WebRTC full form of web real-time communication, it is a web-based real-time communication protocol, and RTC enables web apps to stream audio/video media over the internet. RTC over the web is called webRTC.
WebRTC can perform DATA SHARING and TELECONFERENCING in a peer-to-peer fashion. WebRTC works in tandem and synchronized way with OSI PROTOCOL STACKS and APIS.
WebRTC is integrated with MEDIA APPs and STREAMING APIs, they leverage webRTC to transmit media and stream media files.
Use case of webRTC includes:
- Audio conferencing and video conferencing
- Remote access via screen sharing
- File exchange
- Few other business and end-use cases involving media transfer and streaming.
The connection is webRTC is managed via interface, type, and dictionary all of them work in tandem to establish, maintain, and mange webRTC connections. Ok
NOTE: The peer-to-peer connection between applications is enabled by ICE( interactive connectivity protocol) and the session is also managed by ICE.
The peer-to-peer nature of WebRTC connects two devices via RTC connection interface, once the Peer-to-peer connection is established then DATA CHANNELS ARE ADDED to CARRY OUT MEDIA STREAM AND DATA TRANSFER OPERATION
sender –> PEER TO PEER —> RECEIVER (Media Connection)
sender –> PEER TO PEER —> RECEIVER (Data Connection)
Via PEER TO PEER connection, the media and data transfer will take place.
Key Points to consider:
- RTC will have peer to peer connection
- When connection is established a DATA CHANNEL is added.
- Events will be associated with each data channel.
- The RTC session will be established peer to peer and data transfer will be done via data/media stream.
The RTC sender will be responsible for encoding and data transmission. The RTC receiver will be responsible for data decryption and decoding. RTC events will be tracked and logged.
Some RTC events include:
Buffer amount -> The amount of data buffer
Closing -> The Data channel is Closing.
Close -> The data channel is close
Error -> Will display error during transmission and reception of data and other errors occurred in the data channel,
Message -> Message received on Data channel
Open -> The Data channel has been opened or re-opened.
Track -> tracking of events
State change -> the RTC transport state is changed.
The dictionary, interface, and events are documented in the webRTC website, it is beneficial to go through the documentation before integrating webRTC.
WebRTC provides REAL TIME COMMUNICATION capabilities via web browsers with the help of APIs. The bandwidth consumption is higher for webRTC due to stream data having higher data density.
WebRTC was initially released in 2011, REPO: webrtc.googlesource.com /webrtc.com. WebRTc has been published by IETF and W3C. Web RTC (codec) is supported by major browsers.
RTC is leveraged by the following technologies:
- Web app
- mobile app
- browser
- IoT devices
- Video Calling
- Audio Calling
NOTE: For user identity management and user security implementation leverage, visit RTC documentation to learn more about it. The encoded transform, interface, and events are leveraged to process video frames and audio frames via the RTC transform stream event.
The transform pipes are inserted in the receiver stream via using RTC encoded transform property called (RTCRtoReceiver.transform), and the transform pipe is inserted in the sender pipeline via using RTC encoded transform property called (RTCRtosender.transform). Transformer pipes are inserted into both the SENDER and RECEIVER ends.
NOTE: NAT firewalls can prevent (P2P RTC communication between two devices.
Common use cases: WebRTC is integrated into ZOOM CALL, MS TENANT, MS TEAM, GOOGLE MAPS, AND MANY POPULAR CALLING, MESSAGING, AND STREAMING SERVICES.
Diagrams
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