WebRTC Solutions Featured Article

IBM WebSphere Liberty Real-Time Communications and WebRTC

September 24, 2015

If you are still skeptical about the momentum WebRTC is gathering, particularly with developers with an eye on the enterprise, you need to rethink things. As noted in another posting this week about AT&T’s promotion of WebRTC on its developers site, readers are encouraged to check out and possibly bookmark the home of IBM’s WebSphere Liberty Real-Time Communications (Rtcomm) feature. As IBM says, “For those struggling to find a WebRTC platform that integrates well with your existing Web and mobile application infrastructure, you’ve come to the right place. If you’re frustrated with solutions that feel more like you’re installing a telephone network than adding a promising new enhancement to your existing Systems of Engagement, we feel your pain.”

The latest posting to the site is illuminating on several fronts. There are such provocative statements as: The WebSphere Liberty approach to supporting WebRTC is different. We view WebRTC as a Web standard, not a telco standard… The vast community of Java and JavaScript developers out there knows little or nothing about telephony.  WebRTC is an HTML5 standard. Shouldn’t solutions that support WebRTC feel the same or at least similar to your existing HTML5 solutions?” They make an interesting point. 

Rather than spoil the enjoyment of replicating all of IBM’s encouragement to HTML5 developers who may know little about telephony, but should seize on WebRTC as a tool to generate significant value-added new services, below is a list of capabilities provided by WebSphere Liberty Rtcomm. It is comprehensive and includes: 

Peer-to-peer calling – Set up an Rtcomm demo using the built in endpoint registry in minutes.

Call Queues -- Queues subscribed-on by agents and called-to by Web users.

Third-Party Call Control – Initiate calls from 3rd-party sources like IoT events.

Contextual Presence – Use context to locate endpoints to connect with.

Real-Time Messaging – Exchange text, audio and video in real-time.

Gateway – Rtcomm-to-SIP gateway for federation with SIP Trunking, IMS and the PSTN.

SIP Servlet Programming Model – For developers wishing to drill deeper into the signaling.

Media server integration – Support for media server control via JSR 309 for multiway, record/playback of media and much more.

They even provide a link on how to enable and configure the Liberty Rtcomm feature.  

The IBM team also provides links to open source JavaScript libraries so developers can deliver real-time features to new and existing JavaScript applications, as well as to the related open source components built on the open Rtcomm signaling protocol. There are also descriptions and links to enable developers to “try it you will like it” starter kits. In addition there is information on how Rtcomm plays well for Internet of Things (IoT) applications, unified communications (UC) and other real-time communications capabilities like Presence. 

One final thing of note that applies to everyone’s WebRTC efforts is that simplicity and interoperability are part of the pitch to developers. This happens to be why WebRTC is busy gaining traction. Indeed, as IBM notes, “While many features and functions can be derived from the Liberty Rtcomm feature with no backend programming, the feature also supports the ability to extend the base platform through the SIP Servlet (JSR 289) programming model and through services like third-party call control that are exposed via MQTT topics, and has been made to be cloud-friendly, and fast and easy to develop and deploy.

In short, whether it is communications service providers or enterprises, WebRTC is creating exciting opportunities for developers and clearly the time for developers to get with the program is now. 

Edited by Rory J. Thompson

Article comments powered by Disqus