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
The digital communication landscape is constantly evolving with the emergence of new tools that provide enterprises with ways to collaborate and improve user experiences. The end users encounter new tools, enterprise decision makers seek ways to integrate new features into existing services and tailor those features to meet customer and end user needs. When it comes to evaluating new digital communication tools, enterprises primarily look for solutions that will improve interactions and make it easier for customers to connect with the business and each other.
Web Real-Time Communication (WebRTC) is one such tool that is enabling the customization of interactions both internally and externally.
What is WebRTC?
WebRTC is an open source standard that allows browsers and apps to communicate directly with other apps and browsers without using external plugins. WebRTC’s purpose is to enable rich, high-quality RTC applications that can be developed for browsers, mobile platforms and IoT devices, and allow all applications to communicate via a common set of protocols. WebRTC is a technology that is becoming more and more common in facilitating real-time collaboration.
As developers look to differentiate their company’s offerings to gain competitive advantage, they can explore deeper customizations of existing tools. Despite its inception in 2011, WebRTC is still considered an emerging tool and will eventually replace most native apps found on mobile phones and tablets—making it much more than just a web-based application. Any rules or features that applied when connecting web users is now true when it comes to connecting mobile users.
Below are a few ways developers and IT teams can customize WebRTC to afford customers better control over their browser behaviors for web-based telecom tools.
WebAssembly Strengthens Browsers
Developers and IT teams can customize WebRTC by incorporating WebAssembly. WebAssembly is up leveling media applications on the Web and powering the next generation of rich web and mobile client applications. This tool allows for the creation of media processing features by running code as fast as compiled C/C++ with hardware optimization allows.
Like native code on Android devices, WebAssembly allows for integration of new codecs, noise suppressors, speech/image recognition, and other features directly into browsers. As enterprise 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.
WebRTC, VoIP & SIP Explained
WebRTC is becoming a part of the fabric of our everyday communications. Due to its functionalities enabling real-time communication, WebRTC has the potential to make similar technologies like VoIP and SIP/RTP even more powerful.
WebRTC is a derivative of VoIP technology, mainly SDP/RTP in SIP/SDP/RTP, but SIP is meant to be complementary to WebRTC rather than comparable to WebRTC. SIP provides efficient transmission of real time voice, music, video or other data in their most primitive formats, directly over an internet connection from a Web browser. SIP can also include other data such as video and additional media forms. Though WebRTC and SIP do not rely on each other to function, bringing them together can extend users’ communication possibilities. 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.
Since SIP and WebRTC can both include non-voice data, SIP can be used as a tool to enhance WebRTC. The enhancement here occurs 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 using and understanding the protocol easy for developers. By using SIP as a signaling protocol for WebRTC, developers can simplify interoperability and seamlessly integrate existing systems and PBX.
In some cases, the combination of SIP and WebRTC also offers HD audio quality and more reliable audio transmission through codecs that come with WebRTC such as Opus (a prong 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 these features may already be supported on the existing PBX.
WebRTC provides multiple ways to enhance browsers and facilitate the cloud-based real-time communication that developers and enterprises continue to look for. As technology continues to evolve with user preferences favoring personalized experiences, deeper customizations of WebRTC will drive more enhanced communications. Further innovation of tools that facilitate simple, direct and custom communications will be a promising way to stand apart from competitors.
Source: TelecomReseller
