Browser-Based Telephony Poses Threat & Opportunity for NSPs

Dean Bubley, founder, Disruptive Analysis

Dean Bubley, founder, Disruptive Analysis

By Fred Dawson
June 27, 2013 – Once again developments in over-the-top applications are creating a pressure point threatening to marginalize network service providers, this time in the realm of the next generation of unified communications.
The disruptive force in this emerging scenario, known as WebRTC (Real-Time Communications), allows users with WebRTC-enabled browsers to access robust video and voice communications apps embedded in Web pages from any type of Internet-enabled device over any type of network without special plugins and installs. The technology allows Web developers to easily create communications apps and embed them into existing Web pages, much like video apps can be embedded for browser access through HTML5.

WebRTC, which leverages the object-based JavaScript architecture along with two key JavaScript APIs, can be used for pure OTT peer-to-peer communications that completely bypass existing voice services infrastructures. At the same time, by enabling access to SIP (Session Initiation Protocol)-related technologies such as STUN/ICE (Session Traversal Utility for Network Address Translation/ Interactive Connectivity Establishment) and SBC (Session Border Control) from the browser, WebRTC makes it possible to connect Web-originated calls through the mobile and fixed IMS (IP Multimedia Subsystem) and PSTN telecommunications infrastructures.

In other words, WebRTC can be used to create an app-development environment that could greatly benefit network service providers as well as OTT entities. But while there are WebRTC initiatives in play here and there among carriers such as AT&T, BT and Telefonica, it remains to be seen whether NSPs can adapt quickly enough to get on the offensive as opposed to defensive side of the trend.

The problem is, with the standard not yet stabilized, Web entities are racing ahead to take advantage of the benefits. Leading the Web charge are browser suppliers Google, Mozilla and Opera, who have embedded WebRTC capabilities in hundreds of millions of users’ Chrome, Foxfire and Opera browsers, totaling over a billion by some estimates. This has spawned a global burst of development activity by app developers and Web entities large and small.
“This pace is making it difficult for [telecommunications companies] – and associated vendors/enabler partner companies – to know how and where to engage with WebRTC,” says Dean Bubley, founder of the research firm Disruptive Analysis. “It is a moving target, where even the direction is unclear. It is not yet standardized, is being drawn along multiple paths simultaneously, is subject to numerous uncertainties in terms of codec support, uptake path and developer engagement – and has a different cast of companies acting as proponents, sponsors or blockers.”
Nonetheless, Bubley advises in a recent blog, NSPs can’t afford to sit still. “Disruptive Analysis exhorts telcos of all stripes – mobile, fixed, cable, wholesale, MVNOs and so forth – to get involved in WebRTC as quickly as possible, even if it is just for learning,” he says.

“Even mistakes are valuable,” he adds. “Sticking with the normal telco multi-year cycle of standardization, trial and adoption is not an option. Unless operators are thinking in timescales of weeks and months – and are willing to be seen doing experiments – they will fall even further behind, as WebRTC speeds up the OTT players’ innovation cycles.”

WebRTC, now an emerging standard within the IETF and WC3 HTML5 initiative, is based in part on proprietary codecs and other innovations Google obtained with its acquisition of Global IP Solutions and On2 a few years ago and which it has since made available to the open-source community. Whereas one strategy that Google promoted following those acquisitions, namely, an open-source video compression and streaming platform known as WebM, has not taken hold, WebRTC is another story.

The standard pulls together these and other advanced communications technologies in conjunction with the use of Real-time Transport Protocol (RTP), a cornerstone of VoIP, to create a solid foundation for robust real-time voice and video communications over any type of network. For example, the platform is designed to support audio encoding suited to virtually any bandwidth and device environment by offering three choices of audio codecs – iSAC (internet Speech Audio Codec) supporting 8 KHz sampling with bitrates at 13.33 kbps and 15.2 kbps, depending on frame rates; iLBC (internet Low Bit Rate Codec) supporting wideband and super wideband (16 KHz or 32 KHz sampling) with adaptive variable bit rates of 12 to 52 kbps, and Opus, which supports constant and variable bitrate encoding from 6 kbps to 510 kbps along with frame sizes from 2.5 ms to 60 ms, and various sampling rates from 8 kHz to 48 kHz.

As for video, WebRTC employs the WebM VP8 codec, which is designed to support low-latency communications. The platform also includes a framework for the end-to-end video media chain known as VideoEngine, which enables seamless connectivity from cameras through networks to screens. There are also highly specialized components for dynamic voice and video jitter protection, audio noise reduction, video image enhancements to improve image capture by webcams and echo cancellation.

WebRTC uses two APIs to support app development, one for creating the full suite of applications, including communications components, and the other providing pre-configured video and audio chat capabilities to be incorporated into additional app components. A third API supports integration into browsers. There are also two key JavaScript APIs, one for capturing and turning video and audio data from a user’s device into usable JavaScript objects and one that allows peer-to-peer connectivity from one user’s browser to another’s browser.

To enable maximum flexibility in how applications are used and monetized beyond the pure peer-to-peer mode WebRTC does not specify protocols for the signaling and session management layer. Instead the standard uses HTML5 WebSockets to interconnect the apps to whatever protocols the app supplier wants to use in instances where the communications streams are to be transmitted through gateway servers, such as are required to interconnect the WebRTC apps with mobile phones over mobile networks. The mobile standards body 3GPP is in the process of defining the specifications for such gateways.

WebRTC is not without its detractors. Microsoft, for example, has sought to improve on this dual-API connectivity architecture by proposing a uniform approach to accommodating both pure peer-to-peer and browser-to-server gateway connections known as CU-RTC-WEB (Customizable, Ubiquitous Real Time Communications over the Web), and Apple has so far shown little interest in supporting the emerging standard on its Safari browser or on its devices. “A lack of clarity from Apple and Microsoft on WebRTC support timelines is a particular hindrance in smartphones,” Bubley says.

But it remains to be seen how detrimental these uncertainties will be. So far, W3C has merely recognized CU-RTC-WEB as a potential specification for WebRTC without formally including it in the proposed specifications. Moreover, absence of support in Safari does not appear to be a major barrier, given the scale of support in Foxfire, Chrome and Opera. WebRTC has even found its way into Microsoft’s Internet Explorer, owing to the fact that WebRTC can be supported on recent versions of Explorer that run Chrome Frame.

WebRTC, especially with the support of Opera, a strong browser presence in mobile Web usage, is meant to surmount issues associated with the mobile environment, but it has been slower to take off in that space than in fixed Internet applications. This is an opportunity for wireless SPs.

Leading telecom vendors, including Ericsson, Cisco, Alcatel-Lucent, Acme Packet and GENBAND, have embraced WebRTC as an opportunity for NSPs of all types, providing the network-based components that can extend applicability to all devices. Ericsson, for example, has incorporated WebRTC into its Bowser browser, which runs on iOS and Android devices. And the supplier has introduced a development platform with WebRTC APIs for creating WebRTC apps, enabling extension of apps to any device using an existing mobile number.

As for Alcatel-Lucent, “We describe our Web RTC strategy as a one-network open offer,” says Mike Lambert, a solutions director for Alcatel-Lucent. The one-network concept means “you leverage your IMS investment to deliver seamless services from VoLTE to Web applications with use cases that give your customers the ability to consume services the way they want to. Open means combining A-L’s new conversation APIs with our WebRTC client libraries so that our customers can rapidly deliver new services to customers in Web-based use cases as well as allow third-party developers to take advantage of services from their networks.”

The solution set allows service providers to extend existing IT communications service packages developed for business customers into the WebRTC environment, he adds. “This is important, because WebRTC isn’t just a browser, a gateway or a network. We at Alcatel-Lucent are investing across all these areas to deliver a solution that service providers can use to quickly leverage their strengths and brands into Web services.”

Alcatel-Lucent is also pitching WebRTC ideas to CableLabs, notes Jay Fausch, senior director of market development at A-L. “We have an ongoing dialog with CableLabs about how cable operators can use our APIs to leverage elements of the IMS infrastructure to quickly activate communications from users’ browsers,” Fausch says.

The U.K.’s BT is among the first group of NSPs jumping on the WebRTC bandwagon. “WebRTC gives us better capabilities with communications infrastructure at lower cost and better services,” says Ed Stadtmueller, solutions director for unified communications at BT. He notes that unified communications is a hot item for enterprises but has been slowed because “it’s dominated by specific vendors and specific vendor solutions. This really changes that for BT, for a global service provider. It’s not based on specific closed enterprise solutions. It’s based on an open approach.”

Stateside the service provider that has been most vocal about WebRTC is AT&T, which has made clear the technology is an integral part of its voice-over-LTE (VoLTE) initiative, which in turn is part of the previously reported Project VIP (Velocity IP) initiative aimed at developing a broadband-based service infrastructure. AT&T participated in a demo of WebRTC in action at the World Mobile Congress in February with Ericsson and Foxfire supplier Mozilla and has made app development on WebRTC a focus of its work with developers through the AT&T Foundry. The app shown at WMC allowed the Firefox browser to sync with the user’s phone number, allowing communications to be conducted as the user is doing things on the Web without downloading any plugins.

AT&T’s embrace of WebRTC is a vivid illustration of how a technology that looks like a cannibalizing intrusion on the way things are done today by mobile and fixed service providers can actually become the heart of a new approach to providing consumer and business services where terms like “the connected life” and “ubiquitous broadband” take on new meaning. Or, as AT&T executives phrase it, a service provider can get into the “personal intimacy business” by giving everyone a “personal digital footprint” in the personalized cloud.