What Is WebRTC Simulcast and How Does It Work?

In live streaming and video conferencing, video quality isn’t just about being high; it’s about matching the right level of quality to each device and network.

A 4K stream can look impressive on a fast fiber network, but pushing the same stream to a mobile user on a 4G connection quickly results in a poor viewing 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 simulcast 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 concurrent streaming, and some WebRTC simulcast example use cases.

What Is WebRTC Simulcast?

WebRTC Simulcast, including implementations such as WebRTC H264 Simulcast, is a technique that enables a single video source to send multiple streams at different resolutions and bitrates to a media server. 

To understand what a simulcast is, think of a single video source delivering multiple streams at different resolutions. WebRTC H264 Simulcast is one implementation that allows this efficiently.

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 receives a 480p, low-bitrate stream 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?

Some confuse Simulcast with Multistreaming, but the two serve very different purposes. To understand simulcast, it’s essential to differentiate it from Multistreaming: Simulcast manages multiple resolutions of the same stream efficiently, while Multistreaming distributes different streams across platforms. Platforms that leverage Chrome Simulcast WebRTC can efficiently manage various video layers without sending separate streams, unlike Multistreaming.

Below are the key differences

How Does WebRTC Simulcast Work?

Simulcast in WebRTC relies on RTP (Real-time Transport Protocol) and Selective Forwarding Units (SFUs) to efficiently manage multiple streams. Implementing various layers requires careful simulcast integration so the SFU can efficiently forward the appropriate stream to each participant based on their network conditions.

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). When using Chrome WebRTC Simulcast, the browser encodes various layers so the SFU can forward the optimal stream to each participant.
• 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 automatically upgrades them to a higher-resolution stream.

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 receives the best possible video quality given 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 prioritizes the shared screen at high resolution while keeping the participant video feeds at lower quality.
• When a user has a weak connection, their video might drop to 360p, but the audio stays intact, keeping the conversation flowing smoothly.

2. WebRTC Simulcast for Live Streaming

On platforms such as Twitch, YouTube Live, and Facebook Live, Simulcast:

• Delivers multiple video resolutions, allowing viewers to watch at 1080p, 720p, or 480p based on their network speed.
• Automatically selects the best resolution, removing the need for manual adjustments.
• Enhances stream stability and minimizes buffering for users with inconsistent connections.

How Simulcast functions in live streaming with WebRTC:

• One video source supports multiple quality levels, including 1080p, 720p, and 480p, using VP8 Simulcast WebRTC to deliver multiple resolutions to viewers efficiently.
• The SFU or CDN assesses each viewer’s network speed and sends the version that best fits their connection.
• If a viewer’s connection slows, the SFU/CDN switches them to a lower-quality stream rather than pausing playback.

WebRTC simulcast configuration

To set up Simulcast, WebRTC developers need to:

• Use VP8 or VP9 codecs, or implement WebRTC H264 Simulcast solutions with compatible SFUs.
• 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 platform striving for seamless communication across varying networks or a live streaming service aiming to keep viewers engaged without interruptions, Simulcast delivers the optimal experience for every user. 

On live streaming platforms, the SFU/CDN automatically adapts the resolution per viewer, a key way how simulcast improves video quality by minimizing buffering and maintaining smooth playback. For businesses that rely on real-time video, implementing Simulcast WebRTC is a wise choice.

Hire VoIP Developer’s WebRTC experts focus on Simulcast implementation, SFU performance tuning, and tailor-made video solutions. Work with the team to create a smooth, high-quality experience tailored to your users. Contact us today!