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 uses the WebRTC protocol to ensure low latency. It adopts the Opus codec and does not support B-frames. If the original stream contains B-frames or the codec is not Opus, CSS backend will remove the B-frames and transcode the stream to Opus format, which will incur standard transcoding fees.
You can integrate the MLVB SDK with apps on iOS and Android clients for live push and playback.
The MLVB SDK uses CSS, IM, TRTC and other services for low-latency audiovisual communication for multiple parties. It offers co-anchoring for interaction between viewers, and other viewers who don’t join co-anchoring can also watch the live streaming. For details, please see Co-anchoring.
You can use the following ways to achieve live push and playback on your websites:
WebRTC push uses the Opus audio codec. If you use a standard live streaming protocol (RTMP, HTTP-FLV, or HLS) for playback, the CSS backend will automatically convert the audio streams to AAC format to ensure normal playback, which will incur audio transcoding fees. For details, please see Live Transcoding > Audio Transcoding. Audio transcoding will not be initiated when only LEB service is used.
On a browser which does not support WebRTC, a WebRTC URL passed into the player will be converted to ensure normal playback. By default, WebRTC URLs are converted to HLS URLs on mobile browsers and HTTP-FLV URLs on desktop browsers.