<img src="https://trtc.io/media/233cf2e6-174a-4e60-9aa7-8f1fad484902.png" alt="trtc-meaning.png"><a href="https://trtc.io/">TRTC(Tencent RTC</a>) is a powerful cloud service designed to facilitate real-time audio and video communication. In this article, we will delve into the world of TRTC, exploring its capabilities and applications, and how it can revolutionize the way we communicate.<h2 id="What-is-TRTC?">What is TRTC?</h2>Let's start by analyzing the meaning of the name TRTC. Tencent Real-Time Communication is the official definition of TRTC.Firstly, let's decipher what TRTC stands for:T — TencentR-T — Real-TimeC — CommunicationTRTC(Tencent RTC) is a cloud-based service that enables real-time audio and video communication across various platforms. Developed by Tencent, a leading technology company known for its popular instant messaging apps QQ and WeChat, TRTC leverages the company's expertise in communication technology to provide a seamless and high-quality experience for users.<h2 id="Capabilities-of-TRTC">Capabilities of TRTC</h2>TRTC(Tencent RTC) offers cross-platform solutions for audio/video calls and low-latency interactive live streaming. With SDKs for mini programs, web, Android, iOS, Electron, Windows, macOS, and other platforms, you can quickly integrate TRTC services into your projects and connect to the TRTC backend. You can also combine TRTC with other Tencent Cloud products such as <a href="https://trtc.io/products/call">Instant Messaging(IM)</a> , Cloud Streaming Services (CSS), and Video on Demand (VOD) to explore more use cases. The figure below shows how TRTC works together with other products.<img src="https://trtc.io/media/7d88b4b8-64a9-4234-938c-428b631b3298.png" alt="trtc.png">High-quality audio and video communication: TRTC offers high-quality audio and video communication, ensuring clear and smooth conversations. It employs advanced audio and video processing algorithms, such as noise reduction and echo cancellation, to deliver an exceptional user experience.Cross-platform compatibility: TRTC supports a wide range of platforms, including Android, iOS, Windows, macOS, and web browsers. This cross-platform compatibility allows users to communicate seamlessly, regardless of the device they are using.<a href="https://trtc.io/blog/details/throughput-latency">Low latency</a>: One of the key features of TRTC is its low latency, which ensures real-time communication with minimal delays. This is particularly important for applications such as online gaming, remote collaboration, and live streaming, where real-time interaction is crucial.Scalability: TRTC is designed to handle a large number of concurrent users, making it suitable for applications that require massive scalability, such as online conferences, virtual events, and social networking platforms.Security and privacy: Tencent's Real-Time Communication service prioritizes user security and privacy. It employs end-to-end encryption to protect user data and ensure secure communication.<h2 id="Technical-Implementation-of-TRTC">Technical Implementation of TRTC</h2>The original design intention of WebRTC was to solve 1v1 audio and video call problems. However, in actual enterprise-level applications, we often need more complex functions, such as multi-person calls, real-time meeting control, cloud mixing, and real-time recording. This requires us to innovate and expand on the basis of WebRTC.TRTC(Tencent RTC) has successfully addressed this issue by introducing mirrored instances in the cloud. These mirrored instances act like "pseudo-users" running in the cloud. They can communicate data with real users in the browser, while simultaneously "translating" the received data in real-time, converting it into a data format that the TRTC system can understand, thereby integrating it into the existing TRTC system.This is akin to adding a translator who understands both Arabic and Chinese between two Arabic-speaking individuals, enabling those who originally could not understand each other to comprehend what the other party is saying. Through this method, Tencent has successfully extended its RTC capabilities to browsers that support WebRTC, thereby meeting the needs of enterprise-level applications.This innovative solution fully demonstrates Tencent's leading position and strong capabilities in real-time communication technology,and it also provides a good reference for other enterprises.<h2 id="Applications-of-TRTC">Applications of TRTC</h2>Online education: TRTC can be used to power online education platforms, enabling real-time interaction between teachers and students. Its high-quality audio and video communication, coupled with low latency, make it an ideal solution for virtual classrooms and remote learning.Remote collaboration: Businesses can leverage TRTC to facilitate remote collaboration among team members. Its cross-platform compatibility and low latency make it suitable for video conferencing, screen sharing, and real-time collaboration on projects.Live streaming: TRTC can be used to power live streaming platforms, providing high-quality audio and video communication with low latency. This ensures a smooth and engaging experience for both streamers and viewers.Telemedicine: Healthcare providers can use TRTC to offer telemedicine services, enabling real-time communication between doctors and patients. Its high-quality audio and video communication, along with its security features, make it a reliable solution for remote consultations and diagnoses.Social networking: TRTC can be integrated into social networking platforms to enable real-time audio and video communication among users. Its scalability and cross-platform compatibility make it an ideal choice for connecting people across the globe.<h2 id="Conclusion">Conclusion</h2><a href="https://trtc.io">TRTC(Tencent RTC)</a> is a powerful cloud service that offers a wide range of capabilities, making it suitable for various applications. Its high-quality audio and video communication, low latency, cross-platform compatibility, and scalability make it a game-changer in the world of real-time communication. By leveraging TRTC, businesses and individuals can revolutionize the way they communicate and collaborate, opening up new possibilities for innovation and growth.If you have any questions or need assistance, our support team is always ready to help. Please feel free to <a href="https://trtc.io/contact?utm_source=website&amp;utm_medium=blog&amp;utm_term=trtc&amp;utm_campaign=blog&amp;_channel_track_key=EB6l06UE">Contact Us</a> or join us in <a href="https://discord.gg/hq7jW7zChW">Discord</a> .

RTC Engine

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The Importance of Balancing Throughput and Latency

What are the differences between Throughput and Latency in Real-Time Communication?

How does Latency impact Throughput in Real-Time Communication?

Strategies to Optimize Throughput and Latency in Real-Time Communication

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<img src="https://trtc.io/media/732babee-e333-422d-9198-9190a110d803.png" alt="Throughput-Latency.png">Throughput is characterized as the volume of data transmitted over the network per unit time, generally measured in bits per second. It gives a measure of a network's capacity to carry data. In contrast, Latency refers to the delay that occurs when transmitting data over the network, representing the time taken for a data packet to travel from source to destination.<blockquote>In real-time communication, maintaining an optimal balance between throughput and latency is critical for ensuring efficient data transfer and minimizing network congestion.</blockquote>Deeply understanding the terms, in real-time communication, throughput and latency must exist in equilibrium. High throughput may lead to congestion and increase in latency, disrupting real-time data flow. On the other hand, minimal latency necessitates efficient and fast data transfer, requiring an optimized throughput. This balancing helps maintain efficient communication.<h2 id="What-is-Throughput?">What is Throughput?</h2>In essence, throughput refers to the total amount of data that the network is capable of transmitting in a specific time frame. It is often measured in bits per second (bps), outlining the volume of information efficiently moved from source to destination.In the realm of real-time communication, throughput is a critical factor. High-throughput rates mean that data is transferred quickly and efficiently, enhancing the seamlessness and effectiveness of voice and video calls.Beyond its importance in delivering clear and uninterrupted communication, harnessing the power of throughput also influences other key components. This includes network performance, bandwidth utilization, and overall user experience.<h2 id="What-is-Latency?">What is Latency?</h2>Latency is a critical element that determines the rapid response times during audio or video real-time communication. It refers to the delay that occurs as data packets travel from the source to the destination.Typically, a lower latency indicates a faster transmission. This is critical as high latency levels could lead to disjointed conversations, creating a poor user experience in real-time communication.Diving deep into latency, it's synonymous with the term 'transmission delay.' This encompasses the time it takes for a packet to travel from the sender to the receiver, including propagation, serialization, and queuing delays.<h2 id="The-Importance-of-Balancing-Throughput-and-Latency">The Importance of Balancing Throughput and Latency</h2>Having both high throughput and low latency is crucial to attain optimal real-time communication performance. Their balance ensures seamless and instant data transmission, thus enhancing user experience.Striving for harmony between throughput and latency is key to achieving superior communication. This synergy optimizes network efficiency and guarantees a smoother and more responsive interaction for users.<h2 id="What-are-the-differences-between-Throughput-and-Latency-in-Real-Time-Communication?">What are the differences between Throughput and Latency in Real-Time Communication?</h2>Real-time communication, such as audio and video calls, relies on efficient and reliable data transmission. When discussing the performance of real-time communication systems, two important metrics to consider are throughput and latency. While both are crucial in evaluating the quality of communication, they represent different aspects of the system's performance. Let's delve into the differences between throughput and latency in real-time communication.Throughput refers to the amount of data that can be transmitted over a network in a given amount of time. It is measured in terms of bits per second (bps) or its multiples such as kilobits per second (Kbps) or megabits per second (Mbps). In the context of real-time communication, throughput determines the capacity of the network to handle the data required for smooth and uninterrupted communication. It reflects the system's ability to transmit audio and video streams without significant delays or packet loss.On the other hand, latency refers to the delay experienced by data packets as they travel across a network. It is measured in milliseconds (ms) and represents the time taken for a packet to travel from the sender to the receiver. In real-time communication, latency directly affects the responsiveness and quality of the communication experience. Lower latency means less delay between sending a packet and receiving it, resulting in more real-time and synchronized communication.While throughput and latency are related, they are distinct metrics. Throughput focuses on the quantity of data transmitted, while latency emphasizes the speed and responsiveness of the communication. A high throughput indicates anetwork's capacity to handle large amounts of data, while low latency ensures quick and smooth data transmission.Optimizing Throughput and Latency in Real-Time Communication To optimize throughput and latency in real-time communication, consider the following strategies:1. Network Bandwidth: Ensure that your network has sufficient bandwidth to handle the data requirements of real-time communication. A higher bandwidth allows for higher throughput and reduces the chances of congestion and latency.2. Quality of Service (QoS): Implement QoS mechanisms to prioritize real-time communication traffic over other non-real-time traffic. This ensures that voice and video data packets are given higher priority, reducing latency and improving the overall quality of communication.3. Codec Selection: Choose efficient and optimized codecs for audio and video compression. These codecs reduce the size of data packets, resulting in higher throughput and lower latency. However, be mindful of the trade-off between compression and audio/video quality.4. Network Optimization: Optimize your network infrastructure and configuration to minimize latency. This includes reducing the number of network hops, using faster routing protocols, and implementing traffic shaping and prioritization techniques.5. Buffering and Packet Loss: Implement proper buffering mechanisms to handle temporary network congestion or fluctuations in throughput. Additionally, use error correction techniques to minimize packet loss, which can significantly impact the quality of real-time communication.6. Real-Time Protocol (RTP) Optimization: RTP is a standard protocol for real-time communication. Utilize RTP optimization techniques, such as header compression and congestioncontrol, to reduce latency and improve throughput. These techniques help in efficient transmission of audio and video streams, ensuring a smoother communication experience.<h2 id="How-does-Latency-impact-Throughput-in-Real-Time-Communication?">How does Latency impact Throughput in Real-Time Communication?</h2>Latency is a crucial aspect of real-time communication that directly affects throughput. Throughput refers to the rate at which data is transmitted and received within a given time frame. In the context of audio and video calls, latency is the delay between sending a packet of data and its reception at the destination.High latency can significantly impact throughput in real-time communication. Here's how:1. Increased Round-Trip Time (RTT): Latency causes a delay in the transmission of data packets, resulting in an increased RTT. RTT is the time it takes for a packet to travel from the sender to the receiver and back. Higher latency means a longer RTT, which reduces the number of packets that can be transmitted within a given time frame, ultimately reducing throughput.2. Delayed ACK and NACK: In real-time communication, acknowledgment (ACK) and negative acknowledgment (NACK) packets are crucial for ensuring reliable data transmission. Latency can cause delays in the reception of these packets, leading to retransmissions and reduced throughput.3. Packet Loss and Retransmissions: Latency can increase the likelihood of packet loss due to network congestion or other factors. When packets are lost, they need to be retransmitted, which adds to the overall latency and reduces throughput.4. <a href="https://trtc.io/blog/details/Jitter-and-Jitter-Buffer">Jitter Buffer</a> Management: To compensate for varying latency, real-time communication systems employ jitter buffers. Jitter refers to the inconsistency in packet arrival time caused by variable latency. Larger jitter requires larger buffer sizesto accommodate the delayed packets, which can result in increased latency and decreased throughput. Optimizing latency is essential to minimize jitter and maintain a consistent throughput in real-time communication.The Importance of Optimizing Throughput and LatencyOptimizing throughput and latency in real-time communication is crucial for delivering a seamless and high-quality user experience.<img src="https://trtc.io/media/019fbb82-e5ca-4d1f-8c0a-c5a1a715c003.png" alt="low-latency.png"><h2 id="Strategies-to-Optimize-Throughput-and-Latency-in-Real-Time-Communication">Strategies to Optimize Throughput and Latency in Real-Time Communication</h2>Real-Time Communication (RTC) is a critical aspect of modern audio and video call applications. To ensure a smooth user experience, it is important to reduce latency and improve throughput. Here are some techniques employed to achieve these goals:1. Network Optimization: One of the primary factors affecting latency and throughput is the network. Implementing techniques such as Quality of Service (QoS), traffic shaping, and network congestion control can help optimize the network for real-time communication. These techniques prioritize RTC traffic and ensure a stable and reliable connection.2. Codec Selection: The choice of audio and video codecs can significantly impact latency and throughput. It is crucial to select codecs that strike a balance between compression efficiency and computational complexity. Low-latency codecs, like Opus for audio and VP9 for video, can be preferred to reduce latency. Additionally, utilizing hardware acceleration for encoding and decoding can improve throughput.3. Packet Loss Recovery: Packet loss, a common issue in network communication, can severely impact RTC quality. Implementing techniques such as Forward Error Correction (FEC) and packet loss concealment can help recover lost packets or mitigate their impact. These techniques can improve perceived quality and reduce the need for retransmission, reducing latency.4. Buffering and Jitter Control: Buffering can help smooth out network fluctuations and reduce the impact of jitter. By carefully managing buffer sizes and implementing adaptive playout algorithms, it is possible to reduce latency caused by jitter. Additionally, incorporating jitter buffer management techniquescan help optimize the delivery of packets and improve overall throughput.5. Network Infrastructure: Optimizing the network infrastructure can have a significant impact on latency and throughput. Reducing the number of network hops, using faster routing protocols, and ensuring a stable and reliable connection can help minimize latency. Additionally, implementing traffic shaping and prioritization techniques can optimize the flow of data and improve overall throughput.6. Real-Time Protocol (RTP) Optimization: RTP is a standard protocol used for real-time communication. Employing RTP optimization techniques such as header compression and congestion control can help reduce latency and improve throughput. These techniques ensure efficient transmission of audio and video streams, resulting in a smoother communication experience.7. Monitoring and Analysis: Regularly monitoring and analyzing network performance can help identify potential bottlenecks and areas for improvement. By using tools like network analyzers and performance monitoring software, developers can gain insights into latency and throughput issues and make necessary adjustments to optimize real-time communication.<h2 id="Conclusion" style="margin-left:0pt;text-align:justify;">Conclusion</h2>Reducing latency and improving throughput are critical for ensuring a smooth and high-quality real-time communication experience. By implementing network optimization techniques, selecting efficient codecs, utilizing packet loss recovery mechanisms, managing buffering and jitter, optimizing the network infrastructure, and employing RTP optimization, developers can enhance the performance of audio and video call applications. Regular monitoring and analysis of network performance also play a crucial role in identifying and addressing latency and throughput issues. By prioritizing these strategies, audio and video call developers can provide users with seamless, real-time, and reliable communication experiences.<a href="https://trtc.io/">Tencent </a>RTC supplements the advantages of low latency by providing seamless and immersive experiences, whether you're gaming, attending virtual conferences, or streaming live events. With advanced technology and algorithms, Tencent RTC ensures minimal delays, allowing for responsive interactions and enhanced engagement.So, next time you're seeking an unparalleled audio and video experience, remember the importance of low latency and how Tencent RTC can elevate your real-time communication!If you have any questions or need assistance, our support team is always ready to help. Please feel free to&nbsp;<a href="https://trtc.io/contact?utm_source=website&amp;utm_medium=blog&amp;utm_term=trtc&amp;utm_campaign=blog&amp;_channel_track_key=EB6l06UE">Contact Us</a>&nbsp;or join us in&nbsp;<a href="https://discord.gg/hq7jW7zChW">Discord</a>&nbsp;.&nbsp;

Exploring the Trade-Offs: Throughput vs Latency in Real-Time Communication

Developer

Game Voice Evolution: RTC SDK, In-Game Voice, and Immersive Audio

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<img src="https://trtc.io/media/e82e606b-db16-4cdb-8fd9-149237b3694e.png" alt="Game Voice Evolution: RTC SDK, In-Game Voice, and Immersive Audio.png"><h2 id="1.-Unveiling-the-Evolution-of-Game-Voice" style="margin-left:0pt;">1. Unveiling the Evolution of Game Voice</h2>Social networking now is an essential aspect of gaming experiences. On one hand, gaming is inherently a social activity that offers a range of topics and scenarios for social interaction. On the other hand, social networking fulfills our human nature and improves the gaming experiences, which can significantly increase player retention.As an important feature of social networking, game voice has drawn growing attention from game developers and gained considerable popularity among players. There are myriads of game voice applications and solutions in the market, which provide a wide variety of features. Below is an overview of the evolution of game voice.Game voice tools have evolved with the development of the internet. The last 20+ years have witnessed huge leaps in game voice technology：<ul><li>from support for a single platform to cross-platform interoperability.</li><li>from one-to-one chat to interactive voice chat in a room with tens of thousands of online users.</li><li>from third-party voice communication SaaS tools to PaaS SDKs.</li><li>from monotonous voice chat to immersive voice experiences.</li></ul>Instead of following a chronological order or revisiting the evolution of their features, this article looks at the development of game voice tools from the perspective of game voice experiences.<img src="https://trtc.io/media/8f950a8f-8638-439f-9bc8-b64613712e74.webp" alt="game voice experiences in the game scene.webp">The most basic experience brought by game voice is that players communicate with each other through voice chat. Gaming creates a virtual world where the dialog between players is actually that between game characters. In tandem with the improvements in gameplay design and image quality, players have higher expectations of voice chat, and the voice experience similar to a conference call can no longer meet their demand. In response, a disruptive technical solution has emerged, which is called "Immersive Voice Solution".<img src="https://trtc.io/media/444811c1-a9ac-4200-8176-5f26bea5a66b.webp" alt="Immersive Voice Solution.webp">Game voice technology has gone through several stages, starting from the most basic voice chat to immersive voice experiences and beyond. As breakthroughs in sensors, computing power, audio algorithms, IoT, and other technologies are on the horizon, all-real voice will eventually become a reality, delivering the ultimate voice experience the metaverse demands.<h2 id="2.-Version-Iterations" style="margin-left:0pt;">2. Version Iterations</h2><h3 style="margin-left:0pt;">/ 2.1 Game voice v1.0: Third-party voice chat tools</h3>At this stage, players use third-party voice chat tools to communicate with each other in the process of gaming. Whether the game itself offers a voice communication feature or not, using third-party tools allows players to quickly create chat channels and communicate with each other through voice chat.<img src="https://trtc.io/media/df1dd867-f833-496b-bf12-b66097b07e06.png" alt="Third-party voice chat tools: skype, discord, Wechat, yy, zoom and so on.png"><h3 style="margin-left:0pt;">/ 2.2 Game voice v2.0: In-game voice</h3>In-game voice solutions mainly take the form of game developers connecting SDKs developed by voice communication PaaS providers. The basic APIs that come with the SDKs are used to implement various in-game voice scenarios:<ul><li>channel voice between teammates: teammates can have a voice chat at any position coordinates in the game.</li><li>range voice between different teams: players of different teams can hear each other only when their position coordinates in the game are within a specified range.</li><li>blocklist/allowlist.</li></ul><img src="https://trtc.io/media/fb014d31-9fd5-490a-afed-8dd7af43c3cf.png" alt="In-game voice chat with Tencent RTC's gaming voice solution.png">Unlike third-party voice chat tools, third-party voice solutions require game developers to integrate and design specific voice scenarios. For players, the establishment of voice channels, audio attributes, and the features will be automatically and adaptively adjusted based on the game progress and specific scenarios, thus taking the game voice experience to the next level. Some voice SDKs. Tencent RTC's gaming voice solution (GME, Game Multimedia Engine) provides a broad suite of APIs to meet other game voice needs than voice chat:<ul><li>voice messaging</li><li>speech recognition</li><li>accompaniment playback</li></ul>Based on these APIs, game developers can design more powerful social networking features for games.In-game voice solutions have preliminarily integrated voice and gaming business scenarios, yet basically at the level of feature integration. As a result, standalone voice SDKs can only give players a game voice experience similar to a conference call, although players don't have to establish a chat channel themselves, or run any resource-consuming third-party software.<h3 style="margin-left:0pt;">/ 2.3 Game voice v2.5: Upgraded version of in-game voice</h3>To further improve players' game voice experiences, voice SDKs like TRTC offer voice processing capabilities such as voice changing and virtual 3D sound field. With these features, players can change their voice in real time based on their selected voice type, which adds fun to gaming and allows a vast design space for game voice features.<img src="https://trtc.io/media/5ceb7dcb-db05-474f-a59f-cdaaa6c1feb1.jpeg" alt="3D audio effects in real time in vr. game.jpeg">In FPS, battle royale, or VR games, players need to accurately identify the positions and directions of other players, which is vital to player communication about tips/tactics and an immersive gaming experience. The 3D audio effects technology that comes with voice SDKs offers an effective solution. The positions perceived by players through voice correspond to the relative positions of their characters in the game. As the relative distance between the characters changes, the voice intensity will increase or decrease accordingly, which well simulates real-world dialog scenarios.<img src="https://trtc.io/media/bb7f3a23-db71-4ef0-a882-40997ad3943c.jpeg" alt="3D audio effects in game scene.jpeg">Through the 3D audio effects technology, voice processing and gaming scenarios are combined, which, however, are limited to position and distance information in gaming scenarios. For a truly immersive experience, voice processing should cover all aspects of gaming scenarios. A voice SDK is unlikely to provide a dedicated API for every potential factor; otherwise, the SDK would be extremely complicated and bulky, and that's not really necessary. To take the game voice experiences up a notch, we need a new solution, namely the "Immersive Game Voice Solution".<h3 style="margin-left:0pt;">/ 2.4 Game voice v3.0: Immersive voice</h3>An immersive voice solution means that players' voice effects are rendered in real time based entirely on the game process. All players' voices are processed through digital signal processing (DSP) algorithms, and then played back in the headphones to simulate voice communication in real-world settings. Voice chat processed in this way can deliver a more immersive game voice experience, allowing players to communicate in a natural way.<img src="https://trtc.io/media/550e2c40-4b59-4fe9-ab6d-e635587b11ed.png" alt="Wwise with GME.png">Then, how is an immersive voice solution implemented? As mentioned above, it is not advisable to have a single voice SDK packed with all sorts of APIs. Moreover, voice service providers are generally not experts in audio processing algorithms compared with specialist audio technology companies. Therefore, to develop an all-encompassing voice SDK is virtually unviable.In view of this, a combination approach will work best, just as with the Wwise + GME solution. GME is dedicated to end-to-end real-time voice communication, and the Wwise interactive audio engine is adopted by many game developers as a tool for game sound design. The Wwise plugin acts as a bridge for data interactions between GME and the Wwise engine, and GME voice streams are seamlessly connected to the Wwise audio pipeline, so Wwise's rich sound effects processing and control features can be used in voice chat. Such a design makes it possible to deliver an immersive game voice experience.<figure class="image"><img src="https://trtc.io/media/3952e728-9ee0-4196-8111-40fc4b536c89.jpeg" alt="immersive game voice in real time in the game scene.jpeg"></figure>As an interactive audio authoring tool, Wwise is generally used to create high-quality audio content for games, and GME complements Wwise in the field of game voice. Now sound engineers can also use Wwise to create immersive and interesting voice features, opening up new gameplay possibilities.<h3 style="margin-left:0pt;">/ 2.5 Game voice v4.0: All-real voice</h3>Immersive voice, however, is definitely not the acme of game voice experiences – all-real voice takes it further.With the advances in AR, VR, and MR technologies, the metaverse has become a hot topic. Many technology giants are expanding into the metaverse, which is considered the next biggest opportunity in the realm of the internet in the coming decade. The metaverse refers to a parallel virtual world that is both independent of and interconnected with the real world, where people can interact, work, and do much more realistically.To make virtual worlds more lifelike, software and hardware technologies need to be integrated to simulate human senses. As voice communication is an important form of human interaction, metaverse scenarios have higher requirements for voice, that is, all-real voice. Currently, the metaverse is still more of a concept than reality, and we'll see what the future holds.Gaming is inherently a social activity in the internet age. Although voice chat is not a core feature for most game genres, it makes gaming more enjoyable and thus increases player retention. Therefore, it has become a common feature of online games.<img src="https://trtc.io/media/b3d8afa0-b6cd-41fb-a1ca-b9ac3d35e0c2.jpeg" alt="all-real voice in the online social games like WEREWOLF.jpeg">Game voice technology has evolved in response to players' growing demand for better experiences and gameplay. As players have higher expectations of gaming experiences, voice is bound to hold greater weight in gaming.

Understanding TRTC and Its Implementation

What is TRTC?

Capabilities of TRTC