When providing services for your customers, it’s all about convenience. Imagine a customer trying to find out some information about your service from your website. Her question is very specific and can’t be answered by the FAQs you have on the site. She spots a button on the bottom right of the screen that says “Can’t find what you’re looking for? Talk to a rep now!” She clicks it and is immediately connected via voice to a customer representative and gets her question answered on the spot. Now that’s service!
This is just one of the functionalities that Web Real-Time Communications (WebRTC) provides. A survey conducted by The SIP School found that 88% of professionals have either never heard of WebRTC or only have a cursory understanding of it. Here we present a few facts that will help you better understand how it can benefit your business.
WebRTC is an open-source project that provides browsers and other mobile applications with the capability of real-time communication via the use of simple application programming interfaces (APIs). This allows users viewing a web page to have a rich communications experience with your business that can include text, voice and video – all without downloading any plug-ins or files. Instead, the technology is embedded directly into the browser or mobile app.
The revolutionary aspect of this service is that it can easily be integrated with your VoIP telephony solution, allowing your web portal to be a conduit of voice communications into your business. Take it one step further and you can have users connect directly to a help desk or contact center agent for customer support.
WebRTC was originally envisioned as a peer-to-peer technology, but it can be adapted and integrated into existing communications infrastructure (VoIP telephony, contact centers, and the like) using appropriate gateways and other intermediary devices.
WebRTC does not compete with SIP
WebRTC is not the same as SIP and does not compete with it. On the contrary, they are complementary. WebRTC by itself is not enough to provide real-time communications (RTC). Rather, it lays out the framework and the necessary application interfaces, but it still requires a signaling protocol to transfer content. It was originally designed to use open protocols such as Opus for audio and VP8/H.264 for video. However, it can also leverage SIP (session initiation protocol), making it ideal for VoIP telephony solution integration.
One example of this is Grandstream's UCM IP PBXs, which have WebRTC already integrated and enabled. The Grandstream UCM supports both WebSocket (ws) and encrypted WebSocket (wss) protocols over either HTTP or HTTPS, giving you the flexibility to determine which port you want open for WebRTC and whether or not you want it encrypted. In addition to making your company more accessible to your customers by using WebRTC, Grandstream offers other integration features that elevate your experience, such as integrating with popular CRMs like Sugar CRM or SalesForce.
Click to download the free Grandstream UCM WebRTC set-up guide
WebRTC is secure
Even though content is streamed over the unsecured internet, all media sent over WebRTC is secure. WebRTC uses Secure Real-time Transport Protocol (SRTP) for encryption purposes, using the widely adopted Advanced Encryption Standard (AES) specification with a 128-bit key for encryption, although this can be configured to use key lengths of up to 256 bits.
WebRTC is versatile
WebRTC is widely compatible with today’s web browsers and mobile applications.
CONCLUSION
WebRTC is relatively new in the VoIP universe, and as is the case with all new technologies, it may take a while to become more mainstream. SIP, having been around for two decades, is a mature and widely deployed protocol that has changed the way VoIP has developed. The combination of the two, within the next few years, has the potential to take customer support to a new level by providing a more effective way to engage customers and other stakeholders.
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