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How to deploy telephony resources with WebRTC

In our last post, we took a look at how WebRTC might be embraced by the broad telecoms industry. This time, we’ll explore how telephony resources can be advantageously deployed in conjunction with WebRTC.

WebRTC is a compelling technology trend that has changed ideas about how customers, end users and subscribers can communicate with businesses and enterprises. With WebRTC, the promise is that they will be able to communicate using any browser capable, connected device – from PCs to laptops and smartphones to tablets.

WebRTC is a means of enabling real-time communication (RTC) from within the browser. WebRTC equipped browsers will be able to connect with each other, enabling users to initiate voice and video sessions, and even to exchange files and share screens. In addition, users will be able to establish multi-party sessions i.e., conferencing.

The need for telephony resources

Despite the idea that there’s no need for a separate communication solution if you can simply use the browser, there are many use cases where there will be a need for additional functionality such as telephony media resources and particularly, interconnectivity. That means associated software applications and WebRTC gateways will be needed.

In the last post, we showed how service providers and others offering contact centre solutions, can benefit from WebRTC, either to enhance their current offerings or to develop new, customer-centric applications. Let’s now look at a couple of those use cases in a little more detail.

Communication channels

Communication channels are the lifeblood of contact centre operations and there’s no doubt adding WebRTC to the list of options is going to be a requirement, sooner rather than later. If WebRTC is going to be just another method of reaching the agent, those business units will need to accommodate a new WebRTC channel.

The last thing an agent needs is more accoutrements; having separate means of receiving (or making) different types of calls is an anathema. Therefore, routing WebRTC initiated calls to an agent means terminating those calls at the agent’s existing console, which is quite likely to be a SIP phone. The solution has to fit seamlessly into the framework of today’s enterprises.

That means, for those with well-established infrastructures and methods (i.e., a lot of contact centers), WebRTC gateways.

Such gateways will interwork between Internet/Cloud-based callers and agents equipped with their standard issue SIP phone and headset. At the technology level, the gateways will deliver independent signalling and media interworking capabilities. Those will include the functions needed to interwork diverse over-the-top and legacy TDM/SIP signalling protocols (yes, SIP is now legacy – it’s twenty years old!) and bridge between media streams, including transcoding.

Call media handling

No one would argue against enabling a potential customer to dynamically initiate a real-time call with an agent being good for business. However, it would be naive to assume that contact centres will not present WebRTC callers with the ubiquitous IVR menu. You better believe it!

In practice, whether it is indeed IVR functionality or it’s e.g., providing the ability to leave a message (record and playback), or it’s to support agent monitoring (third party supervision), telephony resources will be needed.

In addition, full conferencing capabilities can be envisaged, with participants using a mix [sic] of end points, from legacy TDM and SIP, to tablets and smartphones with WebRTC-enabled browsers.

Therefore, many of the critical use cases involving WebRTC depend on what you might call the media server, which represents a key part of the equation.

Of course, the ideal place for these gateways and telephony media resource servers (media mixing, any-to-any connectivity, etc.) to reside is in the ‘Cloud’, which is tailor made for a WebRTC environment.

After all, those WebRTC calls will originate in the global inter-web (a.k.a. the Cloud) and it’s in the Cloud where the benefits of on-demand scalability and availability come into their own.

Now that we’ve helped to demystify WebRTC, here’s some evidence that we’ve been actively working on its integration – check out our demo.