RTS configuration flow changes-ApsaraVideo Live(LIVE)-阿里云帮助中心

The Alibaba Cloud Real-Time Streaming (RTS) console has been upgraded to simplify configuration. Learn how to migrate legacy configurations to the new version and get started with the new interface.

Overview of changes

The RTS console has been fully upgraded to simplify enabling, configuring, and testing RTS stream ingest and playback. The RTS 1.0 and 2.0 concepts have been removed to reduce the learning curve and eliminate the need to manage version differences.

Comparison of changes

Legacy version

In the legacy version, RTS 1.0 offered an end-to-end latency of 500 ms to 1,000 ms, required SDK integration only on the playback side, and was forward-compatible with standard live streaming. RTS 2.0 delivered an end-to-end latency of 200 ms to 400 ms and required RTS SDK integration for both stream ingest and playback to implement a full WebRTC pipeline. An RTS Playback switch was available at the bottom of the page.

On the legacy RTS Ingest configuration page, the features of Low-latency Ingest (RTS 1.0) and Real-time Ingest (RTS 2.0) were compared side by side. The RTS Ingest switch was enabled and set to Low-latency Ingest (RTS 1.0), with a prompt indicating that an HTTPS certificate was required.

Configurations were separated into versions 1.0 and 2.0. Different versions corresponded to different protocols and latencies.
Browsers did not support video B-frames or audio AAC during RTS playback. You had to enable or disable H5 automatic transcoding in the RTS configuration of the streaming domain.
Ingest and streaming domains had to be configured separately. Inconsistent configurations often caused service unavailability.
You could not directly generate a streaming URL in the console to test the service.

New version

When you select the half-second latency option on the new configuration page, a warning indicates that it is not compatible with standard live streaming and that you cannot use cloud features such as transcoding or recording.

Version distinctions have been removed. Instead, select an option based on your stream ingest and playback protocols:
- ARTC ingest and playback only: end-to-end latency is 200 ms to 400 ms.
- RTMP/ARTC ingest + HLS/FLV/RTMP/ARTC playback: end-to-end latency is 400 ms to 800 ms.
After you enable Real-Time Streaming (RTS) and select sub-second latency, the system automatically enables H5 automatic transcoding to remove B-frames and transcode audio to Opus. No extra configuration is required.

Note
H5 automatic transcoding is triggered only during browser playback and cannot be manually disabled. If you need to disable this feature, submit a ticket.

If this feature was disabled in the legacy version, it remains disabled. To enable it, toggle the Real-Time Streaming (RTS) switch off and on again. This action enables the feature by default.
Ingest and streaming domains are configured together to ensure consistency and prevent service unavailability.

Note
If the ingest and streaming domain configurations were inconsistent in the legacy version, the new interface displays a configuration error. You can click Reconfigure to adjust the settings.
You can now generate a streaming URL in the console to test RTS stream ingest and playback.

Note
For quick testing, the ApsaraVideo Live console uses a proxy signaling domain. However, when you use the Web RTS SDK for RTS stream ingest and playback, you must also configure an SSL certificate and configure an HTTP header, such as the Access-Control-Allow-Origin response header (for example, *), for your domain.

Handling abnormal configurations

In the legacy RTS configuration, inconsistent ingest and streaming domain settings often caused stream ingest or playback failures.

Case	Streaming domain	Ingest domain	Impact
Case 1	Not enabled	2.0	RTMP stream ingest is not supported, causing playback issues.
Case 2	1.0	Not enabled	ARTC stream ingest is not supported, causing playback issues.
Case 3	2.0	Not enabled	ARTC stream ingest is not supported, causing playback issues.
Case 4	1.0	2.0	RTMP stream ingest is not supported, causing playback issues.
Case 5	2.0	1.0	Playback issues occur.

Go to the Real-Time Streaming console and click Reconfigure to correct the settings. If the issue persists, submit a ticket.

Sub-second latency

Features: 400–800 ms end-to-end latency. Compatible with standard live streaming and other ingest and playback protocols.
Supported protocols: ARTC (WebRTC-based) for ingest and playback. Forward-compatible with standard live streaming: supports RTMP ingest, origin pull, and RTMP/FLV/HLS playback.
Ingest and playback tools: Compatible with most common tools. Recommended: Alibaba Cloud Live Ingest SDK and Alibaba Cloud Live Player SDK.
Other notes: If the stream contains B-frames or non-Opus audio, the system automatically removes B-frames and transcodes audio to Opus for H5 playback, incurring live transcoding fees.

Alibaba Cloud Web SDK playback: Native WebRTC does not support B-frames and uses Opus audio. The system auto-triggers transcoding based on the ingest content:
- If the stream contains B-frames and AAC-encoded audio, video transcoding removes B-frames and audio transcoding converts AAC to Opus. You are charged standard video transcoding fees based on the source resolution.
- If the stream does not contain B-frames but uses AAC audio, audio transcoding converts AAC to Opus. You are charged for audio-only transcoding.
- If you also configure a live transcoding template, B-frame removal and Opus transcoding apply in addition to the template. You are charged for the combined transcoding operations.
Alibaba Cloud Native SDK playback: The SDK natively supports B-frames and AAC, so no automatic transcoding is triggered and no additional fees apply.

Half-second latency

Features: 200–400 ms end-to-end latency. Best for scenarios that demand the lowest possible latency.
Supported protocols: ARTC (WebRTC-based) only. Not compatible with standard live streaming: does not support RTMP ingest, origin pull, or RTMP/FLV/HLS playback.
Ingest and playback tools: For ingest, you can use OBS with the WHIP protocol or the Alibaba Cloud Live Ingest SDK. For playback, you must use the Alibaba Cloud Player SDK.
Other notes: The streaming domain cannot use cloud features like transcoding or recording. Use a restreaming configuration to push the stream to another streaming domain for these features.
- Click Configure Stream Relay and select an ingest domain to receive the restreamed RTMP stream. The domain must be associated with a streaming domain and have the ultra-low latency half-second mode and dual-stream disaster recovery features disabled. For example, if push.example.com is associated with pull.example.com, you can pull the stream from pull.example.com using standard protocols and use transcoding and recording.