WebRTC provides an easy and simple API which allows implementing video calling capabilities for both web and mobile applications and browsers including Firefox, Chrome, and Opera along with all the important mobile platforms and iOS supports this function. This is mainly considered to be a regular set of protocols that allows real-time communication in order to create video chat apps and peer-to-peer video streaming applications.
WebRTC is a browser’s feature, so whenever a browser supports this technology, a user’s device is absolutely ready to run a WebRTC based app. Mobile users hold a good advantage as it has low bandwidth requirements. Moreover, WebRTC allows removing connections to a server and cloud which has a great positive impact on the performance with a lower connection speed. With the usage of WebRTC, there is a remote chance of losing any video and audio quality.
This technology rapidly enhances and increase the attractiveness of social media by providing an additional way of interpersonal communication. A customer can access and control a video device via web browser. WebRTC primarily allows the developers to focus on user experience rather than on media streaming as the API takes care of the media engine.
Let’s consider some examples of how you can implement WebRTC live streaming functionality.
Advantages of WebRTC
- It's free WebRTC is an open-source application programming interface (API)
- Platform and device independenceWebRTC enabled browser with any single operating system and a web services app can control the browser to build a real-time voice or video connection to any another WebRTC device or a media server.
- Secure voice and videoWebRTC has an always-on voice and video encryption. The Secure RTP protocol (SRTP) is mainly used for encryption and confirmation of both voice and video. This is extremely useful in WiFi networks.>
- Advanced voice and video qualityWebRTC uses the Opus audio codec which produces high fidelity voice. The VP8 codec is used for video. These selections make sure that the interoperability is maintained and it avoids the need for codec downloads which sometimes may contain malicious code.
- Reliable session establishmentWebRTC supports a reliable session establishment. The reliable operation avoids server-relayed media and thereby reduces latency and increases quality. It also reduces the server load.
- Multiple media streamsWebRTC is an adaptive network solution that can do adjustments in changing network conditions. It adjusts the quality of communication, responds to the availability of bandwidth, detecting and avoiding congestion. This is achieved using the multiplexed RTP Control Protocol (RTCP) and Secure Audio Video Profile with Feedback (SAVPF).
- Adaptive to network conditionsWebRTC supports the negotiation of multiple media types and endpoints. This produces an efficient use of bandwidth by delivering the best possible voice and video communications. The APIs and signaling can agree on the size and format for each endpoint separately.
- Interoperability with VoIP and videoThe most prominent advantage of WebRTC is its commitment to interoperability with the current voice and video systems. This almost includes all the devices that use SIP, Jingle, XMPP, and the PSTN. Alternately, gateways can be one of the best solutions to interoperability.
- Rapid application developmentDevelopment process becomes fast and reduces the time for application implementation. Detailed knowledge of WebRTC is not necessary because of the standardized APIs.
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