WebRTC Simulcast Explained: Improve Video Quality Without Bandwidth Burn

With video calls, webinars, and virtual events reaching record levels, transmitting high-quality video without wasting bandwidth is increasingly the problem. Fortunately, WebRTC Simulcast is an effective solution that makes real-time video more efficient and scalable than ever before. In this blog, let’s cover what simulcast is, how it works, and why it’s a must-have for any contemporary WebRTC solution.
What Is WebRTC Simulcast?
Simulcast in WebRTC is a method where the video sender sends several instances of the same video stream at varying resolutions and bitrates. For instance, the same video is sent at 360p, 720p, and 1080p. This enables the receiver — or a media server such as an SFU (Selective Forwarding Unit) — to select the best stream for each viewer depending on their device and network.
Imagine it as providing YouTube-like video flexibility to live communication.
⚙️ How does WebRTC Simulcast work?
Multiple versions of the video stream (low, medium, high quality) are produced by the client-side encoder.
All of these are transmitted to the SFU server.
The SFU smartly directs only the correct stream to each viewer.
Viewers with good internet and screens receive HD video; others receive a lower-resolution version that won’t lag or buffer.
Why WebRTC Simulcast Is Essential in 2025
These are important reasons why simulcast is no longer desirable, but indispensable for today’s VoIP and video applications:
✅ 1. Improved Quality for All
Various users, devices, and networks require various video qualities. Simulcast ensures that the appropriate quality is matched with each user without sacrificing performance.
✅ 2. Bandwidth-Efficient
Rather than simulcast sending high-quality video to all (even if they don’t require it), simulcast prevents bandwidth waste while providing a seamless experience.
✅ 3. No Server-Side Transcoding
Server-side transcoding is costly and uses pricey server resources. Simulcast prevents server-side video conversion, thereby lowering latency and expense.
✅ 4. Perfect for Multi-Participant Calls
Not everyone requires the same stream in a group call or webinar. Simulcast enables scalable broadcasting without burdening client or server systems.
✅ 5. Enhances WebRTC App Performance
Simulcast-using apps report quicker connections, smoother playback, and reduced dropouts, particularly on mobile and low-end devices.
Where Simulcast Truly Excels
Telemedicine: Provide doctors with HD feeds while patients with lagging connections still view the video.
EdTech: Enable students on any device and location.
Remote Work & Meetings: Streamline team calls on laptops, mobiles, and desktops.
Live Events: Scale broadcasts without sacrificing performance.
Native Compatibility
Modern browsers like Chrome, Firefox, and Safari now fully support simulcast.
WebRTC media servers (e.g., Janus, Jitsi, mediasoup) handle simulcast distribution effortlessly.
👨💻 Want to Integrate Simulcast Into Your App?
Adding simulcast isn’t plug-and-play — it requires:
Expertise in WebRTC codecs and signaling
Proper configuration of SFUs and video encoding
Cross-browser and mobile testing
👉 That’s where our experts can help.
Hire proven WebRTC Developers to implement simulcast, SFUs, adaptive streaming, and more.
Final Thoughts
WebRTC Simulcast is the unsung hero for anyone creating high-performance real-time video platforms in 2025. It enables you to optimize video quality without sacrificing bandwidth usage and infrastructure loads — a double win for both users and developers.
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Written by

Jack morris
Jack morris
I'm Jack Morris, a VoIP specialist dedicated to helping businesses scale with expert VoIP staff augmentation and tailored VoIP solutions. With a passion for connecting companies to top-notch VoIP developers, I focus on delivering reliable and innovative communication solutions that drive success in the VoIP space.