Real-time multimedia communications with WebRTC
Seamless real-time customer escalation from the web to the contact centre
One technology that is set to simplify the integration of real-time communications to all your apps, websites and online services is WebRTC (Web Real-Time Communication), the open source project initially acquired by Google and now made available on an open-source basis. WebRTC effectively turns the Web into an open communication platform enabling your customers to make audio and video calls, send IMs or post documents, images and other media directly from their Web browsers.
In practice, WebRTC has the potential to dramatically change our use of voice. Because WebRTC is browser-based it can create communications endpoints from a wide range of devices, allowing people to seamlessly transfer their conversations from laptops to iPads, or from their mobiles to in-car audio – or any combination that's applicable.
And unlike proprietary services such as Skype that require a dedicated client and a registered customer account, WebRTC will support core applications such as browser-based telephony, single and multiple audio and videoconferencing sessions, as well as more complex telepresence services. This gives your customers the ability to initiate voice or video calls directly into the contact centre without the need for any additional software.
The result will be a seamless real-time customer escalation from the web into the contact centre, supported by all the context that will provide customer service teams with details of where exactly customers are on their digital journey. This is critical, for while potential developments such as click-to-call or click-to-video chat directly from apps, will prove popular, they can only prove successful if they take full advantage of your core routing engines and are managed in the same way as other interactions.