Some people say: 2G to see the text, 3G look at pictures, 4G watch videos, then for 5G era has opened the prelude to it? With the rise of short videos, online classroom, interactive live audio and video applications such as, how to adapt differentiated network environment, to provide users with a smoother high-definition real-time audio and video services to become the focus of attention. The current audio and video technology still can not avoid all kinds of Caton packet loss, network coverage issues, can not provide chat, storage, transcoding, AI processing, full-stack solutions.
In this context, UCloud recently introduced a new technology based on real-time audio and video products RTC URTC, relying UCloud computing power and network coverage, can provide users with the nearest access (500 + accelerate global node), high quality ( the availability of up to 99.99%), anti-weak network (30% of video packet loss, packet loss is still 70% of normal audio communications), network audio and video communications latency (delay national average 70ms, the international average delay of no more than 300ms), and and supports push people live.
Global real-time transmission, weak anti-low latency network
Based UCloud in 30 global deployment of the available area, 28 lanes, 500 + accelerated node, URTC can provide audio and video services, global coverage, the domestic average delay of 70ms, the international average delay of no more than 300ms. The nearest access point by global access, since the inquiry HTTPDNS scheduling algorithm, a retransmission packet loss, to realize high quality communication network is weak, a 30% loss of video, audio packet loss of 70% can normally communicate. 70% loss under URTC after the network transmission of audio effects play out as follows, by virtue of its ability to resist loss is still a more complete presentation of the human voice:
URTC underlying network covering all BGP routes, using the nearest access edge node, the data packet transmitted through the AES encryption, end to end link quality detection, multiple access lines to ensure their availability and other disaster recovery of 99.99%. For transmission network, UCloud made the following optimization.
Anti-Jitter: URTC strategy through intelligent dynamic buffer, the buffer size dynamically adjusted according to changes in the network against network jitter, network 800ms maximum support anti-shake.
Anti network packet loss: URTC by smart NACK + FEC + ARQ retransmission policies against the loss, by dynamically adjusting the proportion of retransmissions and the redundant data, and achieve network delay between call quality to achieve a good balance. Meanwhile URTC will be important data classification, to ensure that important data first, stable transmission, while achieving fairness and stability of the transmission.
Avoid network congestion: using linear regression mode, delayed by a change in slope and predict changes in network congestion occurs, URTC able to do earlier discovery of network congestion, network packet loss avoided earlier, much earlier to adapt to network changes.
In-one, easy to use
Under the traditional way, to self-study audio and video functions on the APP application, developers only need to have a basic audio and video encoding and decoding technology, but also need to deal with complex network architecture issues. From the deployment server device, developed by the late operation and maintenance are time-consuming, and difficult to guarantee the quality of services, technical threshold of their own have become increasingly demanding, which for many start-up businesses is undoubtedly a huge challenge .
URTC SDK package in the form of a key with integrated audio and video capture, encoding, transmitting, decoding, rendering, before and after treatment (e.g., facial, filters, echo cancellation, noise suppression) function, and can be extended service, the client application all such as interactive whiteboards, transcoding, Francis, recording and so on. Because of its SDK interfaces close to the mainstream RTC two fast mode SDK, developers need only minimal code can be achieved fast access, simple and approachable.
Meanwhile URTC support Android, iOS, Mac OS, Windows, Electron, WEB multi-platform access, access to meet the needs of different clients.
Push million people live support
URTC RTC real time using the cluster, the cluster RTC live, interactive achieve audio and video even wheat can simultaneously push people live, concrete works as follows.
Business Services
URTC in the realization of flow control including communications services, real-time interactive cluster (less than 15 rooms), low-latency live cluster (people crowd) three kinds of services cluster.
Flow control flow communication service provider status, throttle position reporting, and control flow subscriptions and publishing; real-time interaction is mainly responsible for a cluster of 15 people within a two-way real-time interactive services; low-latency cluster provides live one-way low-latency million people live viewing . Between media services as well as recording and transcoding clusters transmitted through self-development private UDP protocol.
底层网络
依托于 UCloud 全球加速网络 Rome，通过专线服务，优化跨区域、跨洋通信问题，提供更低的延时，更优质的传输质量和通话体验。
（图：Rome 公网加速）
服务灾备
基于 Rome 的多线多云灾备能力，同时在单点服务故障时实现故障自动转移，提供更稳定的服务。
打破地域限制，助力在线教育
随着互联网技术的高速发展，不受地域和时间限制的在线教育逐渐成为很多人学习、进修的选择。为了保障不同地域学生的在线学习体验，提升在线课堂的品质，在线教育对音视频应用服务的质量也提出了更高的要求。
URTC 高质量、抗弱网、低延迟的优异性能，能满足在线教育对于音视频服务的基本需求。通过 UCloud 自研和优化的 webrtc 架构，URTC 具有一对一、一对多、多对多的音视频通话功能。既能满足 1V1、小班课、双师课堂，也能满足视频互动大班课、万人直播课。
互动演示文稿、互动白板，为师生学习、讨论提供交流的平台，且老师、学生可以在各自的白板内实时批注交流。白板内容支持录制、存储、回放，满足课堂教学、实时讨论、课后复习的需要。IM 聊天室支持老师设置全员禁言，让学生们集中注意听讲也可以发布讨论话题，让学生们畅所欲言。
（图：互动白板界面）
除了在线教育外，URTC 还可以广泛应用于智能家居、远程医疗、视频会议、娱乐直播、安防监控等场景。在实时音视频领域，UCloud 将围绕用户的场景和痛点不断丰富 URTC 产品特性，提供低延时、高可靠、安全稳定的音视频解决方案。为了让每个视频的通信质量、实时状态能够实时监测，也便于快速定位排障，URTC 近期将支持可视化运维面板，敬请期待。
免费体验 URTC
为了帮助开发者更直观体验 URTC 在弱网环境下的抗丢包效果，这里给大家介绍一个网损测试方法，具体操作步骤如下：
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开源弱网模拟软件（网损软件）Clumsy 下载：https://github.com/jagt/clumsy/releases
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URTC demo download: https://github.com/ucloud/urtc-win-demo
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Compile URTC demo: need Visual Studio 2015 and Windows10 SDK
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Run two demo client: a subscription and a release
- Open and set Loss software below:
Set net loss of direction (sending representatives to upload dropping receiving on behalf of download packet loss)
Set loss: Confirm drop option is selected, the packet loss function is turned on again, set behind the loss of 70% (value 0.0 - 100.0)
Click Start, start net loss settings, software starts running:
- Test results
Relying on the human ear to hear;
implemented UCloudRtcAudioFrameCallback, pcm data acquired by remote audio onRemoteMixAudioFrame interface, and then compare the waveform.
Currently URTC within 10,000 minutes per month free, quota beyond the range shown in Figure billing, please click http://ucloudtml.mikecrm.com/C3aLi69 apply for trial!
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