <style>
6 .participant {
7 display: inline-block;
8 margin: 10px;
9 }
10 .participant video {
11 width: 200px;
12 height: 150px;
13 }
14 </style>
15</head>
16<body>
17 <h1>Janus WebRTC Demo</h1>
18 <div id="joinScreen">
19 <input type="text" id="username" placeholder="Enter your name">
20 <button id="joinButton">Join</button>
21 </div>
22 <div id="videoScreen" style="display:none;">
23 <div id="localParticipant" class="participant">
24 <video id="localVideo" autoplay muted></video>
25 </div>
26 <div id="remoteParticipants"></div>
27 <button id="startButton">Start</button>
28 <button id="callButton">Call</button>
29 <button id="hangupButton">Hang Up</button>
30 <button id="muteButton">Mute</button>
31 <button id="videoButton">Video Off</button>
32 </div>
33 <script src="https://cdn.jsdelivr.net/npm/janus-gateway@0.11.5/janus.js"></script>
34 <script src="app.js"></script>
35</body>
36</html>Next, update
app.js to handle multiple participant streams:JavaScript
1let localStream;
2let remoteStream;
3let janusConnection;
4let janusPlugin;
5let myUsername;
6let myRoom = 1234; // Example room ID
7let remoteFeed = null;
8
9document.getElementById('joinButton').addEventListener('click', joinRoom);
10document.getElementById('startButton').addEventListener('click', start);
11document.getElementById('callButton').addEventListener('click', call);
12document.getElementById('hangupButton').addEventListener('click', hangUp);
13document.getElementById('muteButton').addEventListener('click', toggleMute);
14document.getElementById('videoButton').addEventListener('click', toggleVideo);
15
16function joinRoom() {
17 myUsername = document.getElementById('username').value;
18 if (myUsername === "") {
19 alert("Please enter a username");
20 return;
21 }
22 document.getElementById('joinScreen').style.display = 'none';
23 document.getElementById('videoScreen').style.display = 'block';
24 initializeJanus();
25}
26
27async function start() {
28 localStream = await navigator.mediaDevices.getUserMedia({ video: true, audio: true });
29 document.getElementById('localVideo').srcObject = localStream;
30 janusPlugin.createOffer({
31 media: { data: true },
32 success: function(jsep) {
33 const body = { request: "join", room: myRoom, ptype: "publisher", display: myUsername };
34 janusPlugin.send({ message: body, jsep: jsep });
35 },
36 error: function(error) {
37 console.error("WebRTC error:", error);
38 }
39 });
40}
41
42function initializeJanus() {
43 Janus.init({
44 debug: "all",
45 callback: function() {
46 janusConnection = new Janus({
47 server: 'ws://localhost:8188/',
48 success: function() {
49 janusConnection.attach({
50 plugin: "janus.plugin.videoroom",
51 success: function(pluginHandle) {
52 janusPlugin = pluginHandle;
53 janusPlugin.send({
54 message: { request: "create", room: myRoom }
55 });
56 },
57 error: function(error) {
58 console.error("Error attaching plugin", error);
59 },
60 onmessage: function(msg, jsep) {
61 let event = msg["videoroom"];
62 if (event) {
63 if (event === "joined") {
64 let publishers = msg["publishers"];
65 for (let i in publishers) {
66 let id = publishers[i]["id"];
67 let display = publishers[i]["display"];
68 newRemoteFeed(id, display);
69 }
70 } else if (event === "event") {
71 // Handle events
72 }
73 }
74 if (jsep) {
75 janusPlugin.handleRemoteJsep({ jsep: jsep });
76 }
77 },
78 onlocalstream: function(stream) {
79 // Do nothing
80 },
81 onremotestream: function(stream) {
82 let remoteVideo = document.createElement('video');
83 remoteVideo.autoplay = true;
84 remoteVideo.srcObject = stream;
85 document.getElementById('remoteParticipants').appendChild(remoteVideo);
86 }
87 });
88 },
89 error: function(error) {
90 console.error("Error connecting to Janus", error);
91 }
92 });
93 }
94 });
95}
96
97function newRemoteFeed(id, display) {
98 janusConnection.attach({
99 plugin: "janus.plugin.videoroom",
100 opaqueId: "remoteFeed" + id,
101 success: function(pluginHandle) {
102 remoteFeed = pluginHandle;
103 let body = { request: "join", room: myRoom, ptype: "subscriber", feed: id };
104 remoteFeed.send({ message: body });
105 },
106 error: function(error) {
107 console.error("Error attaching plugin", error);
108 },
109 onmessage: function(msg, jsep) {
110 if (jsep) {
111 remoteFeed.createAnswer({
112 jsep: jsep,
113 media: { audioSend: false, videoSend: false },
114 success: function(jsep) {
115 let body = { request: "start", room: myRoom };
116 remoteFeed.send({ message: body, jsep: jsep });
117 },
118 error: function(error) {
119 console.error("WebRTC error:", error);
120 }
121 });
122 }
123 },
124 onremotestream: function(stream) {
125 let remoteVideo = document.createElement('video');
126 remoteVideo.autoplay = true;
127 remoteVideo.srcObject = stream;
128 document.getElementById('remoteParticipants').appendChild(remoteVideo);
129 }
130 });
131}
132
133function call() {
134 // Logic to start a WebRTC call using Janus
135}
136
137function hangUp() {
138 // Logic to hang up a WebRTC call
139}
140
141function toggleMute() {
142 const audioTracks = localStream.getAudioTracks();
143 if (audioTracks.length > 0) {
144 audioTracks[0].enabled = !audioTracks[0].enabled;
145 document.getElementById('muteButton').textContent = audioTracks[0].enabled ? 'Mute' : 'Unmute';
146 }
147}
148
149function toggleVideo() {
150 const videoTracks = localStream.getVideoTracks();
151 if (videoTracks.length > 0) {
152 videoTracks[0].enabled = !videoTracks[0].enabled;
153 document.getElementById('videoButton').textContent = videoTracks[0].enabled ? 'Video Off' : 'Video On';
154 }
155}Handling Multiple Streams with Janus SFU
The
onremotestream callback in the initializeJanus function handles the display of remote streams. When a remote stream is received, a new video element is created, and the stream is assigned to it. This video element is then appended to the remoteParticipants div.The
newRemoteFeed function attaches a new instance of the janus.plugin.videoroom plugin for each new remote participant. It handles the signaling and stream negotiation required to receive remote streams.By implementing these changes, you can display video and audio streams from multiple participants in your WebRTC session, creating a dynamic and interactive user experience. In the final section, we'll discuss how to run and test your application to ensure everything works smoothly.
Step 6: Run Your Code Now
With the code and configuration in place, it's time to run your Janus WebRTC application and test its functionality. Follow these steps to start the Janus server and load your application in a web browser:
[a] Start the Janus Server
Open a terminal and navigate to the Janus installation directory. Start the Janus server by running:
bash
1 sudo janusEnsure the server starts without any errors.
[b] Open the Web Application
Open your web browser and navigate to the directory where your HTML file is located. Open the
index.html file. For example:bash
1 file:///path/to/your/janus-webrtc-app/index.html[c] Join and Test
Enter a username, click "Join," and then click "Start." Allow access to your camera and microphone. Test the "Call" and "Hang Up" buttons to ensure functionality.
Troubleshooting Common Issues
- WebSocket Connection Errors: Ensure the WebSocket server URL in your
app.jsmatches the Janus server's WebSocket configuration. - Media Access Issues: Check browser permissions for accessing the camera and microphone.
- Plugin Errors: Ensure the necessary plugins are enabled and correctly configured in the Janus server.
Conclusion
In this article, we building real-time chat application using Janus WebRTC Media Server with JavaScript from scratch. We've covered everything from installing Janus, configuring it, creating a user-friendly join screen, implementing essential controls, and displaying participant views. By following these steps, you now have a foundational understanding of how to leverage Janus WebRTC for real-time communication applications. Janus's flexibility and plugin-based architecture make it a powerful tool for developing scalable and customizable WebRTC solutions.
Free $20 Balance for AI Voice Agents & Video Calls
FAQ
