Anyrtc implements application scenarios based on webrtc

Preface

WebRTC, The name comes from web instant messaging （ English ：Web Real-Time Communication） Abbreviation , Is a web browser to support real-time voice or video dialogue API. It's on 2011 year 6 month 1 Japan open source and in Google、Mozilla、Opera With the support of W3C Recommended standards .anyRTC stay WebRTC On the basis of , With years of development experience and in combination with the actual situation , Developed be based on WebRTC Cloud platform for audio and video communication , Provide interactive teaching 、 Lianmai live 、 Videoconferencing 、 Command and dispatch and other audio and video cross platform solutions .

WebRTC history

• 2010 year 5 month ,Google With 6820 Million dollars for VoIP Software developers Global IP Solutions Of GIPS engine , And changed it to “WebRTC”.WebRTC Use GIPS engine , Realized video conference based on Web page , And support 722,PCM,ILBC,ISAC Equal coding , Also use Google's own VP8 Video decoding ; Support at the same time RTP/SRTP Transmission, etc .

• 2012 year 1 month , Google has integrated this software into Chrome Browser . meanwhile FreeSWITCH The project claims to support iSAC audio codec.

WebRTC The core API

WebRTC Native APIs The document is based on WebRTC It's written out of specifications , these API Can be divided into Network Stream API、 RTCPeerConnection、Peer-to-peer Data API Three types of ：

Network Stream API

• MediaStream：MediaStream Used to represent a media stream .
• MediaStreamTrack Represent a media source in a browser .

RTCPeerConnection

• RTCPeerConnection： One RTCPeerConnection Object allows users to communicate directly between two browsers .
• RTCIceCandidate： It means a ICE Candidates for the agreement .
• RTCIceServer： It means a ICE Server.

Peer-to-peer Data API

• DataChannel： Data channel （DataChannel） Interface represents a bidirectional data channel between two nodes .

WebRTC Yes IT The benefits of infrastructure

TCO WebRTC It works with the help of a web browser , So you don't need a separate audio device , Or connected to the private exchange IP/ Analog telephone and software telephone . Traditionally , Used to use a physical phone , But maintaining them costs a lot . and , Money associated with maintaining physical phones has been added to more and more infrastructure budgets . When WebRTC After the invention , All the above expenses have been reduced . When you use WebRTC When , No installation at all , maintain , Or support work , The audio channel is just you UI Part of .WebRTC It also allows your salesperson to log in remotely .

ad-hoc Expand   Usually , The infrastructure team needs a whole month of hard work and a lot of expenses to launch a contact center . But with WebRTC, It only takes less than a week for the same contact center to go online . This will also help organizations and start-ups deploy staff in various regions , And there is no need to invest time and money in audio equipment .

Network and bandwidth usage and security  WebRTC Support multiple media formats and terminals , This can help provide the best audio and video communication between your staff and the server .WebRTC Use Opus Audio codec , Experts say this is based on Skype SILK Codec technology codec Is the best audio decoding .Opus The code supports all bandwidth , Narrowband , Broadband , Ultra wide band , And full band . It means WebRTC Better than the human ear to distinguish sound .

WebRTC The overall architecture of the audio engine

• On the internal implementation , Audio engine VoiceEngineImpl Through a series of objects to achieve audio processing , Include VoEAudioProcessingImpl、VoECodecImpl、VoENetworkImpl wait , Each object is responsible for a specific function , for example VoEAudioProcessingImpl Responsible for calling the underlying AudioProcessing The module preprocesses the audio data . In these functional objects , More important are VoEBaseImpl、SharedData and Channel. among VoEBaseImpl It's connecting audio devices AudioDevice And the audio engine VoiceEngineImpl The link between , It's an important station in the audio data pipeline ;SharedData It's an aggregate class , Holding a range of important objects ;Channel Represents a channel of audio data , Responsible for most of the important operation of the data , Including audio data preprocessing 、 codec 、 Send and receive 、 post-processing 、 Mix, etc .

• In terms of functional dependency ,VoiceEngineImpl It relies on five important underlying functional modules ： Audio data acquisition and playback AudioDeviceModule 、 Audio data preprocessing AudioProcessing、 Audio data codec AudioCodingModule、 Receiver audio data buffer NetEq、 Receiver mix AudioConferenceMixer. In addition, audio data encoding and decoding also relies on a series of audio encoding and decoding, such as G711、G722、Opus wait . At the sending end , The audio data is generated by AudioDevice Collect and get , after AudioProcessing After pretreatment , arrive AudioCodingModule Encoding , Then from RTPRTCP The module is sent to the network . At the receiving end , The audio data goes through RTPRTCP The module receives and arrives at AudioCodingModule, Stored in NetEq Jitter control and error elimination are carried out in , Then decode . The decoded data goes through AudioConferenceMixer Mix , Finally sent to AudioDeviceModule Play it .

• From the whole WebRTC In terms of frame structure , The audio engine and the video engine are both at the bottom , Responsible for the collection of audio and video data 、 codec 、 Rendering and playing . The upper layer of audio and video engine is multimedia engine WebRtcMediaEngine2, It's about the underlying audio and video engine VideoEngine A further high-level abstraction of , from WebRtcVoiceEngine Yes VoiceEngine encapsulate ,WebRtcVideoEngine2 Yes VideoEngine encapsulate .

anyRTC Audio and video communication cloud platform SDK Which use scenarios use WebRTC？

1、 Online education

ArRoomEngine SDK, Based on Google WebRTC Open standards build , Ultra low delay 、 Full terminal coverage , Can meet all kinds of needs , Support large class 、 A small class , Provide whiteboard function , Free trial .

2、 Videoconferencing

RTMeetEngine SDK, High definition smooth audio and video 、 High security 、 Full platform operation 、 Rich conference management functions , Support video 、 Voice multiplayer Conference , For meetings 、 train 、 Interactive multi person mobile Conference .

3、 Command and dispatch

RTMaxEngine SDK, High definition smooth audio and video 、 Ultra low delay 、 Strong command 、 Resource support and other comprehensive and coordinated video communication command platform , Realize real-time video communication between on-site emergency and rear emergency command center 、 Synchronous transmission 、 Precise scheduling 、 Efficient collaboration at all levels . For security monitoring 、 Smart home .

4、 Interactive Lianmai

RTMPCHybirdEngine SDK be based on RTMP and RTC Online video live broadcast of hybrid engine .iOS live broadcast （ Network adaptive rate RTMP publisher）、 On demand player （ The player has been professionally optimized , It can be switched on in seconds RTMP Player）、 be based on RTMP and RTC Hybrid engine's video link and MAC interaction （ Up to four ways of Lianmai interaction are supported ）, For live games 、 Beauty show .

5、 voice call

RTCP2PEngine SDK, Support video 、 voice 、 Priority video and other call modes , For Internet telephony 、 Activities 、 Education and other call scenarios .

6、 Live in real time

RTCPEngine SDK, Realize fast real-time live broadcast , comparison RTMPC Faster , Ultra low delay 、 The minimalist API Interface 、 Ultra fast access . For online doll machines 、 Smart hardware 、 Online medicine 、 Video recruitment 、 Dating and other scenes .