Tencent Real-Time Communication

Overview

Leveraging Tencent's 21 years of experience in network and audio/video technologies, Tencent Real-Time Communication (TRTC) offers multi-person audio/video calls and low-latency interactive live streaming solutions. With TRTC, you can quickly develop cost-effective, low-latency, and high-quality interactive audio/video services.

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TRTC Documentation

Strengths

Cross-Platform Global Interconnection

Implementing authentic cross-platform compatibility in the industry, TRTC is perfectly compatible with over 5,000 device models and provides client SDKs and TencentCloud APIs for mobile devices and desktops.

High-Quality Communication on Weak Networks

Tests show that with a packet loss prevention rate of over 80% and network jitter prevention of over 1,000 ms, TRTC can guarantee smooth and stable audio/video communication of high quality even in weak network environments.

Easy and Quick Integration

TRTC provides a rich set of scenario-based custom components to help you quickly implement various features in the simplest way, such as voice chat, conferencing, interactive live streaming, and interactive teaching.

Scenarios

Entertainment and Social Networking

Voice Chat Room

Co-anchor live streaming

Online Karaoke

Radio Station

Web Interactive Live Streaming

Online Education

Interactive Big Class

Interactive Small Class

AI Class

Collaborative Communication

Multi-Person Audio Call

One-to-One Audio Call

Multi-Person Video Call

One-to-One Video Call

Online Meeting

Video Customer Service

Interactive Gaming

Werewolf

Script Game

Live Q&A

Online Healthcare

Video Consultation

Online First Aid

Online Finance

Audiovisual Recording

Insurance Loss Assessment

Voice Chat Room

TRTC allows up to 50 users to mic on and chat at the same time and smoothly mic on/off with a latency of below 300 ms.
It supports multiple sound effects such as voice changing, stereo sound, ambient sound effects, and reverb, enriching the voice chat experience.
It can be used in combination with Tencent Cloud IM to support various chat modes such as public screen chat, private chat, and group chat, as well as diversified message types such as on-screen comment, like, and gift, delivering an excellent chat experience.
It provides a scenario-based component that can be directly reused to minimize the development costs. Click here to see the component guide.

Co-anchor live streaming

TRTC supports cross-room anchor competition with a latency of below 300 ms and allows viewers to co-anchor with anchors and mic on/off smoothly, eliminating the barriers to interaction and creating an engaging live room atmosphere.
It can be used in combination with Tencent Cloud IM to support various chat modes such as public screen chat, private chat, and group chat, as well as diversified message types such as on-screen comment, like, and gift, delivering an excellent live chat experience.
It supports smart beauty filters, such as skin smoothing, skin brightening, eye enlarging, and face slimming, and allows quickly creating a natural makeup effect, making anchors more charming.
It provides a scenario-based component that can be directly reused to minimize the development costs. Click here to see the component guide.

Online Karaoke

TRTC supports 48 kHz sample rate, 128 Kbps bitrate, and stereo sound, delivering a high audio quality comparable to CD.
It features an ultra-low latency below 300 ms, making it ideal for multi-person singing.
It provides multiple sync mechanisms such as message passthrough and timestamp, ensuring a precise alignment among the accompaniment, vocal, and lyrics.
It also supports in-ear monitoring, helping users carry a tune with ease.

Radio Station

TRTC supports 48 kHz sample rate, 128 Kbps bitrate, and stereo sound, delivering a high audio quality comparable to CD.
It allows using local music files in formats such as MP3, AAC, and WAV as background music for human voice.
It provides diversified voice changing effects such as middle-aged man, young girl, and metallic, making the radio more entertaining.
It provides a scenario-based component that can be directly reused to minimize the development costs. Click here to see the component guide.

Web Interactive Live Streaming

TRTC supports web-based interactive live streaming and offers various features such as stream pull/push, device management, chat, like, and sharing on WeChat.
It can meet ultra-high-concurrency watch needs and allows 100,000 viewers to watch a live stream simultaneously, with a latency of below 1 second.
It provides a scenario-based component and encapsulates easy-to-use APIs to minimize the development costs. Click here to see the component guide.

Interactive Big Class

TRTC allows up to 50 teachers and students to co-anchor interactively. They can mic on/off smoothly as needed without waiting for switchover, and the communication latency can be down to below 300 ms.
It supports live streaming to up to 100,000 concurrent students with the watch latency down to 1,000 ms in low-latency live streaming mode. There can be an unlimited number of viewers in relayed live streaming mode.
It offers various class application features such as screen sharing, interactive whiteboard, and recording and playback, diversifying online teaching methods.
It provides a scenario-based component that can be directly reused to minimize the development costs. Click here to see the component guide.

Interactive Small Class

TRTC supports interactive small classes in multiple sizes such as 1-to-1, 1-to-2, 1-to-6, and 1-to-32, where the teacher can interact with students with a latency below 300 ms, delivering a smoother teaching experience.
It offers various class application features such as screen sharing and courseware sharing, which diversify the online teaching methods.
It allows the entire lecture to be recorded and then played back on demand later, helping enhance the learning results.
It supports interactive whiteboard that features interactive graffiti, real-time track sync, real-time recording and playback of whiteboard writing and audio/video, delivering a vivid interactive experience comparable to offline teaching.

AI Class

TRTC supports online lecturing by AI virtual teachers, where students can interact with the AI teachers and enjoy a real-time and efficient AI learning experience.
It offers a server-side SDK to flexibly push teaching videos in real time and make AI teaching more intelligent.
AI teachers provide personalized feedback to students, delivering differentiated teaching services to each of them.

Multi-Person Audio Call

TRTC allows 300 users to call at the same time and up to 50 of them to mic on simultaneously and supports 48 kHz sample rate and 128 Kbps bitrate, making audio communication clearer.
With a call latency of less than 300 ms, packet loss prevention rate of over 80%, and network jitter prevention of over 1,000 ms, it guarantees smooth and stable calls even in weak network environments.
It features excellent 3A audio processing to restore human voice, cancel ambient sound, and suppress echo and howling.
It provides a scenario-based component that can be directly reused to minimize the development costs. Click here to see the component guide.

One-to-One Audio Call

With a call latency of less than 300 ms, packet loss prevention rate of over 80%, and network jitter prevention of over 1,000 ms, TRTC guarantees smooth and stable audio calls even in weak network environments.
It supports 48 kHz sample rate, 128 Kbps bitrate, and stereo sound, delivering a smooth and excellent audio call experience.
By integrating with the rich set of call signaling management APIs provided by Tencent Cloud IM, it is ideal for various audio call scenarios.
It provides a scenario-based component that can be directly reused to minimize the development costs. Click here to see the component guide.

Multi-Person Video Call

TRTC supports multi-person video call and provides 720p and 1080p HD image quality, with one single room able to sustain up to 300 concurrent online users and allow 50 of them to enable video simultaneously.
With a packet loss prevention rate of over 70% and network jitter prevention of over 1,000 ms, it guarantees smooth and stable calls even in weak network environments.
It has diverse features such as instant messaging, video on-demand, recording, and porn detection, making it ideal for different use cases.
It provides a scenario-based component that can be directly reused to minimize the development costs. Click here to see the component guide.

One-to-One Video Call

TRTC supports one-to-one video call and provides 720p and 1080p HD image quality, delivering a high-quality video call service.
With a call latency of less than 300 ms, packet loss prevention rate of over 70%, and network jitter prevention of over 1,000 ms, it guarantees smooth and stable calls even in weak network environments.
It has diverse features such as instant messaging, screen sharing, recording, and interactive whiteboard, making it ideal for a lot of use cases.
It provides a scenario-based component that can be directly reused to minimize the development costs. Click here to see the component guide.

Online Meeting

TRTC provides 1080p HD image quality and 48 kHz high sound quality with an audio/video latency of below 300 ms, delivering a smooth HD meeting experience.
It supports screen sharing and file sharing to make meetings more efficient.
By integrating with Tencent Cloud IM, it offers multiple discussion methods such as text/image-based communications without interrupting the conferencing process.
It provides a scenario-based component that can be directly reused to minimize the development costs. Click here to see the component guide.

Video Customer Service

With a call latency of less than 300 ms, packet loss prevention rate of over 70%, and network jitter prevention of over 1,000 ms, TRTC guarantees smooth and stable calls even in weak network environments.
It enables interconnection across various platforms, including mobile application, PC, WeChat Mini Program, and web, making the video customer service accessible anytime, anywhere.
It offers convenient screen sharing and file sharing features and supports communication in various forms such as text and image with the aid of Tencent Cloud IM, which helps quickly locate problems.
It can record and play back the video customer service process, helping ensure the service quality.

Werewolf

With a call latency of less than 300 ms, packet loss prevention rate of over 80%, and network jitter prevention of over 1,000 ms, TRTC guarantees smooth and stable Werewolf gameplay even in weak network environments.
It features excellent 3A audio processing to restore human voice, cancel ambient sound, and suppress echo and howling.
It supports the real-time monitoring of user network status so as to ensure an outstanding gaming experience.
Plus, it can test user audio devices to eliminate exceptional muting and further improve the gaming experience.

Script Game

With a call latency of less than 300 ms, packet loss prevention rate of over 80%, and network jitter prevention of over 1,000 ms, TRTC guarantees smooth and stable script gameplay even in weak network environments.
It features excellent 3A audio processing to restore human voice, cancel ambient sound, and suppress echo and howling.
It can be used in combination with Tencent Cloud IM to provide a rich variety of message types, so as to meet all needs in game social networking scenarios.
It supports interactive whiteboard to assist with inferencing to find out the real culprit.

Live Q&A

TRTC supports low-latency live streaming under high concurrency, which minimizes viewers' watch latency and is suitable for displaying questions synchronously on viewer clients.
It provides multiple sync mechanisms such as message passthrough, timestamp, and signaling channel so as to precisely display the audios, images, and questions in sync.
It can satisfy the ultra-high-concurrency IM interaction needs and supports quiz interaction, result statistics collection, and co-anchoring in live streaming, making online Q&A more fun.
It can filter keywords and hints in real time during interactive live streaming, which helps improve the user experience and reduce the risks of business non-compliance.

Video Consultation

TRTC supports 1080p HD image quality and allows you to flexibly adjust the video device focus, which is highly comparable to the medical consultation experience in hospitals.
It has a lot of convenient features such as file sharing, screen sharing, and instant messaging for medical record and image sharing, greatly improving the consultation efficiency.
It supports real-time recording and VOD playback, making online consultation more efficient.
Moreover, it supports multi-person video call, where multiple physicians and patients can join an online consultation session, helping deliver a smooth healthcare communication and collaboration experience.

Online First Aid

TRTC enables doctors to use audio/video calls to guide personnel at the site of accidents through first aid procedures before medical help arrives.
With a packet loss prevention rate of over 80%, network jitter prevention of over 1,000 ms, and strong immunity to bad network condition, it can easily cope with weak network environments.
It supports VoIP and is connected to first-aid call centers to provide a one-stop online first-aid solution.

Audiovisual Recording

TRTC provides a real-time on-cloud recording feature throughout the entire call that supports recording file storage, playback, and download as well as deployment on local servers, helping ensure your business compliance.
With a call latency of less than 300 ms, packet loss prevention rate of over 70%, and network jitter prevention of over 1,000 ms, it guarantees smooth and stable calls even in weak network environments.
In addition, by leveraging Tencent's 21 years of experience in data security, it can guarantee data security at its best.

Insurance Loss Assessment

TRTC enables video connection between clients and loss assessors for remote loss assessment, which greatly accelerates loss assessment and claim processing and improves the efficiency.
It can record the entire loss assessment and claim process in the cloud to guarantee business compliance.
It supports quick video connection through WeChat Mini Program, making video-based loss assessment more efficient.

Entertainment and Social Networking

Voice Chat Room

TRTC allows up to 50 users to mic on and chat at the same time and smoothly mic on/off with a latency of below 300 ms.
It supports multiple sound effects such as voice changing, stereo sound, ambient sound effects, and reverb, enriching the voice chat experience.
It can be used in combination with Tencent Cloud IM to support various chat modes such as public screen chat, private chat, and group chat, as well as diversified message types such as on-screen comment, like, and gift, delivering an excellent chat experience.
It provides a scenario-based component that can be directly reused to minimize the development costs. Click here to see the component guide.

Co-anchor live streaming

TRTC supports cross-room anchor competition with a latency of below 300 ms and allows viewers to co-anchor with anchors and mic on/off smoothly, eliminating the barriers to interaction and creating an engaging live room atmosphere.
It can be used in combination with Tencent Cloud IM to support various chat modes such as public screen chat, private chat, and group chat, as well as diversified message types such as on-screen comment, like, and gift, delivering an excellent live chat experience.
It supports smart beauty filters, such as skin smoothing, skin brightening, eye enlarging, and face slimming, and allows quickly creating a natural makeup effect, making anchors more charming.
It provides a scenario-based component that can be directly reused to minimize the development costs. Click here to see the component guide.

Online Karaoke

TRTC supports 48 kHz sample rate, 128 Kbps bitrate, and stereo sound, delivering a high audio quality comparable to CD.
It features an ultra-low latency below 300 ms, making it ideal for multi-person singing.
It provides multiple sync mechanisms such as message passthrough and timestamp, ensuring a precise alignment among the accompaniment, vocal, and lyrics.
It also supports in-ear monitoring, helping users carry a tune with ease.

Radio Station

TRTC supports 48 kHz sample rate, 128 Kbps bitrate, and stereo sound, delivering a high audio quality comparable to CD.
It allows using local music files in formats such as MP3, AAC, and WAV as background music for human voice.
It provides diversified voice changing effects such as middle-aged man, young girl, and metallic, making the radio more entertaining.
It provides a scenario-based component that can be directly reused to minimize the development costs. Click here to see the component guide.

Web Interactive Live Streaming

TRTC supports web-based interactive live streaming and offers various features such as stream pull/push, device management, chat, like, and sharing on WeChat.
It can meet ultra-high-concurrency watch needs and allows 100,000 viewers to watch a live stream simultaneously, with a latency of below 1 second.
It provides a scenario-based component and encapsulates easy-to-use APIs to minimize the development costs. Click here to see the component guide.

Online Education

TRTC allows up to 50 teachers and students to co-anchor interactively. They can mic on/off smoothly as needed without waiting for switchover, and the communication latency can be down to below 300 ms.
It supports live streaming to up to 100,000 concurrent students with the watch latency down to 1,000 ms in low-latency live streaming mode. There can be an unlimited number of viewers in relayed live streaming mode.
It offers various class application features such as screen sharing, interactive whiteboard, and recording and playback, diversifying online teaching methods.
It provides a scenario-based component that can be directly reused to minimize the development costs. Click here to see the component guide.

TRTC supports interactive small classes in multiple sizes such as 1-to-1, 1-to-2, 1-to-6, and 1-to-32, where the teacher can interact with students with a latency below 300 ms, delivering a smoother teaching experience.
It offers various class application features such as screen sharing and courseware sharing, which diversify the online teaching methods.
It allows the entire lecture to be recorded and then played back on demand later, helping enhance the learning results.
It supports interactive whiteboard that features interactive graffiti, real-time track sync, real-time recording and playback of whiteboard writing and audio/video, delivering a vivid interactive experience comparable to offline teaching.

TRTC supports online lecturing by AI virtual teachers, where students can interact with the AI teachers and enjoy a real-time and efficient AI learning experience.
It offers a server-side SDK to flexibly push teaching videos in real time and make AI teaching more intelligent.
AI teachers provide personalized feedback to students, delivering differentiated teaching services to each of them.

Collaborative Communication

TRTC allows 300 users to call at the same time and up to 50 of them to mic on simultaneously and supports 48 kHz sample rate and 128 Kbps bitrate, making audio communication clearer.
With a call latency of less than 300 ms, packet loss prevention rate of over 80%, and network jitter prevention of over 1,000 ms, it guarantees smooth and stable calls even in weak network environments.
It features excellent 3A audio processing to restore human voice, cancel ambient sound, and suppress echo and howling.
It provides a scenario-based component that can be directly reused to minimize the development costs. Click here to see the component guide.

With a call latency of less than 300 ms, packet loss prevention rate of over 80%, and network jitter prevention of over 1,000 ms, TRTC guarantees smooth and stable audio calls even in weak network environments.
It supports 48 kHz sample rate, 128 Kbps bitrate, and stereo sound, delivering a smooth and excellent audio call experience.
By integrating with the rich set of call signaling management APIs provided by Tencent Cloud IM, it is ideal for various audio call scenarios.
It provides a scenario-based component that can be directly reused to minimize the development costs. Click here to see the component guide.

TRTC supports multi-person video call and provides 720p and 1080p HD image quality, with one single room able to sustain up to 300 concurrent online users and allow 50 of them to enable video simultaneously.
With a packet loss prevention rate of over 70% and network jitter prevention of over 1,000 ms, it guarantees smooth and stable calls even in weak network environments.
It has diverse features such as instant messaging, video on-demand, recording, and porn detection, making it ideal for different use cases.
It provides a scenario-based component that can be directly reused to minimize the development costs. Click here to see the component guide.

TRTC supports one-to-one video call and provides 720p and 1080p HD image quality, delivering a high-quality video call service.
With a call latency of less than 300 ms, packet loss prevention rate of over 70%, and network jitter prevention of over 1,000 ms, it guarantees smooth and stable calls even in weak network environments.
It has diverse features such as instant messaging, screen sharing, recording, and interactive whiteboard, making it ideal for a lot of use cases.
It provides a scenario-based component that can be directly reused to minimize the development costs. Click here to see the component guide.

TRTC provides 1080p HD image quality and 48 kHz high sound quality with an audio/video latency of below 300 ms, delivering a smooth HD meeting experience.
It supports screen sharing and file sharing to make meetings more efficient.
By integrating with Tencent Cloud IM, it offers multiple discussion methods such as text/image-based communications without interrupting the conferencing process.
It provides a scenario-based component that can be directly reused to minimize the development costs. Click here to see the component guide.

With a call latency of less than 300 ms, packet loss prevention rate of over 70%, and network jitter prevention of over 1,000 ms, TRTC guarantees smooth and stable calls even in weak network environments.
It enables interconnection across various platforms, including mobile application, PC, WeChat Mini Program, and web, making the video customer service accessible anytime, anywhere.
It offers convenient screen sharing and file sharing features and supports communication in various forms such as text and image with the aid of Tencent Cloud IM, which helps quickly locate problems.
It can record and play back the video customer service process, helping ensure the service quality.

Interactive Gaming

With a call latency of less than 300 ms, packet loss prevention rate of over 80%, and network jitter prevention of over 1,000 ms, TRTC guarantees smooth and stable Werewolf gameplay even in weak network environments.
It features excellent 3A audio processing to restore human voice, cancel ambient sound, and suppress echo and howling.
It supports the real-time monitoring of user network status so as to ensure an outstanding gaming experience.
Plus, it can test user audio devices to eliminate exceptional muting and further improve the gaming experience.

With a call latency of less than 300 ms, packet loss prevention rate of over 80%, and network jitter prevention of over 1,000 ms, TRTC guarantees smooth and stable script gameplay even in weak network environments.
It features excellent 3A audio processing to restore human voice, cancel ambient sound, and suppress echo and howling.
It can be used in combination with Tencent Cloud IM to provide a rich variety of message types, so as to meet all needs in game social networking scenarios.
It supports interactive whiteboard to assist with inferencing to find out the real culprit.

TRTC supports low-latency live streaming under high concurrency, which minimizes viewers' watch latency and is suitable for displaying questions synchronously on viewer clients.
It provides multiple sync mechanisms such as message passthrough, timestamp, and signaling channel so as to precisely display the audios, images, and questions in sync.
It can satisfy the ultra-high-concurrency IM interaction needs and supports quiz interaction, result statistics collection, and co-anchoring in live streaming, making online Q&A more fun.
It can filter keywords and hints in real time during interactive live streaming, which helps improve the user experience and reduce the risks of business non-compliance.

Online Healthcare

TRTC supports 1080p HD image quality and allows you to flexibly adjust the video device focus, which is highly comparable to the medical consultation experience in hospitals.
It has a lot of convenient features such as file sharing, screen sharing, and instant messaging for medical record and image sharing, greatly improving the consultation efficiency.
It supports real-time recording and VOD playback, making online consultation more efficient.
Moreover, it supports multi-person video call, where multiple physicians and patients can join an online consultation session, helping deliver a smooth healthcare communication and collaboration experience.

TRTC enables doctors to use audio/video calls to guide personnel at the site of accidents through first aid procedures before medical help arrives.
With a packet loss prevention rate of over 80%, network jitter prevention of over 1,000 ms, and strong immunity to bad network condition, it can easily cope with weak network environments.
It supports VoIP and is connected to first-aid call centers to provide a one-stop online first-aid solution.

Online Finance

TRTC provides a real-time on-cloud recording feature throughout the entire call that supports recording file storage, playback, and download as well as deployment on local servers, helping ensure your business compliance.
With a call latency of less than 300 ms, packet loss prevention rate of over 70%, and network jitter prevention of over 1,000 ms, it guarantees smooth and stable calls even in weak network environments.
In addition, by leveraging Tencent's 21 years of experience in data security, it can guarantee data security at its best.

TRTC enables video connection between clients and loss assessors for remote loss assessment, which greatly accelerates loss assessment and claim processing and improves the efficiency.
It can record the entire loss assessment and claim process in the cloud to guarantee business compliance.
It supports quick video connection through WeChat Mini Program, making video-based loss assessment more efficient.

Architecture

TRTC focuses on providing multi-person audio/video call and low-latency interactive live streaming solutions for various platforms. It offers SDKs for web, Android, iOS, Electron, Flutter, Windows, and macOS for you to quickly integrate with and connect to its backend. Through its interconnection with different Tencent Cloud services such as IM, CSS, and VOD, TRTC makes it easier for you to develop a wider diversity of business scenarios.

Performance Module

In call mode, with an end-to-end latency of below 300 ms, TRTC is suitable for multi-person real-time interaction scenarios, including audio call, video call, online meeting, voice chat room, karaoke, and Werewolf, and delivers a face-to-face chat experience. In live streaming mode, with a viewer latency of below 1,000 ms, TRTC allows users to mic on/off smoothly with no need to wait and is suitable for interactive live streaming scenarios, such as interactive classroom, red packet grabbing, Q&A, and flash sales, making it more engaging than traditional CDN live streaming.

Professional Industry Consultation

For more information on use cases and technical architecture, please contact our sales and technical support teams.

Help

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England

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Local hotline to be supported for more regions

Tencent Real-Time Communication

Overview

Overview

Strengths

Scenarios

Architecture

Performance Module