As a lower-latency version of LVB, LEB provides superb live streaming experience with millisecond playback latency, far lower than that of live stream playback using traditional protocols. LEB is designed for scenarios with high latency requirements. In addition to live shopping and online education, it is also suitable for interactive scenarios such as live sports streaming and live game streaming.
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Millisecond-level ultra-low latency playback | By adopting the UDP protocol, millisecond-level latency live streaming capability is achieved in high-concurrency scenarios, improving the traditional live streaming drawback of 3-5 seconds latency. At the same time, it takes into account core indicators such as instant start and lag rate, providing users with an ultimate ultra-low latency live streaming experience. |
Comprehensive features, smooth compatibility | Compatible with all standard live streaming features, including push, transcoding, recording, screenshot, content moderation, and playback, it supports customers to smoothly migrate from existing standard live streaming services. |
Wide coverage of acceleration nodes and high bandwidth capacity | Currently, Live Event Broadcasting (LEB) has super acceleration nodes with global distribution and extensive coverage (supporting 2000+ nodes and 25 countries), capable of supporting 100T+ bandwidth. |
Easy to use | Integration is straightforward, requiring no additional plugins. It can span multiple platforms, supporting a variety of operating systems and devices. |
Excellent network resilience | In various weak network environments (such as high packet loss and high latency), Live Event Broadcasting (LEB) can still ensure high-quality video streams, providing users with a more stable live streaming experience. |
Web low-latency support | Currently, CDN live streaming only supports HLS format streams on the web, but this format has a playback latency of several seconds. Live Event Broadcasting (LEB) can also support web playback with only a few hundred milliseconds of latency. |
Smooth transition between multiple bitrates | Seamlessly switch between transcoded streams with different bitrates without any interruptions or jumps, ensuring a smooth transition in both visual and auditory experiences. |
Adaptive Bitrate Control | Adaptively switch between different bitstreams according to network bandwidth, ensuring a smooth playback experience during varying network conditions. |
Compared to Standard Live Video Broadcasting (LVB), Live Event Broadcasting (LEB) uses the WebRTC protocol during the playback process, thus it has a lower playback latency than Standard LVB. The following will specifically compare the differences between LEB.
Protocol Comparison
Currently, Live Video Broadcasting (LVB) uses common playback protocols such as RTMP, FLV, and HLS. The common feature of these protocols is that they are all based on the TCP protocol. TCP has delayed acknowledgment and piggybacking, which means that it does not immediately respond with an ACK for each received data but waits for a certain amount of data before responding. This can lead to a perceived delay, and in weak network scenarios, this can even result in delays of several seconds or even tens of seconds.
In order to achieve lower latency in live streaming, faster and better-quality playback protocols are needed. Research shows that low-latency live streaming protocols in the industry include QUIC, SRT, WebRTC, and ORTC, all of which are based on the UDP protocol at the underlying level. Comparatively, QUIC has a higher latency because it does not have streaming features; SRT, WebRTC, and ORTC all have millisecond-level latency and streaming features. Among them, SRT and ORTC are less commonly used in the industry, while WebRTC is widely used and has a thriving technical ecosystem.
A good ecological environment is an important consideration for Live Event Broadcasting (LEB) to carry out low-latency transformation using WebRTC. Most popular browsers such as Chrome and Safari already support the WebRTC standard, and mature open-source WebRTC SDKs.
Low-latency live streaming services in the industry use protocols such as QUIC, SRT, WebRTC, and ORTC. Among these, the latency of QUIC is relatively high because it does not have the characteristics of streaming media. SRT, WebRTC, and ORTC have streaming media characteristics and can stream with millisecond latency, but SRT and ORTC are not as widely used as WebRTC. As a result, LEB uses the UDP-based WebRTC to implement low-latency live streaming.
Latency Comparison
The latency of the FLV protocol in standard live streaming is generally between 2 and 10 seconds. The main factors contributing to its latency are the GOP size and TCP backlog in weak network transmission. The latency of HLS is even higher, ranging from several seconds to tens of seconds. The main factors causing HLS latency are the GOP size and TS size, making it the highest latency in standard live streaming.
In contrast, Live Event Broadcasting (LEB) adopts the WebRTC playback protocol based on the UDP protocol, which enables millisecond-level latency between nodes. The latency of LEB is typically between 300 ms and 1000 ms.
Weak Network Resilience Comparison
Live Event Broadcasting (LEB) has stronger weak network resilience compared to Standard Live Video Broadcasting (LVB), providing a more stable playback experience. To further demonstrate the weak network advantages of LEB, we conducted tests on video lag rate and other indicators for LEB and Standard LVB under normal and weak network environments. The specific test results are as follows:
Test Scenarios
The publisher uses RTMP for streaming, and the audience plays FLV and LEB streams separately, with lag rate and other indicators recorded. The publisher has a lossless network, while the audience's network is set to different weak network conditions for testing. The main test indicators are frame rate and lag rate.
Stream Parameter Configuration
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Resolution | 720 × 1080 |
Bitrate | 1800 kbps |
Frame Rate | 15 |
Comparison of Key Indicators in Several Weak Network Scenarios
Video Frame Rate
Video Lag Rate
Audio Lag Rate
Parameter Description
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Video Lag Rate | Video rendering intervals greater than 500 milliseconds are considered as lags. The total duration of all lags divided by the total playback duration is the lag rate. |
Audio Lag Rate | Audio playback intervals greater than 200 milliseconds are considered as lags. The total duration of all lags divided by the total playback duration is the lag rate. |
Video Frame Rate | The number of frames played per second in a video. |
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