Reduce live streaming latency - causes and solutions - ApsaraVideo Live

Causes of live streaming latency

Stream ingest client

A Group of Pictures (GOP) is a sequence of frames starting with a keyframe and serves as the basic encoding and decoding unit. A large GOP size increases latency because the player must load more frames before playback begins.
Third-party ingest applications often increase the encoding buffer to prevent stuttering, but a large buffer adds latency.
Hardware limitations may prevent the encoder from meeting bitrate, frame rate, or high-profile encoding requirements, causing delays.

Server-side

The server caches stream data before playback to ensure fast startup and reduce stuttering, which introduces latency. Network jitter during playback can add a further 2-3 second delay.

Playback client

Most playback clients without fast-forward support begin decoding only after the receive buffer is full. The time to fill this buffer adds latency.

Reduce latency

To reduce latency, adjust the following settings:

Configure the stream ingest client
- Set the GOP size to 1 or 2 seconds to reduce player load time. Configure this in the Custom transcoding console or through the AddCustomLiveStreamTranscode API.
- Use the Alibaba Cloud Push SDK to reduce a large encoding buffer that causes high latency.
- Use hardware encoding on iOS for higher efficiency and lower power consumption. Use software encoding on Android — hardware encoding may cause compatibility issues across device models and CPU types.

Select an appropriate streaming protocol

ApsaraVideo Live supports three playback protocols: HTTP-FLV, HLS, and RTMP. Select one based on your scenario.

HTTP-FLV and RTMP offer lower latency for real-time playback. HLS has higher latency but broader device compatibility.
HTTP-FLV and RTMP require a Flash player. HLS plays directly in browsers.
Mobile browsers require HLS.

Note

With HLS, 10-30 seconds of latency is normal. To reduce it, adjust the server configuration or switch to FLV.

The following table compares the HTTP-FLV, HLS, and RTMP protocols.

Protocol	Description	Transport protocol	Video container format	Recommended scenario
HTTP-FLV	Encapsulates streaming data in FLV format over HTTP. Latency is about 2 seconds. Supports HTTPS and Android.	HTTP	FLV, TAG	Low-latency streaming
HLS	Segments streams into TS shards (typically 3-4 shards of 5+ seconds each), resulting in 10-30 seconds latency but smooth playback. Mainly used on iOS for live streaming and recording.	HTTP	M3U8, TS	Cross-platform streaming
RTMP	Splits messages into chunks transmitted over TCP. The receiver reassembles chunks into stream data. Can be unstable. iOS requires a third-party decoder.	HTTP	FLV, TAG	Interactive streaming

Configure the server
Reduce the server-side cache to decrease latency. Configure latency in the console — a lower setting means a smaller cache. Settings vary by protocol.
1. Log on to the ApsaraVideo Live console.
2. In the navigation pane on the left, click Domain Names to go to the Domain Management page.
3. For the streaming domain that you want to configure, click Domain Settings in the Actions column.
4. Click Streaming Management > Latency Settings.
Note
A smaller cache may cause stuttering on unstable networks because data cannot be downloaded in time. Configure this setting as described in Delay Configuration.

Real-Time Streaming (RTS)

If the preceding methods are insufficient, use Real-Time Streaming (RTS). RTS delivers millisecond-level latency for tens of millions of concurrent viewers, overcoming the 3-6 second latency of traditional live streaming with ultra-low latency, minimal stuttering, and fast startup.

RTS costs

The costs for RTS differ from standard live streaming. Check billing rates on the Alibaba Cloud Pricing page.
Activate the RTS service

The RTS overview explains how to activate and use the RTS service.