Raising the Bar for Business Communications with Deep Customization of WebRTC

Darach Beirne

Business Communications and WebRTC

By Darach Beirne, Vice President of Customer Success at Flowroute, now part of Intrado, and Julien Chavanton, Voice Platform Architecture Lead at Flowroute, now part of Intrado

With rising customer demand for personalized online experiences, businesses have sought new ways to stand out among the competition while also delivering the next generation of customer experience.

Offering customized service and customer support is one way that enterprises are meeting this demand. As an example, some businesses are tailoring their customer support and contact center situations using real-time communications tools such as Web Real-Time Communication (WebRTC)—a technology that enables cross-platform and context-based support to improve interactions between customers and support services.

Using WebRTC, enterprises can gain important historical data and context on a customer, such as their preferred platform for contacting customer service (i.e. via web browser, in-app or phone call). This type of approach enhances the customer’s ability to resolve their question or issue while also making it easier for the business to provide support.

WebRTC has been around since 2011. Over the years, it has been used to enhance deployment of voice and video tools within browsers and apps. Because it is an open source standard, WebRTC enables browsers and mobile apps to communicate directly with others in real-time, from any device, without extra plugins or communication service providers. Using WebRTC, developers can create contextual apps that provide the information to users through the right channel.

WebRTC is a proven way to simplify and enrich direct communication and collaboration. As it becomes more deeply enmeshed in the landscape, WebRTC will someday take the place of most native apps on mobile phones and tablets. This shift would take it beyond a web-based application and anything that would apply about connecting web users would be true when connecting mobile users.

There are two key ways developers and IT departments can harness WebRTC to give customers better control over browser behaviors for web-based telecom tools. When businesses become more customized, they will gain stronger communication applications that give them a competitive advantage.

Amplify WebAssembly with WebRTC

WebRTC can be more deeply customized with the addition of WebAssembly, which allows for the creation of media processing features by running code as fast as compiled C/C++ with hardware optimization allows.

WebAssembly increases the power of WebRTC by allowing integration of features like new codecs, audio controls and image recognition into browsers. Integrating WebAssembly enhances deployments of WebRTC in call centers and enterprise collaboration settings resulting in richer web and mobile applications.

As developers work to enhance voice and call capabilities through WebRTC, WebAssembly will be one of the primary methods to improve browser experiences and generate more powerful customer communication platforms.

Enhance VoIP & SIP Capabilities

While WebRTC is becoming more deeply woven into the fabric of technology, SIP has become the default telecom standard. SIP and HTTP share many concepts, making it fairly easy for developers to learn and incorporate.

WebRTC is related to other VoIP technologies like SDP/RTP in SIP/SDP/RTP, though SIP is complementary to WebRTC but isn’t comparable. While VoIP is relied on for voice communications, SIP can include data such as video and other media. Though WebRTC and SIP can operate independent of one another, uniting them can allow for enhancement of communication possibilities.

A few of the key benefits of bringing WebRTC and SIP together include upgraded user experience with one click-audio contextual communication and the option to receive inbound calls over the internet without crossing the PSTN. This makes it possible to connect legacy PBX equipment with web users with one efficient protocol.

Using SIP/WebRTC instead of PSTN can also improve HD audio quality. In some instances, it could result in more reliable audio transmission using codecs that come with WebRTC like Opus (an offshoot of Skype/Silk). This codec is already well integrated and tested in PBX like FreeSWITCH, Asterisk and many modern softphones. Users can get even more benefits by using SIP like chat or registration/NAT traversal.

As WebRTC becomes more prevalent, VoIP and SIP will also become even more robust. By implementing deeper WebRTC customizations, enterprises can create improved web browsing experiences while also delivering savvier communications tools. Because customers will continue seeking personalized communications and platforms, creating tools that enable simplified and tailored communications will continue to be an important way to stand out against competitors.

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