As the line between voice and data networks continues to blur, a new technology and terminology has surfaced -- SIP trunking.
If you know what VoIP is, you're halfway to understanding SIP trunking -- a frequently misunderstood technology layer that lowers telephony costs while helping enterprises seamlessly communicate.
The term VoIP is typically used to describe the physical equipment that enables Internet-routed voice calls. SIP trunking, however, describes the virtual layer that connects those devices and the larger telephony and data network "clouds." The result is an architecture that links phones, cell phones, mobile devices, desktops, laptops, and any other IP device by bridging the spaces seamlessly across digital networks and traditional PSTN networks between enterprise data networks, analog PSTN phone lines, and larger data networks (e.g., the generic Internet and proprietary carrier networks).
Technically speaking, SIP trunking uses the Session Initiation Protocol (SIP) and Real-time Transport Protocol (RTP) to deliver IP telephony and data traffic across multiple networks. It ensures that local VoIP equipment and IP-PBX switches can connect.
SIP trunking is an enterprise play, requiring robust infrastructure elements that make it secure and reliable for truly end-to-end IP communications. SIP trunking is usually installed by telecom vendors like AT&T, Global Crossing, Verizon, and others.
It's really SIP trunking that enables the grand vision for integrated voice, HD teleconferencing, IM, SMS, MMS, Twitter, IP collaboration tools, and whatever else comes along. The general term for this is Unified Communications (UC).
You can see Google doing this (somewhat) over the Internet with their Gmail, Buzz, Google Voice, Google Talk, Google Docs, and other collaborative applications.
The main reason enterprises caught on to VoIP and SIP trunking early is cost. Economic conditions collided with some technology advances, and early adopters started to rack up savings. Companies realized they were paying too much for legacy PBX equipment, and they noticed the merger of IT and telephony skill sets within their walls. As IT budgets declined or were frozen, the case for all-digital became much more compelling.
One of Gartner Inc. 's latest studies showed 30 percent to 50 percent savings on voice and data connectivity costs for midsized and large enterprises using SIP trunking. That cost reduction comes from a few aspects of the technique:
End-to-end IP transmissions reduce costs to service providers and their customers.
SIP trunking terminates calls to the PSTN without the need for customer premise gateway equipment. There's no need to manage and maintain complexities there.
Peer-to-peer or "on-network" calls that traverse a company's VPN incur no charges (even globally). It's like sending email over the Internet.
There's no need for separate broadband Internet and PRI (Primary Rate Interface) trunk lines.
Voice and data network consolidation lowers the cost of connectivity to the service provider network.
On an end-to-end IP network, there's no difference between interstate and intrastate calling.
In addition, analyst groups are looking closely at how productivity savings from UC will impact budgets and revenue even further. As collaboration and messaging tools become easier to use, people should work more efficiently and communicate more effectively. Companies like Cisco Systems Inc. (Nasdaq: CSCO) (Unified Communications Manager), IBM Corp. (NYSE: IBM)(Converged Communications Services), and Microsoft Corp. (Nasdaq: MSFT) (Office Communications Server) are already trumpeting the value of this communications shift.
One last note: SIP trunking allows companies to keep legacy equipment and transition to all-IP incrementally. This fact alone has gotten the attention of many large enterprises with significant analog telecom equipment investments.
Whether you look at it from a cost or productivity perspective, it's clear that SIP trunking and UC will continue to transform communication in the decade to come.
— Phil Dunn is a journalist, technology consultant, and marketing writer specializing in IT.
where I work we use the Ozeki Phone System XE (http://www.ozekiphone.com), that supports SRTP and other secure ways of telecommunication. We are satisfied with the service, there has never been any successful attacks against us.
You are both very welcome! Telecommunications is a big part of my life (8:00am-4:00pm, Mon-Fri), and it certainly is enjoyable to be able to discuss the VoIP side of things with like-minded people on InternetEvolution. Keep the discussions coming and I shall do the same. :-)
You are both very welcome! Telecommunications is a big part of my life (8:00am-4:00pm, Mon-Fri), and it certainly is enjoyable to be able to discuss the VoIP side of things with like-minded people on InternetEvolution. Keep the discussions coming, and I shall do the same. :-)
Thanks for the expanded reply, Nathan... seems like we're headed to encrypted everything, especially if all this cloud hype is true (or maybe I just drank too much of the Kool-Aide at RSA this week).
"One last note: SIP trunking allows companies to keep legacy equipment and transition to all-IP incrementally. This fact alone has gotten the attention of many large enterprises with significant analog telecom equipment investments."
Great question. I've worked in the VoIP/telecom industry since 2005 (and have studied telecommunications as a hobby - mostly TDM networks, mobile, pager networks since 1997).
To answer your question; yes, SRTP is supported by some carriers, though to be perfectly honest with you not too many do. If you're looking for a decent carrier (many will be resellers of CLECs or resellers of resellers of CLECs), keep your eyes peeled for VoIP SIP-providers that support these as well:
SIPS, MIKEY, SDES, and IPSec.
Currently my employer supports IPSec encryption for SIP signalling and the RTP.
--
Miscellaneous:
For anyone looking to setting up their own PBX (Private Branch Exchange - think: that IVR system that you reach when you call any major business which includes voicemail, call forwarding, call waiting, and in some cases conferences services), then I highly recommend taking a look at these:
FreeSwitch (highly recommended), TrixBox (based upon Asterisk), FreePBX, AsteriskNOW.
If you're someone who wants control over packet routing engine with SIP and you plan to implement it on a carrier-grade VoIP Switch, check out OpenSIPS.
Though any soft phone (I recommend X-Lite and SJPhone) or device (Hardphone, ATA) will work fine so long as they support the SIP protocol. Try to stay away from the IAX2 protocol by Digium (creators of Asterisk) if you can; it has its stability/memory issues after about 100 calls... and is only recommended for small business users. The SIP implementation on Asterisk is fine and is more flexible for a variety of platforms.
Thanks for joining the conversation, Terry. Secure RTTP is virgin territory for me. I haven't taken a dive into that tech pool yet. I'm guessing (just guessing) that algorithm layers and authentication/encryption take a bite out of performance... but, with an optimized set-up (e.g. dedicated channels, QoS etc), that effect could be eliminated.
Thanks for your interest, Chris. When you talk about providers, there are really three layers to SIP trunking. 1) infrastructure (companies like Qwest, Global Crossing, AT&T and Verizon offer MPLS networks that are end-to-end IP), 2) hardware manufacturers that make IP-PBX equipment (like 3COM, Cisco, Siemens, Nortel, Avaya), then 3) the user interface co's mentioned in the article (Microsoft OCS and Cisco UCM).
There are plenty of other "piggy-back" and/or parallel service providers and manufacturers playing in this space, as well. It's evolving rapidly.
The fact that expensive/complex gateways are not required is notable. End-to-end IP brings the cloud or client/server metaphor to telephony. It's like a managed service rather than a premise-based hassle. Once you're cleanly connected to the infrastructure, the software becomes much more important (in terms of matching utility with user demands).
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