What is Low Latency Streaming?
Low latency streaming refers to the process of delivering live video or audio content with minimal delay between the time it is captured and the time it reaches the viewer's device. It is particularly important for real-time applications like live sports events, online gaming, video conferencing, and interactive streaming platforms.
Workflow Of Latency Streaming
Low latency streaming follows a typical workflow:
The content is captured, encoded into small chunks, and delivered to a streaming server via protocols like RTMP or HLS. The server then distributes the chunks to viewers' devices, often utilizing CDNs. Finally, the viewers' devices decode and buffer the content to ensure smooth playback, accounting for network variations.
Why does Latency Occurs In Video Streaming?
The latency in streaming occurs due to several factors:
- Encoding and Packaging Latency: The time it takes to encode the captured content and divide it into chunks can introduce some delay. Efficient encoding and chunking strategies can help minimize this latency.
- Network Latency: The time it takes for the video or audio chunks to travel from the streaming server to the viewer's device depends on the network conditions, including the distance between the server and the viewer, network congestion, and the chosen transmission protocol.
- Buffering Latency: To ensure uninterrupted playback, the viewer's device buffers a certain amount of content before starting playback. This buffering introduces additional latency, especially when the buffer size is large.
How To Reduce Latency in Streaming?
To reduce latency and achieve low-latency streaming, here are some techniques:
- Chunk Size Optimization: Smaller chunk sizes reduce the time it takes for the content to reach the viewer's device. However, extremely small chunk sizes can increase overhead, so a balance must be struck.
- Real-Time Protocols: Using protocols specifically designed for low latency, such as WebRTC (Web Real-Time Communication) or the Low Latency CMAF (Common Media Application Format), can help minimize latency compared to traditional streaming protocols like HLS or DASH.
- Edge Computing and CDNs: Leveraging content delivery networks with distributed edge servers can bring the streaming content closer to the viewers, reducing network latency.
- Adaptive Bitrate Streaming (ABR): ABR techniques dynamically adjust the quality and bitrate of the video or audio stream based on the viewer's network conditions. ABR can reduce buffering and improve overall latency by selecting an appropriate bitrate.
- Buffering Optimization: Reducing the buffer size on the viewer's device can help reduce the playback delay. However, it should be done carefully to avoid interruptions caused by network fluctuations.
It's crucial to remember that attaining extremely low latency (in the range of milliseconds) can be difficult and depends on a number of variables, including the network connection of the viewer and the entire infrastructure. The goal of ongoing optimization is to strike a balance between minimal latency and seamless viewing while maintaining high visual quality.