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    ApsaraVideo Live:Changes to the Real-Time Streaming (RTS) configuration flow

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    ApsaraVideo Live:Changes to the Real-Time Streaming (RTS) configuration flow

    Last Updated:Oct 30, 2025

    The Alibaba Cloud Real-Time Streaming (RTS) console has been upgraded to provide an easier configuration experience. This topic describes how to migrate legacy configurations to the new version, learn the new interface, and improve your operational efficiency.

    Overview of changes

    The Real-Time Streaming (RTS) console has been fully upgraded to simplify enabling, configuring, and testing RTS stream ingest and playback. This update removes the concepts of RTS 1.0 and 2.0 to simplify the process and reduce the learning curve. The new version lets you enable and test the service more easily without handling complex version differences, which helps you get started quickly and enjoy a smoother experience.

    Comparison of changes

    Legacy version

    image

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    1. Configurations were separated into versions 1.0 and 2.0. Different versions corresponded to different protocols and latencies.

    2. Browsers did not support B-frames in video or AAC in audio during RTS playback. You had to enable or disable H5 automatic transcoding in the RTS configuration of the streaming domain.

    3. Ingest and streaming domains had to be configured separately. Inconsistent configurations could easily cause service unavailability.

    4. You could not directly generate a streaming URL in the console to test the service.

    New version

    image

    1. Versions are no longer distinguished. You can select a protocol based on your stream ingest and playback requirements:

      • 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.

    2. After you enable RTS and select sub-second latency, the system automatically enables H5 automatic transcoding to remove B-frames and convert audio to Opus. No extra configuration is required.

      Note

      H5 automatic transcoding is triggered only during browser playback and cannot be manually disabled. To disable it for special reasons, please submit a ticket.

      If this feature was disabled in the legacy version, it remains disabled in the new version. The system does not automatically change the status. To enable this feature, you can turn the RTS switch off and then on again. This action enables the feature by default.

    3. 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 configuration shows a configuration error. You can click Reconfigure in the new interface to adjust the settings.

    4. You can quickly generate a streaming URL in the console to test RTS stream ingest and playback.

      Note

      For a quick test of stream ingest and playback, the ApsaraVideo Live console uses a proxy signaling domain name. However, when you use the Web RTS SDK for Real-Time Streaming (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 your domain name.

    Handling abnormal configurations

    In the legacy RTS configuration, if the ingest and streaming domain configurations were inconsistent, stream ingest or playback would fail.

    Case

    Streaming domain RTS configuration

    Ingest domain RTS configuration

    Impact

    Case 1

    Not enabled

    2.0

    RTMP stream ingest is not supported, and playback issues occur.

    Case 2

    1.0

    Not enabled

    ARTC stream ingest is not supported, and playback issues occur.

    Case 3

    2.0

    Not enabled

    ARTC stream ingest is not supported, and playback issues occur.

    Case 4

    1.0

    2.0

    RTMP stream ingest is not supported, and playback issues occur.

    Case 5

    2.0

    1.0

    Playback issues occur.

    Go to the Real-Time Streaming console and click Reconfigure to make adjustments. If the issue persists after you retry, submit a ticket.

    image

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    Sub-second latency

    1. Features: Provides an end-to-end latency of 400 ms to 800 ms with strong compatibility. This mode is suitable for low-latency live streaming scenarios that require compatibility with standard live streaming and other ingest or playback protocols.

    2. Supported protocols: Supports stream ingest and playback over the Alibaba Real-Time Communication (ARTC) protocol, which is based on Web Real-Time Communication (WebRTC). This mode is forward-compatible with standard live streaming and retains the RTMP ingest protocol, origin fetch, and common playback protocols such as RTMP, FLV, and HLS.

    3. Stream ingest and playback tools: You can use common stream ingest and playback tools, such as the Alibaba Cloud Live Streaming Push SDK and the Alibaba Cloud Live Streaming Push SDK.

    4. Additional information: If the stream ingest contains B-frames or the audio encoding is not Opus, the backend automatically removes the B-frames and transcodes the audio to Opus for H5 playback, which incurs live stream transcoding fees.

      When you use the Alibaba Cloud Web SDK for playback, the system automatically triggers transcoding based on the ingested stream content to ensure playback quality. This happens because the underlying native WebRTC does not support B-frames and uses Opus for audio encoding.

      • If the ingested stream contains B-frames and uses AAC encoding, the system automatically triggers video transcoding to remove B-frames and audio transcoding from AAC to Opus. You are charged for standard video transcoding based on the source resolution tier.

      • If the ingested stream contains no B-frames but uses AAC encoding, the system automatically triggers audio transcoding from AAC to Opus. You are charged for audio-only transcoding.

      • If you have also configured a live stream transcoding template, the system performs B-frame removal and Opus transcoding in addition to the template's configuration. You are charged based on the combined transcoding results.

      When you use the Alibaba Cloud Native SDK for playback, automatic transcoding is not triggered and no extra fees are incurred. This is because the SDK natively supports B-frames and AAC.

    Half-second latency

    1. Features: Provides an end-to-end latency of 200 ms to 400 ms. This mode is suitable for live streaming scenarios that require extremely low latency.

    2. Supported protocols: Supports stream ingest and playback over the ARTC protocol, which is based on WebRTC. This mode is not compatible with standard live streaming. It does not support the RTMP ingest protocol, origin fetch, or playback with common protocols such as RTMP, FLV, or HLS.

    3. Stream ingest and playback tools: For stream ingest, you can use OBS with the WHIP protocol or the Alibaba Cloud Live Push SDK. For playback, you must use the ApsaraVideo Player SDK.

    4. Additional notes: The current streaming domain cannot use cloud features such as transcoding or recording. However, you can use the relay configuration to relay the live stream to another streaming domain.

      • Click Stream Push Configuration and select an ingest domain to accept the pushed RTMP stream. The domain must be associated with a streaming domain and must not have ultra-low latency (half-second) mode or dual-stream disaster recovery enabled. For example, if the receiving ingest domain is push.example.com, its associated streaming domain is pull.example.com. After the stream push is successfully configured, you can pull the stream from pull.example.com using standard live streaming protocols and use features such as transcoding and recording.