With the range of digital communication tools available on the market, businesses and end users have begun to demand more customised offerings that can improve their experience and make connection simpler and easier.
While once considered disruptive and not complimentary to existing methods of digital communication, Web Real-Time Communication (WebRTC) is one such technology that enhances deployment of voice, video and chat tools within browser and app experiences.
What is WebRTC?
WebRTC is an open source standard that makes it possible for browsers and mobile apps to communicate directly with others without extra plugins or providers. Since its inception in 2011, WebRTC has enabled web browsers to communicate easily with others in real-time, from any device, while seamlessly allowing voice and video communication to function within web pages.
WebRTC is used for a number of common tasks, such as:
- Adding video to endpoints (like ATMs and retail kiosks)
- Collaborating in real-time
- Building contextual applications
- Sharing screens
- Sharing data
By simply making direct communication easier and more enriched, it’s clear that WebRTC can provide value on its own. But what about deep customisations of WebRTC? How are tailored uses of WebRTC enabling more enriched and complex deployments of voice and video communications? Below are a few ways I envision developers and IT departments will customise WebRTC to enable customers more control over their browser behaviours for web-based telecom tools and to improve how businesses build better applications that facilitate communication.
Expanding VoIP & SIP Capabilities
As WebRTC becomes a widely adopted standard, VoIP and SIP/RTP will become even more robust, user friendly, and flexible. These technologies are already compatible and offer some of the same functionalities, but it’s important to understand how they’re different and how WebRTC can enhance SIP.
WebRTC is actually an offshoot of VoIP technology because it enables voice, video, chat, or other data to be transmitted in their most basic explanations, directly over an internet connection. SIP and WebRTC, for the same reason, are derivatives of VoIP. However, VoIP is used mostly for voice communications, while SIP, like WebRTC, can include other data.
“It’s also important to note that VoIP and SIP/RTP vary from WebRTC because they are compatible with the existing network, instead of being designed initially only for rich browser apps and now more often found in mobile apps.” With these differences and similarities in mind, experts have found that WebRTC can benefit from SIP functionalities by reusing a well-defined standard and a rich set of features already provided and implemented by many, making it simple to interconnect WebRTC endpoints.
SIP has won the VoIP signalling war, it is now de facto standard in telecom and all mobile phone calls are now using it. SIP has solved many problems, it is the brother of HTTP so many concepts are also present in SIP making it easy to understand by most, even if it is quite complicated to implement a SIP stack using one is not that hard. With the enduring success of SIP, it is now heavily used, implemented and deployed, by using SIP as a signalling protocol for WebRTC we can benefit from this great success and simplify interoperability and seamlessly integrate existing systems and PBX.
By connecting your WebRTC to SIP, you can extend your communication possibilities exponentially.
Benefits of integrating WebRTC to SIP include:
- Improved user experience with one click-audio communication
- The ability to receive inbound calls over the internet without crossing the PSTN
- Seamless integration to existing systems and PBX, making is possible for legacy equipment to connect with users on the Web
- Reusability of many features already well defined in SIP, which has been very successful at solving interoperability problems
- Additional information about the caller through contextual data in SIP headers, or other SIP methods like MESSAGE
- HD Voice when using the Opus codec-QoS call reports over SIP in addition to RTCP-Shortest media path by using our multi-cloud provider point of presence
Better Browser Experiences with WebAssembly
Another example of a deep customisation of WebRTC is by integrating WebAsemmbly into WebRTC applications. With WebAssembly it is now possible to create media processing features, benefiting from code running as fast as compiled C/C++ with hardware optimisation. Very similar to native code on Android, WebAssembly makes it possible to integrate new codecs, noise suppressors, speech/image recognition, among other features into browsers.
WebAssembly is bringing media applications in the Web to the next level and enabling a new generation of rich web and mobile client applications. As developers continue to enhance their WebRTC customisations, specifically to enhance voice and call capabilities, WebAssembly will be a leading tool that improves the browser experience and provides increased differentiation across communication offerings.
WebRTC offers a number of benefits to enhance browser behaviour and execution of communications tools which are increasingly relied upon in today’s modern climate. As the world continues to turn toward digital solutions for connection and communication across remote locations, there remains an opportunity to create tools that will facilitate better digital interactions in a simpler and more direct way.
Guest Blog by Darach Beirne, Vice President of Customer Success, and Julien Chavanton, Voice Platform Architecture Lead at Flowroute, now part of Intrado
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 is 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.
