A handy-dandy SIP primer to help enterprises understand what a SIP migration entails and the benefits you can hope to gain
If you’re going through hell, keep going.
— Winston Churchill
I am sure we’ve all fallen into this trap. You know something so well that not only do you expect everyone else to know it just was well, but you speak about it in words that only an expert would understand. I certainly do that when I speak about old rock and roll with my wife. “How could you not know who played bass for the Jefferson Airplane?”
I also sometimes do it with SIP. I have been working with this technology for so long that I am shocked when I find that not everyone is immersed in it as I have been. “What? You haven’t already converted your ISDN trunks to SIP and downloaded a SIP client to your iPhone? Have you upgraded your house to electricity, yet?”
All kidding aside, I am well aware that converting all those ISDN lines to SIP is not a spur-of-the-moment decision. It requires analysis, a compelling ROI story, careful planning, and a dedicated team of professionals to make the switch. Thankfully, enterprises have been putting in the effort, and while there may have been a few hiccups along the way, they are successfully moving their voice communications into the 21st century.
However, many enterprises are still on the fence and require a little education (and perhaps some prodding). Here is a handy-dandy SIP primer that will help you understand what you are doing and hopefully why you are doing it.
The Session Initiation Protocol (SIP) is a signaling protocol used to establish, modify, and tear-down communication sessions in an IP network. These sessions can be as simple as a two-way call or as involved as a multi-party Web conference complete with audio, video, and a shared whiteboard application.
SIP was modeled after the Hypertext Transfer Protocol (HTTP) and contains many of the basic tenets of that protocol. First, SIP is an English-like, text-based protocol that is not only easy to read, but is also easy to understand, debug, and extend. New features can be added to SIP without the need to modify any of the SIP server entities that might exist within any particular call path.
Second, and perhaps most importantly, SIP is media agnostic. In other words, SIP can be used to establish sessions of nearly any media type imaginable. As communications move well beyond that of a simple phone call, SIP is fully equipped to support any and all media (voice, video, instant message, text, etc.) that might come along.
Additionally, SIP has been extended to allow for first- and third-party call control, which means that SIP can be used by one entity to control the call flow of another entity. In its most basic sense, SIP enables applications to be written that direct endpoints to create (e.g. make a call), manage (e.g. answer an incoming call), and terminate sessions (e.g. release call).
In the same way that HTTP allows a Web browser to deliver a wide variety of content types to a PC or other Web-enabled device, SIP-enabled devices can support media from many different sources. Built into SIP is the notion of session description, which allows SIP to establish a session independent of the underlying media stream. This enables session escalation whereby a user might start communicating with an instant message and then later on add voice, file transfer, and multi-party video. SIP has been designed to support any form of communication that a user may require.
SIP is perfect for dealing with the explosion of consumer-grade communications devices that are making their way into the enterprise. Imagine a world where your personal iPhone or iPad can be securely integrated into your communications system. With SIP that world not only exists, but solutions are available today.
In traditional wireline telephony, phone calls are passed to and from an enterprise and the Public Switched Telephone Network (PSTN) over a dedicated line or bundle of circuits. These could be analog trunks such as loop or ground start lines, or digital trunks such as T1, E1, ISDN, or PRI. Since SIP is an IP protocol, it runs on the same network that data traffic runs on. This convergence of voice and data means that a SIP trunk is a logical concept that has more to do with bandwidth than physical wires or circuits.
The benefits of SIP trunks over traditional trunks are many:
Converged voice and data
Equipment reduction which leads to reduced power and space requirements
Flexible costs due to burst pricing
Improved reliability and failover strategies
Computer Telephony Integration (CTI) has traditionally been endpoint-centric, where applications controlled and monitored physical endpoints regardless of who might be using that device. However, with the explosion of communications interfaces, a user might employ numerous devices throughout the day. For instance, the manager of a sales department will typically have an office phone, a cell phone, a soft phone, and an instant messaging client.
With SIP, a user’s communications activities, along with the presence updates generated by those devices (e.g. “on a conference call,” “do not disturb,” etc.), can be managed as a whole. This user-centric model is a break from the device-centric model where each device is treated as a separate entity with no particular connection to its owner.
In the same way that you would never allow a PC to connect to the Internet without the proper security tools such as a firewall and virus checker, voice over IP (VoIP) requires protection from malicious activity. SIP has a number of security mechanisms that are either built into it or work alongside it to create a rock-solid means of defense. For example, SIP itself can be encrypted, and individual SIP messages can be challenged with authentication requests. SIP media streams can also be encrypted to prevent prying eyes and ears.
Finally, SIP-based components such as a session border controllers (SBC) can be deployed as a perimeter defense appliance similar to how an enterprise would deploy a network firewall.