Provide high quality and real-time audio and video interactions
In traditional live streaming, streamers deliver content in a unidirectional way. User engagement degree and lead conversion rate are low. ApsaraVideo Real-time Communication relies on Global Realtime Transport Network (GRTN) and core codec technologies and provides more interactive elements to traditional live streaming, such as voice chat, video co-streaming, and game interaction. ApsaraVideo Real-time Communication reduces the end-to-end latency to an ultra-low level and allows viewers to smoothly start or stop co-streaming. This way, viewers can not only view but also participate in live streaming, which enhances the social connection. ApsaraVideo Real-time Communication is suitable for scenarios such as online education, interactive entertainment, and audio and video communication.
ApsaraVideo Real-time Communication relies on the core technologies such as intelligent noise reduction algorithms, Narrowband HD™, and real-time transmission algorithms to deliver high-quality communication services in noisy environments with unstable network connections and a high packet loss rate.
Reusable network resources
ApsaraVideo Real-time Communication relies on the brand networks of GRTN and more than 3,200 POPs deployed around the world and reuses live streaming and Web Real-Time Communication (WebRTC) resources. This helps deliver reliable global services in poor network conditions.
Real-time interaction
Based on WebRTC, ApsaraVideo Real-time Communication provides real-time communication with a latency within 300 ms and allows users to co-stream in a channel or across-channels. Viewers can smoothly start or stop co-streaming in real time.
Ultra-high quality experience
The excellent 3A audio processing engine supports full-band sampling at 48 kHz. 720p and 1080p resolutions are supported to provide high-quality video call services.
Various media processing capabilities
ApsaraVideo Real-time Communication provides media processing capabilities, including recording, stream mixing, transcoding, and content moderation.
Easy integration
ApsaraVideo Real-time Communication supports low-code integration and provides scenario-specific integration solutions and components and comprehensive best practices to help developers quickly access the service.
Support for multiple terminals
ApsaraVideo Real-time Communication allows you to co-stream with other users on the Android, iOS, or Web clients. Users can start interactive live streaming on various platforms.
Architecture
ApsaraVideo Real-time Communication uses the WebRTC protocol to ensure real-time interaction between co-streamers. Based on the hosted room management component, ApsaraVideo Real-time Communication allows regular viewers to have the same latency as co-streamers. ApsaraVideo Real-time Communication can directly connect to the Alibaba Cloud media processing center based on the stream relay feature and provides capabilities such as recording, transcoding, stream mixing, content moderation, and connection to the third-party content moderation platform. Standard live streaming can be seamlessly extended. Viewers can pull streams over Real-Time Streaming (RTS), Flash Video (FLV), Real-Time Messaging Protocol (RTMP), and HTTP Live Streaming (HLS). The number of viewers is not limited, and the room management component is not required.
Features
Excellent audio and video processing capabilities and comprehensive quality monitoring
ApsaraVideo Real-time Communication uses the audio preprocessing capabilities, video codec, algorithm for resistance to poor networks, and data dashboards developed by Alibaba Cloud to ensure high-quality communication and comprehensive quality monitoring.
Video interaction
Multiple users can interact with each other in a video that has an end-to-end latency within 300 ms. Supported video resolutions include 480p, 720p, and 1080p. For example, a streamer in a live channel can interact with the viewers, or streamers can interact with each other by initiating a streamer challenge across live channels.
Voice interaction
High-quality audio with a 48 kHz sampling rate is supported. The end-to-end latency is within 300 ms. This feature is suitable for various scenarios such as voice chat room and karaoke.
Audio processing
ApsaraVideo Real-time Communication supports audio processing capabilities such as background music playback, music and voice mixing, in-ear monitoring, and noise reduction.
Intelligent noise reduction
Under the premise of high-fidelity human voice, peripheral noise is intelligently eliminated, sudden noise is suppressed, and device buzzing noise is eliminated.
Quality monitoring
ApsaraVideo Real-time Communication collects and analyzes large amounts of audio and video quality data and monitors the audio and video quality based on multiple dimensions and levels.
Various business features and flexible interface configurations
For various enterprise applications, ApsaraVideo Real-time Communication provides various features and flexible interface configurations to help users quickly build real-time communication scenarios.
Stream mixing and relay
Multiple streams can be mixed into a single stream based on specific rules. The single stream can be relayed to ApsaraVideo Live or a third-party platform.
CDN acceleration
Standard streaming and RTS seamlessly apply to the co-streaming feature. Viewers can smoothly start or stop co-streaming. A live channel supports more than 100,000 concurrent viewers.
Cloud-based recording
ApsaraVideo Real-time Communication can synchronize audio and video call images to the cloud for cloud-based stream mixing, and record and save the mixed channel content.
Cloud-based transcoding
ApsaraVideo Real-time Communication can mix and transcode audio and video streams, and optimize the encoding of different content in videos based on different policies, which differentiates the content in the videos.
Content moderation
ApsaraVideo Real-time Communication allows you to review videos and audio by seamlessly using the content moderation feature or by connecting to a third-party content moderation platform.
Scenarios
Co-streaming
Based on the co-streaming feature, streamers and viewers can interact with each other, streamers can interact with each other by initiating a streamer challenge, and multiple viewers can co-stream with each other. This makes live streaming more interesting. ApsaraVideo Real-time Communication reduces the end-to-end latency to within 300 ms and allows viewers to smoothly start or stop co-streaming. Standard streaming and RTS seamlessly apply to the co-streaming feature. A live channel supports more than 100,000 concurrent viewers.
Features
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Ultra-low latencyThe millisecond-level video latency allows streamers and viewers to interact with each other in real time.
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Adaptability to poor network conditions
If the network connection is unstable and the network availability is poor, ApsaraVideo Real-time Communication allows you to push streams over RTC. This improves the smoothness of stream ingest and co-streaming.
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Flexibility and ease-of-use
Push SDK is integrated with the co-streaming feature to meet the requirements of live streaming in different scenarios.
Related Services
One-to-one audio and video calls
720p and 1080p resolutions and high-quality audio with a 48 kHz sampling rate are supported. The end-to-end latency is within 300 ms during calls. This helps provide smooth and high-quality audio calls. Reusable UI components are provided for one-to-one audio and video calls. The source code of the backend service is provided to help integrators perform custom development based on their feature requirements.
Features
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Ease-of-use
Various scenario-based UI components are provided to quickly implement one-to-one audio and video calls and minimize development costs.
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Stability and reliability
Based on years of experience in the global network services of Alibaba Cloud, ApsaraVideo Real-time Communication provides end-to-end resistance to unstable network connections and ensures the stable operation of business in unstable network conditions.
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High quality
720p and 1080p resolutions and high-quality audio with a 48 kHz sampling rate are supported to provide clear and immersive interaction experience.
Related service
Multi-user audio and video calls
Up to 50 viewers can co-stream at the same time in a single live channel. 720p and 1080p resolutions and high-quality audio with a 48 kHz sampling rate are supported. Reusable UI components are provided for multi-user audio and video calls. The source code of the backend service is provided to help integrators perform custom development based on their feature requirements.
Benefits
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Ease-of-use
Various scenario-based UI components are provided to quickly implement multi-user audio and video calls and minimize development costs.
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Stability and reliability
Based on years of experience in the global network services of Alibaba Cloud, ApsaraVideo Real-time Communication provides end-to-end resistance to unstable network connections and ensures the stable operation of business in unstable network conditions.
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High quality
720p and 1080p resolutions and high-quality audio with a 48 kHz sampling rate are supported to provide clear and immersive interaction experience.
Related services
Voice Chatroom
A voice chatroom is composed of the owner, streamer, and viewer roles. The owner and streamer can make real-time voice calls with each other, and viewers can participate in voice interactions.
Features
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Low-latency audio interaction
ApsaraVideo Real-time Communication relies on the brand networks of GRTN and more than 3,200 points of presence (POPs) deployed around the world to build a low-latency transmission network, which ensures real-time communication between streamers and between streamers and viewers.
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Streamer seat and permission management
ApsaraVideo Real-time Communication allows streamers to assign or revoke streamer seats and control the permissions of each user by using data transmission channels.
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Audio mixer
The audio mixer provides features such as volume adjustment, mix ratio adjustment, and voice modification and enhancement. All streamers in a live channel can use the audio mixer.
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Background music import
Users can import and play background music from an external source in voice chat rooms. Common music formats are supported.
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Recording
Users can generate audio data in real time and record streams locally.
Related Services
Online KTV
RTC simulates offline karaoke boxes and provides interactive features, such as solo, chorus, and listening to build online karaoke boxes.
Features
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Ultra-low latency
The reliable audio transmission technology is used to ensure a low-latency singing experience for users.
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Lyrics component
The component that synchronizes lyrics with music is open source for easy customization.
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Real-time group singing
Accurate accompaniment synchronization ensures an authentic karaoke experience for online group singing.
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Song list and singer seat management
Users can manage song lists and singer seats by using data transmission channels.
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Song control
The lead singer can control the playing state of a song in real time, manage the song list, and synchronize the playing state of the song to other singers and viewers in real time.
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Audio mixer
Provides functions such as volume adjustment, mixing ratio adjustment, and bel canto voice variation.
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Low-latency in-ear monitoring
Adapt to a large number of devices, achieve low latency ear return, and provide timely feedback to the singer.
