What Is WebRTC Simulcast and How Does It Work?

Summary:

A single video stream doesn’t work for everyone. High-speed users want crystal-clear video, while mobile users need buffer-free playback. That’s why WebRTC Simulcast allows a sender to transmit multiple versions of the same video so each viewer gets the best possible quality based on their bandwidth. 

This blog breaks down what WebRTC Simulcast is, how it works, where it’s used, and more.

In video applications like live streaming and video conferencing, there’s something more important than ‘high-quality’ video: delivering the right quality to the right devices. A 4K video stream may look stunning on a high-speed fiber connection, but sending that same stream to a mobile user on 4G and buffering ruins the experience.

This is where we need WebRTC Simulcast. Instead of sending a single video stream at a fixed quality, Simulcast allows a WebRTC sender to transmit multiple versions of one video at different bitrates and resolutions. This way, the receiving device can pick the best-quality stream based on its bandwidth and processing power, ensuring a smooth, uninterrupted experience for every user.

Let’s break down how WebRTC Simulcast works, why it’s necessary for some video applications, how it compares to other streaming techniques and some WebRTC simulcast example use cases. 

What Is WebRTC Simulcast?

WebRTC Simulcast is a technique that enables a single video source to send multiple streams of different resolutions and bitrates to a media server. The server then selects the most suitable version for each receiver based on their network conditions.

For example:

• A desktop user on fiber internet receives a 1080p high-bitrate stream.
• A mobile user on a congested 4G network gets a 480p low-bitrate version to avoid buffering.
• A conference room setup with multiple screens might receive both a 1080p stream for the main speaker and a 720p version for participant thumbnails.

This dynamic stream selection is crucial for real-time applications like WebRTC-based video conferencing, live streaming, and virtual collaboration tools.

Simulcast vs. Multistreaming: What’s the Difference?

Simulcast isn’t the only approach to handling multiple video streams, but it’s one of the most efficient for real-time applications. Some confuse Simulcast with Multistreaming, but the two serve very different purposes. While Simulcast optimizes video quality within a platform, Multistreaming focuses on distributing a video feed across multiple platforms. It’s important to understand the difference to choose the right method for your application.

How Does WebRTC Simulcast Work?

Simulcast in WebRTC relies on RTP (Real-time Transport Protocol) and Selective Forwarding Units (SFUs) to manage multiple streams efficiently. Here’s a more detailed explanation of how WebRTC Simulcast works:

1. Encoding Multiple Video Streams

• The WebRTC sender (browser or app) captures video from the camera and encodes it into multiple layers (low, medium, and high quality).
• Each layer is tagged with a RID (RTP Stream ID) to help the SFU identify and manage the streams separately.

2. RTP Transmission to SFU (Selective Forwarding Unit)

• The multiple video layers are transmitted to an SFU, which routes them to different participants based on their bandwidth and device capabilities.
• The SFU acts as a smart relay; it does not decode or modify the streams, just forwards the most appropriate one to each user.

3. SFU Decision-Making & Bandwidth Adaptation

• The SFU monitors each participant’s network conditions in real time.
• If the user’s network is stable, they receive the best quality stream (e.g., 1080p).
• If bandwidth drops, the SFU automatically switches them to a lower-resolution stream (e.g., 480p) to prevent stuttering and buffering.

4. Client-Side Rendering & Decoding

• The receiving WebRTC client only decodes the stream it receives (no extra processing burden).
• If the user’s network improves, the SFU upgrades them back to a higher-resolution stream seamlessly.

This adaptive real-time streaming allows video conferencing platforms, live-streaming applications, and virtual collaboration tools to deliver optimal video quality to every participant, regardless of device or network conditions.

WebRTC Simulcast Use Cases 

Simulcast is widely used in two key areas:

1. WebRTC Simulcast for Video Conferencing

Platforms like Zoom, Google Meet, and Microsoft Teams rely on Simulcast to:

• Ensure each participant gets the best possible video quality based on their bandwidth.
• Reduce unnecessary load on low-power devices (like smartphones).
• Prevent video freezing or quality drops when bandwidth fluctuates.

How WebRTC Simulcast works in video conferencing:

• Each participant sends multiple resolution streams (e.g., 360p, 720p, and 1080p).
• The SFU dynamically selects the best stream for each participant based on network conditions.
• If one participant shares their screen, Simulcast allows the SFU to prioritize screen sharing at high resolution while keeping participant video feeds lower.
• If a user with poor connectivity speaks, their video may downgrade to 360p, but their audio remains unaffected, ensuring smooth communication.

2. WebRTC Simulcast for Live Streaming

For platforms like Twitch, YouTube Live, and Facebook Live, Simulcast:

• Sends multiple stream resolutions so viewers can watch at 1080p, 720p, or 480p, depending on their connection.
• Eliminates manual resolution switching by automating stream selection.
• Improves stream stability and reduces buffering for audiences with unstable networks.

How WebRTC Simulcast works in live streaming:

• A single video source creates multiple stream qualities (e.g., 1080p, 720p, 480p).
• The SFU/CDN (Content Delivery Network) analyzes each viewer’s bandwidth and delivers the most appropriate version.
• If a viewer’s connection slows down, the SFU/CDN switches them to a lower-quality stream instead of pausing playback.

WebRTC simulcast configuration

To enable Simulcast, WebRTC developers must:

• Use VP8 or VP9 codecs (H.264 doesn’t support Simulcast in WebRTC).
• Modify SDP (Session Description Protocol) attributes to define multiple encoding layers.
• Ensure SFU support (popular SFUs like Janus, Jitsi, and Medooze handle Simulcast well).
• Test network conditions with a WebRTC Simulcast test to optimize quality adaptation.

Whether it’s a video conferencing tool that needs to maintain smooth communication across diverse network conditions or a live streaming service that wants to keep viewers engaged without buffering, Simulcast ensures the best possible experience for every user. 

If your business depends on real-time video, integrating WebRTC Simulcast would be the best choice. 

So, are you looking for a WebRTC-powered solution with Simulcast support?

Hire VoIP Developers’ WebRTC development team specializes in WebRTC Simulcast integration, SFU optimizations, and fully customized video solutions. Let’s build a seamless experience for your users—your way!