YOU ARE AT:OpinionReader Forum: Don’t get caught up in the WebRTC hype without a...

Reader Forum: Don’t get caught up in the WebRTC hype without a game plan

Editor’s Note: Welcome to our weekly Reader Forum section. In an attempt to broaden our interaction with our readers we have created this forum for those with something meaningful to say to the wireless industry. We want to keep this as open as possible, but we maintain some editorial control to keep it free of commercials or attacks. Please send along submissions for this section to our editors at: [email protected]

The world is fascinated with WebRTC and its potential. Consumers are drawn to WebRTC’s promise of convenience and its ability to simplify their lives. No longer is there a need to download proprietary software to make a voice or video call on a PC or tablet, or to conduct a video conference call. Businesses see the opportunity in implementing WebRTC to offer their services in a more accessible way for customers. The user experience in addressing customer service issues is often cited as a good example of this – with the click of a button, customers can interact with the customer care center through an audio/video session. WebRTC also creates the potential for businesses to craft new revenue-generating service offerings. Meanwhile, service providers are interested in WebRTC because it greatly expands their addressable market – to include any browser-enabled device.

Although WebRTC is generating a very high level of interest and excitement in the industry, it will be operating in an environment that has already adopted, and continues to adopt, SIP in a big way, both in the enterprise and service provider market segments. In order for WebRTC to succeed, it cannot function as a separate communication island, but rather must interwork seamlessly with existing SIP-based networks to support real-time voice, video and data. Two important factors IT departments need to consider for this are: security and interworking.


Enterprises cannot implement new products or services that make them vulnerable to cyber attackers, so they need to be judicious about network security when WebRTC browsers start communicating with their networks. Opening such an application presents the possibility of the threat of a malicious web application’s taking over a user’s browser and directly communicating with a contact center – a scenario IT managers find it critical to avoid.

Consider a traditional SIP-based contact center that now has to support WebRTC clients. By hijacking unsuspecting WebRTC clients, cyber attacks can cause excessive bandwidth and CPU utilization at the contact center, filling up of storage disks with useless data and causing damage to system and IT network configuration. This can result in service revenue loss, network and disk storage repair and rebuild costs, and a host of other negative consequences for the contact center.

A security gateway, such as a session border controller deployed at the edge of the enterprise network, can provide the functionality necessary to protect the enterprise from WebRTC-initiated cyber attacks. By monitoring traffic, the SBC can recognize threats to the network and shut them out.

At the same time, interworking is another important part of a comprehensive WebRTC game plan. Every enterprise – and consumer – is at a different stage in adopting the latest and greatest technology, and playing nice with old and new is necessary for WebRTC to be adopted widely.

Interworking is critical to ensure seamless communication between endpoints with different characteristics (WebRTC browser to IP phone, or WebRTC browser to PSTN phone, for example), and to ensure calls can be made between different kinds of networks (IPv4 and IPv6 networks, for example). Interworking is required for both signaling and media (voice/video) communication. Signaling interworking ensures that SIP-based unified communications products from different vendors, like a PBX, work both with each other and with WebRTC clients. Media interworking presents a greater challenge because of the plethora of standards (codecs, protocols) for video and voice communications. H.323, SIP and WebRTC video communication islands all must interwork for widespread adoption of UC video conferencing to happen. At the same time, video transcoding, trans-rating and tran-sizing must be done in real time, while policy control effectively manages the quality of service and network bandwidth for sessions, since UC video communication can involve large volume of data.

A multimedia-enabled SBC deployed at the demarcation of WebRTC and SIP networks can support all of these criteria: IPv4 to IPv6 interworking, communication between WebRTC and different flavors of SIP, transcoding between WebRTC and SIP voice and video codecs, traffic policing, and quality of service management for UC communication.

As the industry looks toward the next transformative technology, it is important to remember that developing a solid game plan is critical to fueling adoption and that integrating with the SIP networks and services already in place will maximize opportunity and success.


Editorial Reports

White Papers


Featured Content