Choosing to go open source over a CPaaS for your WebRTC media stack is a strategic decision about control, flexibility, and long-term ownership. For teams building real-time products, an open source WebRTC media server can offer the freedom to customize media handling, integrate deeply with your architecture, and avoid the constraints of a managed platform’s pricing or roadmap.
“With great power comes with great responsibility,” which is why open source is rarely the easiest path. But for products with demanding requirements like telehealth, contact centers, voice AI, or environments that need to be self-hosted, open source can be the most flexible and future-proof option. It offers deep customization, tight integration with telephony, AI, and broadcasting systems, and a deployment model your team fully owns, even if that means taking on more complexity in operations and maintenance.
If you’ve decided to go open source, the next step is choosing the right server for your use case. In this post, we’ll review why most WebRTC applications need a media server and then break down the leading open source WebRTC media server options and where each one fits best for various use cases.
For teams still weighing open source against a managed CPaaS like Vonage or Twilio or even building native WebRTC, our post Native, Open Source, or CPaaS? covers that decision in depth and remains a useful starting point.
Why most WebRTC applications need a media server
WebRTC is famous for enabling seamless, peer-to-peer (P2P) communication directly between web browsers. If you are building a simple 1-to-1 video chat, that P2P magic is incredibly powerful and efficient. You just connect the two users, step back, and let their browsers do the talking.
However, what happens when you need to scale that call to 50 users? What if you need to record the session, or connect the call to a traditional telephone network? Standard P2P for a 50-person call requires every single user’s device to upload 49 individual video streams and download 49 others. Devices will overheat, batteries will drain, and likely the call will crash even before finishing delivering every video frame. For complex real-time use cases and advanced features, you need a dedicated WebRTC media server.
A media server acts as the “middleman” in the cloud. Instead of users sending streams to each other, everyone sends their stream to the server. The server then processes, routes, or mixes those streams before sending them back out. They handle the heavy lifting, primarily acting as an SFU (Selective Forwarding Unit—routing streams efficiently) or an MCU (Multipoint Control Unit—mixing streams into a single feed). (See: Architecting WebRTC and SIP Integrations with MCU and SFU)
As real-time AI applications have matured, media servers have also become the foundation for voice AI pipelines, connecting live audio streams directly to speech recognition engines and large language models (LLMs) with the low latency those integrations demand.
The WebRTC open source media server landscape
There is no shortage of open source WebRTC media servers, but five have proven themselves in production environments and are worth serious evaluation.
| Server | Language | Architecture | License | Best for |
| LiveKit | Go | SFU | Apache 2.0 | AI agents, telehealth, general use |
| mediasoup | Node.js/Rust/C++ | SFU (library) | ISC | Gaming, spatial audio, custom routing |
| Janus | C | SFU + plugins | GPLv3 | SIP/telephony, broadcasting |
| Ant Media (Community) | Java/C++ | SFU/MCU | Apache 2.0 | Broadcasting, HLS/RTMP |
| Jitsi (JVB) | Java | SFU | Apache 2.0 | Video conferencing out of the box |
LiveKit
Built on Go, LiveKit provides an incredible developer experience with robust SDKs and a complete end-to-end platform out of the box, letting teams ship real-time features exceptionally fast. It is heavily optimized for modern workloads, including a dedicated Agents framework for seamlessly bridging WebRTC with AI pipelines and Large Language Models.
However, that convenience comes with a trade-off: LiveKit is highly opinionated about how your architecture must be structured. If your project requires deeply customizing the internal media routing or completely rewriting the signaling layer, you may find its rigid framework too restrictive.
mediasoup
mediasoup shines by acting as a low-level routing library inside your backend, offering unmatched flexibility and granular control over every single media track. Because you own the entire pipeline, it is perfect for complex projects like gaming, metaverse spaces, or spatial audio where media logic must sync perfectly with a custom application state.
The major limitation, however, is its steep learning curve. Because it does not provide any default architecture, out-of-the-box servers, or signaling protocols, you have to build all the room management, networking, and scaling logic entirely from scratch.
Janus WebRTC Server
Written in C, Janus is the battle-tested industry standard when it comes to interoperability, effectively bridging modern browser communication with legacy infrastructure. Its highly modular architecture lets you enable only the specific features your application needs via plugins, with its SIP plugin being the absolute best in class for enterprise telephony integration.
The downside is its older C-based foundation and complex operational tooling. This can make it harder for modern web developers to maintain, debug, and scale compared to newer Go or Node.js ecosystems.
Ant Media Server (Community Edition)
Written in Java and C++, Ant Media Server is a powerhouse for massive interactive broadcasting, capable of instantly transcoding low-latency WebRTC feeds into traditional streaming protocols like HLS or RTMP for passive viewers. This enables sub-second latency at a massive scale, making it the undisputed choice for live auctions or real-time sports where a multi-second delay means a lost bid.
Keep in mind, however, that the open-source Community Edition is intentionally limited. Scaling features like multi-node clustering and adaptive bitrate streaming (ABR) are locked behind their paid Enterprise license.
Jitsi Videobridge (JVB)
Written in Java, JVB is unparalleled if you want to deploy a secure, full-featured “Zoom clone” quickly, as the ecosystem provides a robust frontend UI straight out of the box. Its mature backend is highly reliable and handles standard multi-party video conferencing features flawlessly.
The main limitation is its monolithic nature; it is heavily optimized for standard meeting rooms. Decoupling the core Videobridge from the rest of the Jitsi stack to build highly custom, non-standard video flows can be a frustrating engineering effort.
Recommending the right server for your use case
There is no single “best” open source media server: only the best one for your specific requirements. Here is how they stack up in the real world:
Telehealth & EdTech Platforms
- Requirements: High reliability, simple room models, HIPAA compliance capabilities, and session recording.
- Recommendation: LiveKit or Jitsi. LiveKit’s robust SDKs and built-in recording make it perfect for building custom health and education platforms quickly. Alternatively, Jitsi is an excellent choice if you need to rapidly deploy a complete, self-hosted video room with minimal frontend development. And as your platform grows, Jitsi’s underlying architecture can be expanded, like in this telehealth scaling project where WebRTC.ventures took a self-hosted Jitsi setup and engineered its AWS infrastructure to support thousands of concurrent sessions.
Contact Centers & Remote Interpreting
- Requirements: Heavy integration with existing SIP, VoIP, and legacy PBX telephone networks.
- Recommendation: Janus WebRTC Server. Its SIP plugin is battle-tested. If you need to bridge the gap between web browsers and traditional telephony, like this WebRTC to SIP Gateway for ASL Interpreters built by WebRTC.ventures, Janus is the industry standard.
Live Auctions & Interactive Broadcasting
- Requirements: Ultra-low latency delivery to thousands of passive viewers simultaneously, and protocol transcoding.
- Recommendation: Janus, Mediasoup or Ant Media Server. Janus and Mediasoup are more flexible and lightweight. Ant Media is built specifically for broadcasting.
AI Voice Agents
- Requirements: The ability to easily connect real-time media streams to LLMs and Speech-to-Text engines with minimal delay.
- Recommendation: LiveKit. With its recent LiveKit Agents framework, it has become the premier media server for orchestrating AI pipelines, allowing developers to easily bridge WebRTC with services like OpenAI and Deepgram. It provides the exact low-latency infrastructure needed to build intelligent, conversational systems, like the AI Voice Agents That Collaborate and Contribute engineered by WebRTC.ventures.
Gaming, Spatial Audio, & Virtual Worlds
- Requirements: Extreme low latency, spatial audio positioning (volume based on distance), custom data channels, and deep integration with game state.
- Recommendation: mediasoup. Because it acts as a low-level library inside your backend, you get fine-grained control to route media tracks dynamically based on a user’s exact coordinates in a virtual world. For highly custom, game-state-dependent routing, this is the way.
Which open source WebRTC media server is right for you?
Choosing an open source WebRTC media server is ultimately an architectural decision as much as a technical one. The server you pick shapes not just how your media is routed today, but how your platform scales, what integrations become possible, and how much flexibility your team retains as your product evolves.
LiveKit and Jitsi offer mature ecosystems that reduce the distance between starting and shipping. mediasoup and Janus trade that convenience for fine-grained control that complex or custom use cases genuinely require. Ant Media solves a specific broadcasting problem better than anything else in the open source landscape.
None of them is the right one in isolation, but depends on the problem you’re solving and the context where such a solution runs.
If you are evaluating your WebRTC architecture for building a real-time application from the ground up, navigating a migration from a CPaaS to an open source WebRTC media server, or having trouble scaling your current media server, the WebRTC.ventures team has helped teams across telehealth, contact centers, broadcasting, and voice AI navigate these architecture decisions. Contact WebRTC.ventures for a media server architecture assessment.
Further Reading:
- How Much Does It Really Cost to Build and Run a WebRTC Application?
- Building an Open Source Voice AI Agent That Avoids Vendor Lock-In
- Reduce WebRTC Infrastructure Costs with a Hybrid P2P Architecture
- WebRTC Tech Stack Guide: Architecture for Scalable Real-Time Applications
- How to Set Up Self-Hosted STUN/TURN Servers for WebRTC Applications
- Architecting WebRTC and SIP Integrations with MCU and SFU
