In recent years, online education, werewolf killing, online doll catching, online KTV and other multi-person video interaction modes continue to emerge, real-time audio and video communication limelight is strong, real-time audio and video technology WebRTC has therefore received widespread attention. Data shows that the global Network Real-time Communication (WebRTC) market will grow at a COMPOUND annual growth rate of 34.37% from 2017 to 2021. (Data source: Technavio “Global Web Real-time Communication (WebRTC) Market, 2017-2021” report)
The New Age of Real-time interaction: Progressive WebRTC
Since the release of the draft WebRTC 1.0 standard in November 2017, it has been widely supported by more and more vendors. WebRTC official data shows that WebRTC has been used by more than 1300 companies and projects, supported by more than 80% of browsers, WebRTC is used in a variety of well-known applications: WhatsApp, Facebook Manager, Appear. In and TokBox.
WebRTC, full name for Web Real Time Communication, is an open source project promoted by Google. Its initial goal is to enable real-time audio and video calls without plug-ins for browsers. To build a platform based on WebRTC to realize the interaction between the MAC and the audio and video communication can greatly reduce the access threshold and development cost:
80% of browsers support WebRTC
Before Google made WebRTC open source, real-time communication between browsers was a difficult task. Developers can now implement audio and video communication on the Web side using simple HTML tags and JavaScript apis without having to pay attention to the details of the audio and video engine implementation. Currently, Chrome, Safari, Firefox, Opera and other major browsers all support WebRTC, as a standard of H5, will be supported by more browsers in the future.
2. All-platform interconnection can be realized
Imagine sharing a Web link to a friend on your phone and expecting them to open their browser and talk to you in real time, so cross-platform connectivity is an important experience. Google has opened the bottom C++ interface, based on which developers can develop applications on iOS, Android, Mac, Windows and other platforms, realizing the interconnection of applications on the whole platform.
3, WebRTC has a strong ability to hole
WebRTC technology includes the key NAT and firewall penetration technologies using STUN, ICE, TURN, RTP-over TCP, and supports proxy to ensure that P2P clients can share file information, processor computing power, storage space and other resources through direct communication.
4, safe and reliable, stable quality
WebRTC provides reliable video and audio data encryption to ensure the security of audio and video data transmission on the public network without information theft. For data sensitive enterprises, as long as they cooperate with private storage, data can be safely transmitted and stored. At the same time, Google’s strong technical endorsement and support can ensure the rapid update and iteration of technology, such as sound noise reduction, volume gain, echo cancellation, which can greatly optimize user experience and ensure stable quality.
WebRTC Distance from industry grade applications
In this way, it doesn’t seem difficult to implement real-time audio and video calls based on WebRTC. However, there is still quite a long way to go from WebRTC to an industry-grade application:
1. The whole platform supports a large amount of development
WebRTC only provides Web access, but does not provide easy-to-use SDK for Android, iOS, and Windows clients. This is a great challenge to the ability and experience of developers, and for companies that lack the accumulation of audio and video technology, it will undoubtedly increase the development cost and prolong the time to launch.
2, P2P connection mode of natural drawbacks
The biggest problem of P2P connection is that the success rate of connection is not high. The official data of Foreign countries given by Google is 86%. P2P models are even weaker in a multi-person interaction scenario. When multiple users interact with each other, they need to distribute their streams to multiple users at the same time, which requires high upstream bandwidth. Therefore, the current network environment cannot support multi-user sessions.
3. Single scene support
P2P uses end-to-end direct connection without a server, so the scenarios supported are limited to simple 1-to-1 communication. There is no way to process audio and video streams on the server, such as confluence, bypass live streaming, watermarking, transcoding and so on.
Therefore, for companies with less audio and video technology reserves, choosing a reliable one-stop solution provider can greatly reduce the development cost and shorten the product launch time.
Qiniu real-time audio and video cloud is a real-time audio and video solution based on the standard WebRTC, while optimizing the above problems of WebRTC. Qiniu Real-time audio and video cloud provides real-time audio and video SDK of the whole platform. Through self-developed RTC accelerated network and strong cloud capability support, qiniu real-time audio and video cloud provides customers with cross-platform, high-quality and customizable one-stop solutions.
Seven cows based on WebRTC Provide industry level solutions
1. P2S connection model to realize 1-pair multi-interaction
The connection model of Qiniu RTN is called Peer to Sever (P2S), which adopts SFU topology scheme, namely forwarding model scheme. We have added A server between Client A and Client B. Signal Sever is responsible for end-to-end signaling transmission, and Media Sever is responsible for establishing data channel. Based on Media Sever, we can complete the forwarding amount. Compared with THE P2P model of WebRTC, P2S can not only avoid the problem of small uplink bandwidth, but also realize real-time audio and video interaction of multiple people.
P2P model
P2S model
2, RTC accelerated network, high reliability and low latency
Seven cows RTC since research to accelerate the network as a real-time transmission network, compared to the general speed network, with global node and multiple suppliers line support, adopt the grind edge acceleration program, fully support to accelerate signaling and data transmission, intelligent distributed streaming media server and accelerate the lines, to ensure that the entire network connectivity and low latency.
RTC accelerated network
Bypass live cloud storage, support on-demand playback
“By-pass live” and “on-demand playback” are common derivative scenarios of real-time interaction. For example, if several people have a real-time meeting online and tens of thousands of people want to watch the meeting live, it is necessary to push the flow live at the server side, namely “bypass live”. Bypass live can maximize the dissemination of interactive information at a lower cost:
For example, in finance, government affairs, customer service and other application scenarios, interactive content also needs to be stored for record and reference, which can be sliced and stored in real-time in Qiniu Cloud. For scenes such as education and show interaction, the audio and video content can be stored for secondary transmission through CDN in the later stage.
4, support the convergence of the server to deal with all kinds of terminal models
From the perspective of distribution cost and storage cost, it is usually necessary to merge multiple interactive pictures into one picture before bypass broadcast and cloud storage, which is called “merging”. Merging can be done on the client side or the server side. Although merging on the client side is a relatively simple way of merging, there are problems such as hot phone and unstable merging. Compared with the client merge, the server merge has lower requirements on the terminal, with smoother picture and clearer picture quality, which greatly reduces the heat and lag of the mobile phone. Qiniu adopts GPU confluence scheme at the server side, which can greatly improve the efficiency, stability and picture quality of confluence.
Seven cattle RTN It supports seamless access in various scenarios
Qiniuyun Real-time audio and video Cloud (RTN) has been officially launched, showing strong availability in social, education, medical, financial, conference, government and people’s livelihood, and can cope with the needs of various scenarios.
· Social field: support anchors to connect with each other or anchors to the audience, provide functions such as beauty, filter, big eyes, thin face, etc., to meet the interesting interaction.
· Education: flexible support for one-to-one education, interactive small class, large class of ten thousand people, interoperability of all platforms, support for screen sharing, to meet the education needs in multiple scenarios.
· Interactive meeting: support small team online communication and large online meeting, can easily make an application similar to WebEx.
· Medical field: support remote multi-party video consultation, break through regional limitations of medical resources and system platform, improve regional flexibility of doctor-patient time, and reduce diagnostic costs.
· Financial field: support to initiate video call requests to technical support personnel, technical support personnel can guide users to operate through video, quickly locate and solve problems, and improve service quality and product reputation.
· Government affairs and people’s livelihood: support online court hearings, remote alarm, remote emergency command and other government audio and video calls, so as to facilitate two-way audio and video calls between municipal personnel and citizens through multi-terminal browsers.
In the above scenario, Qiniu RTN provides a one-stop solution that can save the video and meet the compliance requirements that need to be left behind; It can carry out secondary processing of audio and video, greatly improving the efficiency of external communication.
Nowadays, with the rapid development of the Internet and the increasing maturity of WebRTC technology, real-time audio and video communication will have more application scenarios and greater development space. Qiniu RTN will continue to work deeply in the field of real-time audio and video, directly hit the pain points of real-time audio and video development, and provide customers with more innovative technologies and quality solutions for reference.
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