With WebRTC, you can add real-time communication capabilities to your application that works on top of an open standard. It supports video, voice, and generic data to be sent between peers, allowing developers to build powerful voice- and video-communication solutions.
The WebRTC standard covers, on a high level, two different technologies: media capture devices and peer-to-peer connectivity. Media capture devices includes video cameras and microphones, but also screen capturing "devices".
Peer connections is the part of the WebRTC specifications that deals with connecting two applications on different computers to communicate using a peer-to-peer protocol. The communication between peers can be video, audio or arbitrary binary data (for clients supporting the RTCDataChannel API). In order to discover how two peers can connect, both clients need to provide an ICE Server ...
When developing for the web, the WebRTC standard provides APIs for accessing cameras and microphones connected to the computer or smartphone. These devices are commonly referred to as Media Devices and can be accessed with JavaScript through the navigator.mediaDevices object, which implements the MediaDevices interface. From this object we can enumerate all connected devices, listen for device ...
Here you'll find the different support options for developing WebRTC-based applications, including links to API references, external tutorials, sample code, testing guidelines, and the current state of support for different browsers and platforms.
- Introduction In this codelab, you'll learn how to build a simple video chat application using the WebRTC API in your browser and Cloud Firestore for signaling. The application is called FirebaseRTC and works as a simple example that will teach you the basics of building WebRTC enabled applications. Note: Another option for signaling could be Firebase Cloud Messaging. However, that is ...