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Unified Communications Magazine May 2008
Volume 1 / Number 6
Unified Communications Magazine
Jonathan Rosenberg

Pondering SIP Trunks

It seems I have a habit of picking marketing buzz words for my columns in this magazine. Web 2.0, P2P, and fixed-mobile convergence have all been topics I have written about. This article adds another to the list: SIP trunking.

By Jonathan Rosenberg, Speaking SIP

A SIP trunk refers to a service that replaces traditional PRI-based access with SIP-based access in enterprises. A number of service providers in the U.S. have begun to test-run and offer SIP trunk services.

The basic mechanism is simple. Typically, the service provider offers an IP-based interconnection into the data center of the enterprise. This is not public IP access, but rather a quality-of-service (QoS)-managed IP link. The enterprise and service provider both typically deploy a session border controller, with traffic (both voice and SIP signaling) running between them over the managed IP link. There are variations on the SIP interface, depending on the provider. Some conform to existing SIP profiles (such as the SIP Forum’s SIPconnect specification), while others define their own profiles and extensions.

The enterprise can now move its PSTN interconnects, which are often distributed over many branches, into a centralized location in the data center. Larger enterprises with reliable WANs can now enjoy both an operational cost savings and a capital expense reduction, since fewer TDM line cards are needed. Oftentimes, moving from TDM to SIP is accompanied by a direct cost reduction in the service itself, as the service providers may charge less for calls over the SIP trunk than over the TDM trunk.

Even so, my largest worry is that SIP trunking will forever remain what it is today – a pure replacement for PRI, providing as much (and as little) as PRI offers. As more and more enterprises and consumers move to IP, we have an opportunity for an end-to-end SIP connection.

Might SIP trunk services evolve to go beyond PRI replacement and enable all that SIP has to offer? They might, but there are challenges. Clearly, these additional SIP features become enabled only when the call can be SIP end-to-end. Unfortunately, even if the call is IP all the way, this does not imply a SIP “clear channel”. A SIP clear channel happens when the two enterprises, should their own networks support a particular SIP feature, are able to realize that feature in the presence of service provider intermediaries that are unaware of, or have no support for, that same feature. Some service providers carefully filter SIP signaling for many reasons, resulting in a loss of this clear channel.

Unfortunately, the problem gets worse as the service providers themselves begin to interconnect with each other over SIP. Though there will always be cases where just one provider is in the path, and cases where features do work in the presence of multiple providers, they are just that – cases. Enterprises cannot depend on these features working; they sometimes will, and sometimes won’t. Consequently, systems will be designed around the lowest common denominator: basic voice. If all we ever get between enterprises is basic voice, it will be a sad future for our industry.

This lack of a “SIP clear channel” is only an issue for calls between IP-enabled enterprises or consumers. For calls terminating on the PSTN, a SIP clear channel is not needed. However, there are other problems. There is no shortage of SIP trunk interface specifications, and different providers use different ones. Enterprises are thus burdened with testing and building interconnections to each provider separately, a potentially painful process. Another concern is that, should SIP trunking take off, its costs may not be radically lower than PRI interconnection. As a point of speculation, what would motivate providers to substantially undercut their own existing TDM business?

Put together, SIP trunks offer some attractive benefits but come with some potential risks downstream. Caution is warranted before putting those trunks into the stream and sipping from what they have to offer.

Jonathan Rosenberg is the co-author of SIP and SIMPLE. He is currently a Cisco Fellow and architect for the IP Communications Business Unit in the Voice Technology Group at Cisco (www.cisco.com).

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