The power of enhanced WebRTC
With more digital communication tools available than ever before, enterprises and end users have started to demand more tailored options that can improve their experience and make it even easier to connect.
Web Real-Time Communication (WebRTC) is one such tool that can enhance deployment of voice and video and tools within browsers and apps.
WebRTC: The Basics
WebRTC is an open source standard that allows browsers and apps to communicate directly with other apps and browsers without using external plugins. WebRTC enables web browsers to connect easily with others in real-time, from any device, while also allowing voice and video communication within web pages. It’s a technology we are seeing used more and more in common tasks such as real-time collaboration and screen sharing.
As developers look to differentiate their offerings to gain competitive advantage, they can explore deeper customizations of existing tools. For example, enterprises interested in improving their customer support or online user experiences can work with their IT departments to create more customized deployments of WebRTC that will result in more enhanced voice and video communications. Below are a few ways IT teams can leverage existing communication tools to evolve WebRTC deployments.
The WebRTC, VoIP & SIP Family Tree
WebRTC is becoming more recognized as a part of the fabric of our everyday communications. And due to its functionalities that enable real-time communication, WebRTC has the potential to make similar technologies like VoIP and SIP/RTP even more powerful.
Further, because WebRTC allows voice, video, and other media to be transmitted in their most basic formats, it’s considered to be a derivative technology to VoIP. SIP and WebRTC are complementary, though VoIP is used mostly for voice exchanges. VoIP and SIP/RTP are compatible with the existing network, instead of being designed only for rich browser apps or mobile apps. Though WebRTC and SIP do not need each other to function, uniting their capabilities can help users boost their communications options.
On the other hand, SIP and WebRTC can both include non-voice data, which is one reason why industry experts have highlighted SIP as a tool that can be leveraged to enhance WebRTC. The customization takes place by reusing a well-defined standard and a rich set of features already in use. SIP is also the standard VoIP tool in telecom and is closely related to HTTP, which makes integrations of SIP and WebRTC second nature for developers familiar with the protocols. By using SIP as a signaling protocol for WebRTC, developers can simplify interoperability and seamlessly integrate existing systems and PBX.
When connecting WebRTC to SIP, it is possible to extend communication possibilities exponentially. A few benefits of their integration range from improved user experience with one-click audio communication to seamless integration with existing systems and PBX, which allows legacy equipment to connect with users on the Web.
Another benefit of using SIP/WebRTC is the option for HD audio quality. In some cases, the combination of SIP and WebRTC may also offer more reliable audio transmission through codecs that come with WebRTC such as Opus (a fork of Skype/Silk). This option is better suited to be used over public Internet as the codec is already well-integrated and tested in PBX like FreeSWITCH, Asterisk and other modern softphones. Additional benefits can be realized by using SIP like chat (for one-on-one or groups), presence, registration/NAT traversal and others. Many of them may already be supported on the existing PBX.
Enhancing Browser Experiences with WebAssembly
WebRTC can also be deeply customized by integrating WebAssembly into applications. WebAssembly is up leveling media applications on the Web and powering the next generation of rich web and mobile client applications. With WebAssembly you can now create media processing features, benefiting from code running as fast as compiled C/C++ with hardware optimization. Like native code on Android devices, WebAssembly allows for integration of new codecs, noise suppressors and speech/image recognition, among other features directly into browsers.
As developers look to improve customer support or user experiences by deepening WebRTC customizations, in particular for enhancing voice and call capabilities, WebAssembly will be one of the key tools that drives improvements for browser experiences and provides increased differentiation across communication offerings.
WebRTC provides multiple ways to enhance browsers and facilitate the cloud-based real-time communication that are used more often in the new remote work ecosystem. Lastly, with the global workforces’ increasing reliance on digital solutions for connection and communication across remote locations, there will be a need for tools, such as enhanced WebRTC, that will improve digital engagement and communication.
Darach Beirne is vice president of customer success at Flowroute, now part of Intrado. With more than 25 years of experience building and leading B2B customer success, Darach leads Flowroute’s dedicated customer support team, driving strategy for customer success and improved customer satisfaction. Prior to joining Flowroute, Darach lead professional service and sales engineering teams for providers such as Contenix, Huawei/3Leafsytems, InQuira, Siebel/Scopus and Ingres. He also has assisted high-tech companies develop strategies to improve the customer experience and increase scalability.
Julien Chavanton is the voice platform architecture lead at Flowroute, now part of Intrado. As a voice software engineer, open source/free software enthusiast Julien has spent the last 20 years hacking and engineering. He started his career in computer and telephony integration in 2000 contributing to GNU/Bayonne, where he became an active contributor to a variety of other open source projects like Kamailio, Freeswitch and Linphone, etc. Outside of his work, Julien enjoys reading and studying open source software to continuously improve his skills.