It all started with command-line messaging, then text chat took over, and now it has evolved into real-time audio-video chat and it is being considered as one of the most preferred ways of online communication. We at Prologic Technologies have been advocating advantages of WebRTC application development services and have developed many bespoke webrtc applications for our clients.
Advantages of WebRTC in video conferencing application development?
One of the main reasons why WebRTC is preferred for development of live video conferencing applications is because it is open source technology with a lot of improvisation scope. It doesn’t require any third-party software installation as well.
Not only the web, but you can also create applications for smartphones and various handheld devices such as tabs and Phablets using webRTC. Another benefit offered by WebRTC applications is that being a peer-to-peer protocol-based technology it helps you to cut down on bandwidth costs.
Here are some key advantages of WebRTC:
Instant Audio/ Video call using web browsers and native SDK
Google introduced WebRTC, an open-source application programming interface (API), in 2011. The goal of Google’s WebRTC project is to provide a free, real-time media engine that will run in all accessible browsers.
No software installation required
A web services application can lead any WebRTC-enabled browser with any operating system to build a real-time voice or video connection to another WebRTC device or to a WebRTC media server. The browser’s operating system has no bearing on this. This is accomplished through the use of W3C-approved APIs and IETF-approved protocols. HTML5 code can be written by developers for use on both desktop and mobile devices.
Live Audio- Video Broadcasting
The Opus audio codec is used by WebRTC to provide a high-fidelity voice. The Opus codec is based on the SILK codec technology developed by Skype. For video, the VP8 codec is used. These choices ensure interoperability and eliminate the need for potentially harmful codec downloads.
Data encryption and secure transmission
Always-on voice and video encryption is a feature of WebRTC. Both speech and video are encrypted and authenticated using the Secure RTP protocol (SRTP). This is especially advantageous on WiFi networks. This prevents eavesdropping and recording of the speech and video.
Can be used for text messaging, SIP interconnects, & Screen-sharing
The promise of interoperability with existing voice and video systems is WebRTC’s most valuable feature. This includes SIP, Jingle, XMPP, and PSTN-enabled devices. What may hinder global interoperability will be the upgrades necessary in exiting devices.
Gateways, on the other hand, may be the answer to interoperability. Some are already available. Existing speech and video devices will almost certainly operate with WebRTC-based devices if they use standard protocols.
Reliable session establishment
WebRTC allows for a secure session to be established. This is especially true with Network Address Translators (NAT), which obstruct and potentially block other communication and collaboration protocols. The dependable operation prevents server-relay media, lowering latency and improving quality. It also minimizes the burden on the server.
Rapid application development
Developers will benefit from a simplified development approach that will cut application implementation time in half. Because of the standardized APIs, detailed understanding of WebRTC will not be required. Finally, the speech and video codecs are available without charge.
API’s used by WebRTC for development
Generally, there are three APIs used in the development of WebRTC Applications.
Data Channel is responsible for transferring data between both parties. It was designed on the basis of the WebSocket API. Instead of relying on TCP protocol, Data Channel uses UDP- based streams With SCTP protocol. Which makes it capable of both delivering like TCP and a low congestion network of UDP.
RTC peer connection
PeerConnection is an integral part and acts as a core of WebRTC development. It creates a path for participants to establish connections between them without using any intermediate server. There are three types of information that must be relayed before beginning the connection.
The session control is responsible for the session initialization, closing, and modification. Network data is responsible for determining the IP address and port number allowing both sender and receiver to find each other. Media data is responsible for determining the common codecs and media types of both sender and receiver.
Tokbox- Introduction, And Services
TokBox provides an industry-leading WebRTC API – Opentok, which makes easy to embed high-quality and online video chat, screen sharing, text chat, interactive broadcasting features for web and mobile applications. A few of many advantages offered by Tokbox as a PaaS provider are:
- One-to-One and Multi-party Video, Voice and Messaging
- Screen Sharing
- iOS & Android SDKs
- Internet Explorer Plugin
- Text Chat
- SIP Interconnect
- Security (Data encryption and transmission over SSL)
- Audio & Video Driver
- Firewall Traversal
- Audio Fallback
Why Prologic Technologies is the most promising WebRTC development company?
Prologic Technologies is a Bespoke web and mobile development agency which specialize in WebRTC application development. We are preferred agency partners of Tokbox and have delivered over 60 web and mobile apps using Tokbox Opentok API. We have a solid team of experienced Tokbox WebRTC application developers who can not only develop a robust and efficient solution based on Tokbox but can also deliver an enhanced user experience for your web and mobile applications.
Are you looking for a Tokbox WebRTC Application Developer for your next project?
Click the button below to book a FREE consult with our webRTC Tokbox Expert.
You may also like these:
Due to the rapid development in technology, the use of devices like mobiles has increased ...
webRTC is a technology that enables real-time voice and video communication between browse...