Understanding the Differences Between Adaptive Bitrate (ABR) and Multi-Bitrate (MBR) Streaming

Streaming technology has evolved significantly over the past decade, with a primary focus on delivering high-quality video content efficiently across various devices and network conditions. Two common and confused keywords in this domain are Adaptive Bitrate (ABR) and Multi-Bitrate (MBR) streaming. While they might seem similar at first glance, they have distinct mechanisms and use cases. Before diving into the differences, let's clarify what bitrate means in the context of video streaming. Bitrate refers to the amount of data transmitted per second. A higher bitrate generally equates to better video quality, but it also requires more bandwidth. ?

Multi-Bitrate Streaming

Multi-bitrate (MBR) streaming involves providing multiple versions of the same video at different bitrates. 1 For instance, a video might be available in 360p, 720p, and 1080p. The viewer manually selects the desired quality based on their network conditions and preferences. Here’s how it works:

1. Pre-encoded Streams: Multiple versions of the video are pre-encoded at different bitrates and stored on the server.
2. Initial Selection: When a user initiates playback, the client selects a stream based on the available bandwidth and device capabilities.
3. Fixed Stream: Once a stream is selected, it does not change dynamically in response to fluctuations in network conditions during playback.

Adaptive Bitrate Streaming

Adaptive bitrate (ABR) streaming is a more sophisticated approach. The video player dynamically adjusts the video quality based on the viewer's available bandwidth and device capabilities. It automatically switches between different bitrate versions of the video to ensure smooth playback without buffering. Here’s how ABR works:

1. Segmented Video: The video is divided into small segments (typically a few seconds each), with each segment available at multiple bitrates.
2. Continuous Monitoring: The client continuously monitors the network conditions and buffer status.
3. Dynamic Switching: Based on real-time analysis, the client switches between different bitrate streams to maintain smooth playback and optimal quality.

How can I Achieve Multi-Bitrate in WebRTC Streaming?

Multi Bitrate can be achieved while streaming in WebRTC. However, it's typically implemented through a combination of techniques rather than a direct feature of the WebRTC protocol itself. ?

Commonly it is done using below flows:

1. Simulcast: This involves sending multiple video streams simultaneously at different resolutions and bitrates. The WebRTC client selects the best available stream based on network conditions.
2. Scalable Video Coding (SVC): SVC allows the encoder to create a base layer and enhancement layers. The client can decode the base layer for lower quality or add enhancement layers for higher quality based on available bandwidth. ?

Protocols Supporting Adaptive Bitrate Streaming

While WebRTC itself doesn't directly handle adaptive bitrate streaming, several protocols are used in conjunction with it to facilitate this functionality:

1. HTTP Live Streaming (HLS): This protocol breaks down the video into small segments and provides multiple bitrate versions of each segment. ?
2. Dynamic Adaptive Streaming over HTTP (DASH): Similar to HLS, DASH offers adaptive streaming capabilities but with more flexibility in terms of segment formats and packaging.
3. Smooth Streaming: Developed by Microsoft, Smooth Streaming is another protocol for adaptive bitrate streaming.

Achieving Adaptive Bitrate in WebRTC

Adaptive bitrate in WebRTC is primarily driven by the client-side player, with the server providing necessary information.

Server-Side Responsibilities:

1. Simulcast: Send multiple video streams at different resolutions and bitrates.
2. SVC: Encode video using SVC.
3. Provide metadata: Include information about available bitrates, codecs, and resolutions in the SDP (Session Description Protocol).
4. Bandwidth estimation: While not mandatory, the server can provide bandwidth estimates based on network conditions.

Client-Side Responsibilities:

1. Bitrate estimation: Continuously monitor network conditions (packet loss, jitter, available bandwidth) to estimate the optimal bitrate.
2. Request switching: Request different video streams based on the estimated bitrate.
3. Buffer management: Maintain a sufficient buffer to avoid playback interruptions.
4. Quality adjustment: Smoothly transition between different bitrates to prevent noticeable quality changes.

How the Source Knows About Network Congestion

The source (client-side player) determines network congestion through several indicators:

1. Packet loss: A high packet loss rate indicates potential network congestion.
2. Jitter: Increased jitter suggests network instability.
3. Buffering: Frequent or long buffering periods indicate insufficient bandwidth.
4. Video quality degradation: Visible artifacts or pixelation suggest a need to lower the bitrate.

Both multi-bitrate and adaptive bitrate streaming play crucial roles in delivering high-quality video content to viewers. While multi-bitrate provides a static set of options for users to choose from, adaptive bitrate offers a dynamic solution that automatically adjusts video quality based on network conditions.

WebRTC, while not inherently supporting multi-bitrate or adaptive bitrate, can be effectively integrated with techniques like simulcast and SVC to achieve multi-bitrate streaming. For adaptive bitrate, the client-side player plays a pivotal role in monitoring network conditions and requesting appropriate bitrate streams.