WebRTC is an open source technology that supports direct transmission of audio, video and data between browsers, and its core feature is decentralization. 1. It establishes a point-to-point connection by obtaining local media streams, creating an RTCPeerConnection object, exchanging SDP protocol descriptions, and ICE candidate addresses. 2. DataChannel can be used to transmit any type of data, with the advantage of low latency. 3. In the face of NAT and firewall problems, WebRTC uses STUN to obtain public network addresses and relay data through the TURN server if necessary. It is recommended to configure STUN and TURN services for connection stability during deployment.
WebRTC is a key technology for implementing point-to-point communication in the browser. It allows audio and video calls, file transfer and other functions to be performed directly between users without having to pass through the server. If you are considering integrating real-time communications in web applications, WebRTC is almost the first choice.

What is WebRTC?
WebRTC (Web Real-Time Communication) is an open source project that supports direct transmission of audio, video and data between browsers. Its core feature is "decentralization", which means that two browsers can establish a point-to-point connection and directly exchange information without uploading the data to the server and forwarding it.
The technology is developed by Google and is supported by mainstream browsers such as Chrome, Firefox, Edge and Safari. You can use it to build video conferencing systems, online education platforms, remote assistance tools, etc.

How to establish a point-to-point connection?
To establish a connection using WebRTC, you usually need to go through the following steps:
- Get local media stream : For example, obtain camera and microphone permissions through
navigator.mediaDevices.getUserMedia()
. - Create an RTCPeerConnection object : This is the core object of WebRTC that manages point-to-point connections.
- Generate and exchange SDP protocol description : SDP (Session Description Protocol) contains information such as media format, codec, network address, etc.
- Collect and exchange ICE candidate addresses : The ICE (Interactive Connectivity Establishment) protocol helps find the best communication path between two devices.
For example, when you click the "Start Video Chat" button in your browser, your browser will try to establish a connection with the other browser. During this process, both parties will generate offer/answer and exchange ICE candidate information.

It should be noted that although the communication itself is point-to-point, the initial connection often requires a "signaling server" to help exchange metadata (such as offer, answer, and ICE candidates). This part is not done through WebRTC, but is built by yourself or implemented with the help of third-party services.
Use of DataChannel
In addition to audio and video communication, WebRTC also provides a very practical function: DataChannel. It allows you to transfer any type of data in a point-to-point connection, such as text, binary files, JSON messages, etc.
Compared to WebSocket or HTTP requests, DataChannel has advantages in lower latency and fewer intermediate links. For example, you can send real-time text messages while a video call, or synchronize changes to content when two people collaborate on editing documents.
However, the following points should be paid attention to:
- DataChannel is unreliable by default (similar to UDP), but can also be set to reliable mode (similar to TCP).
- If you need to send a large amount of data, it is recommended to process it in chunks to avoid blocking the connection.
- Compatibility issues should be considered, especially on mobile or older browsers.
How to solve NAT and firewall?
Most users’ devices are on a private network, and there may be routers or firewalls behind them, which can make point-to-point connections difficult. WebRTC uses STUN and TURN servers to bypass these restrictions.
- The purpose of the STUN server is to help the device discover its own public IP address and port so that other devices can access it.
- The TURN server acts as a "relay" when it is impossible to establish a direct connection, forwarding data from one end to the other.
Simply put, if both devices are under the public network, they can communicate directly; if one of them is in the intranet, STUN needs to assist in digging holes; if neither can get out, they can only rely on TURN forwarding.
When deploying WebRTC applications, it is best to configure both STUN and TURN servers. You can use free public STUN services (such as stun:stun.l.google.com:19302
provided by Google), but for production environments, it is recommended to build or purchase commercial services yourself.
Basically that's it. WebRTC looks complicated, but in fact, as long as you understand the basic process, you can get started quickly. Many details such as the collection mechanism of ICE candidates, the structure of SDP, how to deal with disconnection and reconnection are all advanced topics. For those who are just getting started, it is enough to first master the basic processes of connection establishment and data transmission.
The above is the detailed content of WebRTC for Peer-to-Peer Communication in Browsers. For more information, please follow other related articles on the PHP Chinese website!

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