With advancements in communication technology, many enterprises can now differentiate themselves by providing personalized services and customer support. In customer support and contact center situations, for example, real-time communications tools such as Web Real-Time Communication (
WebRTC), enables cross-platform and context-based support to enhance interactions between both parties. These elements provide context on the history of the customer and their preferred platform for interacting (i.e., web browser, in-app, phone call, etc.). The combination of these elements improves the customer experience as they seek resolution to their questions and empowers businesses to help customers in a timely and effective manner.
Today, a business’s capability to leverage WebRTC and deeply personalize it for their customers makes connections simpler and more streamlined for participating parties, while also allowing the business to gain a competitive advantage.
WebRTC isn’t a new offering. Since its creation in 2011, the technology has consistently enhanced deployment of voice and video tools within browsers and apps. As an open-source standard, WebRTC makes it possible for browsers and mobile apps to communicate directly with others in real-time, from any device, without extra plugins or communication service providers. With WebRTC, developers can build contextual applications that deliver relevant information to users with the proper functionality through the appropriate user interface at the right time and place.
WebRTC is a tried and true way to simplify and enrich direct communication and collaboration. It’s often still considered an “emerging” communication solution because as an open-source tool, it can be combined with other applications to enable richer and more complex deployments of voice and video communications.
Though it’s still becoming part of the landscape, WebRTC will eventually replace most native apps found on mobile phones and tablets — making it much more than a web-based application. Anything that applies about connecting web users is now true when it comes to connecting mobile users.
Below are two ways developers and IT departments can customize WebRTC to allow customers better control over their browser behaviors for web-based telecom tools. As businesses customize in these ways, they will build stronger communication applications for their customers and partners.
Incorporate WebAssembly into WebRTC Applications
One example of deeper WebRTC customization is the addition of WebAssembly. WebAssembly allows for the creation of media processing features by running code as fast as compiled C/C++ with hardware optimization allows.
WebAssembly brings WebRTC to the next level by integrating new codecs, audio controls, image recognition, and other features into browsers. The integration of WebAssembly will evolve WebRTC deployments in call centers and enterprise collaboration settings, resulting in a new generation of rich web and mobile applications.
As developers continue to enhance their WebRTC customizations, specifically to enhance voice and call capabilities, WebAssembly will be a leading tool that optimizes the browser experience and delivers increased differentiation across communication offerings.
Enhance VoIP & SIP Capabilities
SIP has surpassed its competition and has become the de facto telecom standard. SIP shares many concepts with HTTP, which makes it relatively easy for developers to use and understand. The history and increased access to SIP have made implementations and deployments of the protocol more widespread throughout real-time communications.
WebRTC is a derivative of VoIP technology, mainly SDP/RTP in SIP/SDP/RTP, though it’s important to know that SIP is complementary to WebRTC—not comparable. It does provide efficient transmission of real-time voice, music, video, or other data in their most basic formats, directly over an Internet connection from a Web browser. While VoIP is used mostly for voice communications, SIP can include other data such as video and other media forms. Though
WebRTC and SIP don’t need each other to function, connecting them together can help users extend their communication possibilities exponentially.
Some of the most attractive benefits of integrating WebRTC to SIP include improved user experience with one click-audio contextual communication and the ability to receive inbound calls over the Internet without crossing the PSTN. This connects legacy PBX equipment with users on the web with one well- defined protocol.
Another result of using SIP/WebRTC instead of PSTN would be HD audio quality. Some scenarios would also result in improved reliability of audio transmission by using codecs included in WebRTC like Opus (a fork of Skype/Silk), which is better suited to be used over the public Internet. This codec is already well integrated and tested in PBX like FreeSWITCH, Asterisk, and most modern softphones. Other benefits can be obtained by using SIP like chat (even group chat), presence, registration/NAT traversal, and others many of them may already be supported on the existing PBX.
As WebRTC adoption increases, VoIP and SIP will become even more robust, user friendly, and flexible. By leveraging deeper customizations of WebRTC, businesses can offer greater web browsing experiences to their customers while also delivering more robust communications tools. As customers continue to value personalized communications and platforms for interactions, further innovation of tools that facilitate simple, direct, and custom communications will be a critical way to differentiate against the competition.
