In our three previous tutorials, we have considered PBX trunks, the SIP Forum’s SIPconnect architecture, and the technical details of the SIPconnect interface specification. In this, our final look at the SIP trunking story, we will look at the breadth of industry support for this emerging technology.
Recall from our earlier discussions that SIP trunking defines an interface between two entities: an IP PBX and a service provider. These connections converge the voice and data service into a single Internet Protocol (IP)-based pipe from the IP PBX to the service provider, with that service provider taking on the responsibility for terminating any calls that need to go to the Public Switched Telephone Network (PSTN). In essence, the connectivity responsibility is shifted from the enterprise to the carrier—after the call gets through the IP pipe to the service provider, that provider becomes responsible for the balance of the connection.
With that single pipe come a number of benefits. These include:
- eliminating the need for hardware-based PSTN gateways
- improving voice quality, since the call stays IP from the enterprise to the carrier
- supporting the integration of desktop and communications applications—such as unified messaging and corporate directories, which can be very beneficial to contact centers
- easing the administration overhead on the telephone part of the enterprise network, since the moves, adds and changes with a SIP-based system are typically easier to configure than with a traditional PBX
- reducing the costs of international calls, since all or part of that call may travel over the Internet or via private IP network instead of the PSTN
Furthermore, SIP trunks can provide an inbound calling service that is independent of the caller’s location. Let us suppose that your company is planning to expand into a new market, but does not have an office or even a physical presence in that new city as yet. Nevertheless, it would be beneficial to have the appearance of an office in that location, even though no one is there to turn on the lights or open the door. Many service providers can provision toll-free direct inward dialing (DID) service that allows their customers to create a virtual presence in a city where they do not yet want to invest in a physical communications system, but still want to reach their customers via a toll-free (local) call. This same capability can also benefit telecommuters and road warriors as well.
So let’s assume that you are convinced of the benefits of SIP trunking; perhaps the next question to address is “who supports SIP trunking?” As it turns out, the list is fairly long. Support on the IP PBX side comes from companies including Allworx, Avaya , Cisco Systems , Digium (Asterisk), Epygi, Mitel, Siemens, and Switchvox.
Notable from the above is Avaya, with their DeveloperConnection program, which unites application developers, product developers, and service providers to create innovative solutions for their customers (see http://www.avaya.com/gcm/master-usa/en-us/corporate/alliances/developerconnection/index.htm). Members gain access to the Application Program Interfaces (APIs) and Software Development Kits (SDKs) that support the Avaya products, and they may participate in joint marketing and product support efforts. Perhaps of greatest interest to enterprise managers are the Application Notes that are available from the DeveloperConnection, published by Avaya’s Solution and Interoperability Test Lab. To spark your interest, here’s one title:
Application Notes for Configuring SIP Trunking between the Global Crossing VoIP Network and an Avaya IP Telephony Network—Issue 1.0.
The document’s abstract goes even further:
Information in these Application Notes has been obtained through compliance testing and additional technical discussions. Test was conducted via the DeveloperConnection Service Provider Program at the Avaya Solution and Interoperability Test Lab.
And Global Crossing isn’t the only one—Avaya has also published notes for AGN Networks and Verizon Business services, plus a host of other hardware configurations as well. Peek inside this app note and you will find details on the specific carrier services that were used, the Avaya hardware and its configuration, any additional hardware that was included in the evaluation, plus the call scenarios that were tested. Perhaps most importantly, the document goes into configuration details including options and parameters, illustrated with a number of screen shots and other visuals.
So what can we conclude from our study of SIP trunking? First, this technology is garnering quite a bit of interest, on both the equipment developer and service provider sides of the fence, and those that are interested are among the heaviest hitters in this industry. Second, the technology is mature enough to have spawned the SIP Forum, and more specifically the SIPconnect interface standard. Finally, these heavy hitters are banding together to perform interoperability testing, and then publishing applications notes to assist their customers with making everything play together. So do you get their drift? If you are planning to implement SIP trunking, a few days of homework should appear on your calendar!
This concludes our tour of SIP trunking. Our next tutorial will look at another piece of the VoIP peering puzzle—the session border controller.
Copyright Acknowledgement: © 2007 DigiNet Corporation ®, All Rights Reserved
Mark A. Miller, P.E. is President of DigiNet Corporation®, a Denver-based consulting engineering firm. He is the author of many books on networking technologies, including Voice over IP Technologies, and Internet Technologies Handbook, both published by John Wiley & Sons.