This topic introduces a solution that uses Real-Time Streaming (RTS) to reduce live streaming latency from more than 3 seconds to 400-800 ms. It is intended for developers and operations personnel who are building and optimizing e-commerce live streaming systems.
Background
E-commerce industry
In recent years, an increasing number of e-commerce businesses have adopted live streaming. However, standard live streaming technology often has a latency of 3–6 seconds or even more, which fails to meet the high-frequency interaction demands between hosts and buyers. Furthermore, the stability of live streaming directly impact business conversion rates.
Pain points
High latency: When viewers ask questions about products, high latency prevents the host from responding in real time, degrading the viewing experience.
Frequent buffering: Comments sections are often filled with viewers complaining about buffering, forcing the host to switch networks or restart the stream, which reduces efficiency.
Slow channel switching: In mobile swiping scenarios, switching from one live stream to another can take several seconds to load.
RTS solution
Overview
RTS extends live streaming by utilizing the UDP protocol. Leveraging Alibaba Cloud's 3,200+ global points of presence (POPs) and end-to-end optimization technologies, RTS reduces e-commerce live streaming latency from over 3 seconds down to the sub-second level. RTS is characterized by its low latency, ease of integration, and reduced buffering.
Advantages
Sub-second latency
Based on 3,200+ global POPs and a smart routing system, hosts and viewers are connected to the nearest POP within the same ISP.
Flexible networking combined with a dynamic path planning system selects the optimal network transmission paths.
Upgrading the transport protocol from TCP to UDP improves transmission efficiency while ensuring reliability.
Reduced buffering
End-to-End transmission quality is optimized for audio and video streams. This ensures a smooth, uninterrupted viewing experience during minor packet loss and minimizes buffering as much as possible even in severe packet loss scenarios
Easy integration
The console provides one-click activation for RTS. You can easily generate an RTS playback URL based on your existing live stream setup for rapid deployment.
Demonstration
Industry use case
Taobao Live: Taobao Live has become a major business model in the new era of e-commerce, embraced by merchants, hosts, and consumers alike. Taobao Live has deployed RTS at scale. RTS helps Taobao Live reduce latency, minimize buffering, and handle millions of concurrent viewers, thereby boosting business conversion efficiency.
Implementation
Before you begin
Limitations
Web RTS SDK limitations: The SDK does not support videos with B-frames or AAC audio. If your stream contains B-frames or AAC audio, selecting Sub-second (End-to-End Latency: 400-800 ms) option will trigger automatic adaptive transcoding to ensure compatibility, which will incur live stream transcoding fees.
Player requirements: RTS uses the UDP protocol. The required player versions are listed in the table below:
Player type
Limitations
Mobile player
V5.4.5.0 or later
Web player
V2.0.3 or later
Configuration
Log on to the ApsaraVideo Live console.
In the navigation pane on the left, click Live + > RTS.
Select a streaming domain.
Turn on the switch and select Sub-second (End-to-End Latency: 400-800 ms).
You can integrate the SDK or follow the Specifications of the RTS signaling protocol in your development environment for production use.
NoteA streaming domain can have both RTS and standard live streaming (RTMP, FLV, and HLS) enabled at the same time. You can distinguish between them by the playback URL.
Example of a playback URL for RTS:
artc://your_streaming_domain/AppName/StreamName?access_token.Example of a standard live streaming playback URL:
rtmp://your_streaming_domain/AppName/StreamName?access_token.
Verify the solution
This section describes how to verify the solution using the OBS streaming tool (For more information, see OBS introduction) and the RTS mobile demo.
Step 1: Generate ingest and streaming URLs
Generate signed ingest and streaming URLs. For more information, see URL generator.
Step 2: Ingest the stream with OBS
Open OBS. In the left pane, select Stream. In the Server text box, enter the generated ingest URL. For example:
rtmp://***push1.ialicdn.com/test***/test?auth_key=1643******-0-0-a922892e06ee18016640e0fe14******.
In the left pane, select Output. Set Keyframe Interval to no more than 3 seconds and Profile to baseline. Return to the home page, select a video source, and start streaming.
NoteWith these parameters, you can achieve 700–900 ms latency.
Step 3: Play the RTS video
This guide uses the mobile demo app for playback verification.
The mobile demo is only available for Android 4.3 and later. For other verification tools and demos, see Run demos.
Billing
Billing rules
RTS is billed under a new pricing model, different from standard live streaming:
RTS is billed according to its own pricing items; charges are not stacked on top of standard live streaming fees.
RTS can be billed by either data transfer (traffic).
Any changes to your standard live streaming billing method will also apply to RTS.
For more information about RTS billing, see RTS billing.
FAQ
Can viewers watch the same live stream using both standard and real-time streaming?
Yes. For the same live event, separate playback URLs are generated for standard and RTS viewing. For example, a standard RTMP stream would use rtmp://{streaming_domain}/{AppName}/{StreamName}?{auth_key}, while the RTS stream would use artc://{streaming_domain}/{AppName}/{StreamName}?{auth_key}.
Can I integrate RTS using a self-developed SDK?
Yes. To do so, you must follow Alibaba Cloud's signaling protocol specification. For details, see Specifications of the RTS signaling protocol.
Which browsers support RTS playback?
RTS allows standard WebRTC access. Any browser that is compatible with WebRTC can be used for playback. For browser compatibility, see Browser compatibility with WebRTC.