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Web Real-Time Communication (WebRTC) is an open-source project and specification that enables the exchange of multimedia data, such as voice, video, and data, between browsers and devices in real-time. This gives users the ability to communicate from their web browser without the need for complex plugins or additional hardware. Today WebRTC is available for Chrome, Firefox, Safari, Edge, Android, and iOS and is a widely popular video calling tool.
Let’s see how exactly does it work.
On the client-side
On the WebRTC server-side A media server is not required for data exchange between two or more participants. Although it is recommended to use it for better quality and fast data transfer. The video server will receive media traffic from various sources, convert it and send it to users who use WebRTC as a terminal. The WebRTC server will also receive media traffic from WebRTC and pass it on to conference participants who use desktop or mobile devices, if available.
What are the advantages while using the WebRTC solution?
It is free WebRTC — first introduced by Google in 2011, Google’s goal was to offer a standard real-time media engine that would be free and available in all available browsers.
Secure voice and video transmission WebRTC always includes voice and video encryption. Secure RTP(SRTP) is used to encrypt and authenticate voice and video. This is especially important in Wi-Fi networks. This prevents eavesdropping and recording of voice and video.
No matter what platform you use and what device you have Any WebRTC-enabled browser with any operating system can route the browser to create a real-time voice or video connection to another WebRTC device or WebRTC media server. The browser operating system does not matter. This is achieved by implementing standard W3C APIs and IETF protocols.
Improved voice and video quality WebRTC uses the Opus audio codec, which provides a high-quality voice. Opus codec based on Skype SILK technology. The VP8 codec is used for video. These options provide compatibility and eliminate the need to download codecs that may contain malicious code.
Reliable session establishment WebRTC supports reliable session establishment. WebRTC supports NAT. WebRTC algorithms do all to avoid relaying, thereby reducing delay and improving quality.
Adapts to network conditions WebRTC supports the connection of multiple media types and endpoints. This ensures efficient use of bandwidth, providing the best voice and video communications. API and signaling can adapt size and format for each endpoint separately.
However, besides the advantages, the WebRTC system also has one disadvantage. If you need to create an audio or video connection for several participants, it is better to use a media server, but not required. This is due to the fact that no browser can connect multiple signals together.
WebRTC is a relatively flexible library and allows wedging into the middle of the process, for example for additional encryption, or for additional processing such as echo cancellation or voice change algorithm. Our development team is currently working on an algorithm for additional encryption and voice change on one of our projects. You also need to remember that for WebRTC it is required to create your own server for communication. It is the signaling server that connects two or more participants.
WebRTC is already the most widespread and supported real-time communication solution. It has a lot of advantages and creates really high-quality communication. For this reason, it is worth paying attention to this system, since it is really worthy of attention, and by choosing it, you get excellent communication quality and reliable protection.
Reach out to the Clover Dynamics team and we would be pleased to share our experience working with WebRTC and help you with the questions in case you have them.
