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When voice-over-IP really began taking off, the industry imagined the end of the public-switched telephone network; but at that point it seemed very unlikely. With hundreds of millions of analog and digital phones in operation around the world, a pure IP telephony system seemed farfetched at best. Now we are standing on the precipice of change. The Federal Communications Commission and other regulatory bodies are diligently mapping the end of the PSTN. And with that, we will lose one of the core PSTN values: a relatively high guarantee of voice quality and high reliability. PSTN quality was delivered by the very structure of a circuit network, meaning that regardless of where you were connected, the level of service and quality was fairly uniform.
Replacing the PSTN with VoIP
We are now replacing the PSTN with the uncertainty of VoIP and, unsurprisingly, this comes with complications. VoIP has rolled out across the world in two forms: as session initial protocol and as over-the-top. Many industry pundits see an end-to-end SIP-based PPacketTN replacing the PSTN as a future reality. In the new PPTN, SIP becomes the connection between service providers, with each SP managing its portion of the overall path – just as the PSTN is managed today with traditional telephony. While this evolution is visible, it is happening slowly and the assurances of end-to-end quality are still minimal, as there are many transcoding and operational issues. This enables the OTT players to capture a growing market share through new services and aggressive pricing. In fact, by the end of 2014, 50% of international long-distance calls will be made via Skype, one of the most notorious OTT players. This is no accident – closed communities such as Skype and Vonage have created an environment where VoIP could be used at low or no cost – often as a way to avoid international long-distance fees – and are now evolving to be a replacement of the PSTN and traditional service providers altogether. The quality is not great, but it’s free and it offers more features than the PSTN. For a consumer, that’s good enough.
Enter WebRTC
With Web real-time communications, it is now possible to get in touch with contacts on any device and IP network, anywhere in the world, regardless of the OTT apps they have installed or wireless or wired network they are on. With WebRTC, service providers are no longer limited to only serving your customers on their own network. In addition, they can deliver services that rival or exceed those that the OTT players offer. But one of the most important innovations of WebRTC is that it enables enterprises to facilitate communications between them and their customers, or prospects themselves, without the need of a telecommunications service provider.
However, even with these benefits, the big challenge of the Internet remains: there are no quality of service or availability guarantees. This is especially important for business communications. While the OTT players have “gotten by” through offering free or low-cost services, a service provider or an enterprise using these new communications technologies needs to account for their customer expectations and their reputation when managing the transition – a significant challenge. As a service provider’s or enterprise’s real-time traffic goes out into the Internet, it mixes with other traffic and is impacted by route changes and paths. The result is that a typical 30- to 60-minute VoIP call experiences multiple quality impacts and interruptions. And the issue is greatest in the Internet core and edge peering, where traffic is impacted most if sent randomly into the network. In fact, a recent PC World story said the most congested edges are in the U.S. and Europe, and are congested on all ports of open connections. Thus, it is clear that the challenge is managing the randomness and paths that traffic will take if left to the open Internet. Admittedly, this puts at risk the ability for WebRTC and VoIP to replace the PSTN/PPTN for some enterprise communications.
Optimizing the potential of WebRTC
So, how do we optimize the potential of WebRTC for opening up new services for consumers and enterprise users and more revenue for you as the provider, while also dealing with the challenges of QoS on the open Internet? In the late ’90s and early 2000s, content distribution on the Internet faced many of the same problems that real-time traffic faces today. That is, if a user visited a local website, the resulting browsing session had reasonable latency and responsiveness, but if the website happened to be located far away, the results became variable to awful. The challenge was that a website operator typically did not want to put instances of that website around the world. What emerged was a powerful concept: A content distribution network. By placing content distribution nodes close to the users and enabling those nodes to be shared among many website operators, the problem was solved. While the low-volume data might flow all the way back to the website, the high-volume traffic – images, files and video – were delivered from this shared resource locally. The result of this overlay content network was the globalization of the Internet for content. Akamai built out the first content network and changed the model of the Internet.
We could think of a similar enabler for real-time communications. For real-time Internet traffic, the same path is the best way to assure the delivery of VoIP traffic with a real-time overlay network optimized to deliver VoIP traffic with quality. Using WebRTC enables that network to be standard, open and global. WebRTC is an open international standard, meaning that solutions can be built by anyone that assure quality end-to-end experiences. Doing this requires local POPs in as many locations as possible, with as many Internet connections as possible. With such a solution, the challenges of variability and randomness can be limited to the shortest possible local path, just as the content distribution networks have done. This removes the causes of bad quality, without having to build QoS into all of the peering points on the Internet – a task that is many years from happening due to the lack of standards and operational challenges and undefined business models. By simple integration with the overlay real-time IP infrastructure for QoS guarantees, the problem is reduced even further, and, as this network is global, it can build the relationships for QoS on a broad basis. When contrasted with the challenge of each SP building its own network for the same goals, it is clear that the shared real-time IP overlay is the better option, just as the content networks were some years ago. And by building the overlay network with a standards based solution, it assures that edge peering can be accomplished rapid and globally.
WebRTC: the future of communications service delivery
Although it still has a long way to go, WebRTC will prove to be the tool for transforming communications. It can enable a “services anywhere” capability for service providers through underlying real-time communication edge networks, providing QoS and security on a global scale. This network will enable enterprises and service providers to deliver services to and from their customers anywhere in the world over the power of IP communications with PSTN-like reliability and availability. It is clear that the path is set and that enterprise communications will transform, and that all of us need to be ready to deliver solutions in this new paradigm where the PSTN is no longer the platform.
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