Calling on SIP: How it’s Transforming Communications

October 13, 2014

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Once the domain of voice and voice only, telecommunications has recently undergone a remarkable transformation. Thanks to convergence, the telephone network and the global Internet are rapidly becoming one, and the protocols that govern the Internet, including the Session Initiation Protocol (SIP), have breathed new life into the incumbent network. In this Podcast we explore the role of SIP, SIP Trunking and how it is transforming service delivery.


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(Episode 101)
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Transcript

The New Network Podcast
Calling on SIP: How Session initiation protocol (SIP)  is Transforming Business Communications

Steve Shepard

One of the benefits that IT and media technology achievements have brought to the market is the consistant improvements in performance due to increased efficiency and scalable network-centric solutions. Because of new coding techniques, compression capabilities, and hardware and software innovations, applications are much more efficient than they ever were before. Voice and video, for example, can be carried over significantly smaller channels with virtually zero loss of perceived quality.

There is a challenge however: legacy communication technologies like time division multiplexing aren’t flexible and scalable enough to dole out bandwidth in as granular a fashion as today’s applications models require, particularly given how media and video- centric they are. The move to IP, which is well underway, will add a great deal of flexibility, but with the move to IP comes the need for a much more capable and flexible signaling protocol for establishing sessions between communicating devices. Luckily, there’s a solution. It’s called the Session Initiation Protocol, or SIP, and it’s changing the way services are delivered over today’s advanced networks.

The intense interest in moving voice to an IP infrastructure is driven really by two simple and understandable factors. The first is the cost of delivered service and the need for enhanced network flexibility. However, in keeping with the “Jurassic Park effect”—just because you can, doesn’t necessarily mean you should—it is critical to understand the differences that exist between simple voice and full-blown telephony with its many enhanced features. It is the feature set that gives voice its range of capabilities: a typical legacy switch such as Alcatel Lucent’s 5ESS offers more than 3,000 features. Of course, the features and services are possible because of the protocols that have been developed to provide them across the telephony infrastructure, and SIP’s job is to make sure that needed features are available to the IP user community.

And of course the second driver, which is a really a subset of the first, is that the marketplace is demanding a much broader and richer experience in their use of telecommunications services.

So exactly what is SIP—and why should you care? Well, to answer that question, let’s start with what happens when you make a phone call.

[Sound of dialing, ringing, hello?]

Sounds simple, doesn’t it? But behind the scenes there is a complex chaos of activity going on that the calling parties never see. The network has to determine whether the number you’re calling is local or long distance, domestic or international, a fixed line or mobile; it has to check a database to ensure that you receive whatever supplemental services you are entitled to, like caller ID and distinctive ringing; and of course it must reserve network resources to ensure that your call can actually be connected. And, it has to monitor the call so that it can free up the resources when you’re finished and bill you appropriately for whatever resources you used. Needless to say, this is a very, very complicated process—and we’re just talking about a telephone call!

Now imagine a much simpler environment, thanks to the arrival of IP telephony and the various protocols (now you should know that protocols are the rules we referred to earlier) that make it all work. One of those protocols is SIP, and its job is to establish, modify as required, and then terminate sessions when they are placed across an all-IP network, and that’s where more and more business customers are migrating today. The truth is that IP as a fundamental protocol for telecommunications is not really an if question today, it’s a when question.

Now SIP, which is part of the IP family of protocols, is designed to establish logical connections between communicating devices. The protocol recognizes a variety of server types including the feature servers that manage the add-on capabilities that callers sometimes require; registration servers, which manage IP addresses so that called parties can be found; and redirect servers, which handle calls between domains— similar to a long distance switch in the legacy telephony world. And because SIP is based on the protocols that govern the global Internet, it provides a seamless integration path for the integration of voice and data.

The power behind SIP is that it works for many more applications than just voice. It can be used to set up point-to-point and point-to-multipoint videoconferences, streaming media sessions, file transfers, instant messaging, IP fax transmissions, and online gaming—not to mention full-featured voice phone calls. But it’s actually much more than that.

One shift that has taken place, thanks to the growing capability of cloud services, is the migration of content and applications from the user’s device to the cloud. Those of you who have been in the industry for a while will recall the days of the so-called dumb terminal, an intelligence-free display device (hence the name, dumb terminal) and a centralized mainframe computer (for all intents and purposes, a cloud resource), connected together over a low-speed facility. All of the intelligence in those early systems was concentrated in the mainframe, leaving the device to serve as a display.

Today, a similar thing is happening in the mobile world of laptops, tablets, and phones. Applications are moving into the cloud where they execute; a small client replaces them on the mobile device. This frees up real estate on the mobile device that can be used for other functions—expanded memory, a more complex camera (or in some cases, multiple cameras), a larger battery, more radio technology. And as these devices become more multifunctional, a need arises for a control protocol that allows all of those separate functions to deliver everything they’re capable of to the user. SIP plays a major role in this regard.

A second piece of this evolutionary puzzle that is catalyzing the role of SIP is unified communications, sometimes called UC. Unified communications assumes a common IP protocol platform and includes unified voicemail, fax and e-mail; IP telephony; presence; web conferencing; document sharing; and cross-platform application execution.

IP Telephony and UC are growing at a rapid pace. According to consultancy Frost & Sullivan, IP communications systems accounted for 88.7 percent of all enterprise line licenses shipped in 2013, with the remainder being legacy TDM-system line licenses. Furthermore, the installed base of activated PC communications soft clients is expected to more than triple over the next five years, signaling the growing adoption of UC technologies.

SIP extends the capabilities of IP to include signaling for both wireline and wireless networks, thus bridging the gap that exists between different networks, PBXs, and a host of end-user devices. It also creates a “whole is greater than the sum of its parts” reality. Because SIP can control all kinds of media-based sessions, multiple media can be created that seamlessly blend together—voice, video and other information presentation modalities—into a single, rich, converged experience. For example, a two-person voice call can easily be transformed into a multipoint call, then into a video call. A fax can be sent from within the session, as can e-mail. One user can search for something on the Internet and display it for all to see, thanks to the multimodal capabilities of SIP.

SIP trunks are an evolutionary benefit of SIP’s growing influence. There’s nothing magic about a SIP trunk; they’re virtual circuits that are provisioned across a service provider’s private IP backbone to connect to an organization’s IP PBX and LAN. They’re most commonly used as a replacement technology for the ISDN PRIs that typically interconnect a PBX to the wider area global network. But they have another advantage as well: they can provide tremendous reliability and fallback capability in the event of a network failure. For example, in the event of a failure of the legacy network due to natural disaster, SIP trunks can redirect calls to any available medium including wireline, wireless, the PSTN, or a public IP network, drastically reducing the impact of the failure and restoring service in minutes or hours rather than days.

Let me give you an example. In 1989 I was an employee of the phone company that served the state of California at the time. On October 17 of that year at precisely 5:04 in the afternoon, I was walking to my car when the 8.3-magnitude Loma Prieta earthquake struck the Bay Area. That was an epic tragedy on so many levels, and the damage it did to the communications infrastructure was extraordinary. Among other things, very high bandwidth microwave facilities were knocked out of kilter and ceased to function. Luckily, the design of the network included redundant paths, so communications was restored within a few days. Had SIP been in place and the Internet and IP networking as widely deployed as it is today, more virtual alternate routing paths might have been established to lessen the impact of a major disaster.

Another area where SIP trunking can have a positive impact on the contact center is in the world of cost. By replacing T1 or PRI connections into the network with SIP trunks and routing calls across the Internet (or a private IP network), toll charges and recurring monthly costs can be drastically reduced. For contact center environments that span multiple continents, this Internet-based calling model can result in dramatic cost savings without unduly sacrificing call quality.

Like TCP/IP, the fundamental protocol upon which the Internet is built, SIP provides an open architecture that can be used by literally any vendor to develop products, thus enabling multivendor interoperability. And because SIP has been adopted by all of the major equipment manufacturers and is designed for use in large carrier networks with potentially millions of ports, its success is essentially assured.

The evolution of the network from a traditional time division multiplexed architecture to IP, as I said earlier, is not an if question—it’s a when question. And, according to Infonetics Research, business adoption of SIP trunks is forecasted to increase 52 percent between 2013 to 2015. IP and all the protocols associated with it, including SIP, represent not only a major technological shift in networking, but a radical forward-looking shift in the way enterprises do business, thanks in large part to the enhanced capabilities of SIP. This technology is a game changer.

Thanks for listening, folks. For Time Warner Cable Business Class, I’m Steve Shepard.

 

 

 


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