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SIPconnect and the Rationale for Widespread Adoption of SIP TrunkingSIPconnect and the Rationale for Widespread Adoption of SIP Trunking

SIP trunks could save money and deliver better feature/functionality to IP telephony. But there have been obstacles to overcome.

July 22, 2008

10 Min Read
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IP PBXs are starting to dominate the enterprise equipment market. But TDM trunking remains the norm, unnecessarily limiting the advantages business customers can realize from IP voice in both capabilities and cost.

The answer to this dilemma is still relatively little used. End-to-end IP voice "peering" between those customers and service providers is urgently needed to release the capabilities of increasingly ubiquitous VoIP equipment from the constraints of TDM trunking.

IP PBX sales in North America, estimated at over $3 billion (including IP-enabled lines) by analyst firm Gartner, now well exceed those of traditional TDM-based PBXs. Research firm Dell' Oro estimates US IP PBX annual sales are tripling in value from 2005 to 2010. A majority of installed PBXs are now at least IP-enabled.

Today's VoIP communications systems offer a wealth of advanced features with the ability to readily add new ones as demands evolve. Service providers have meanwhile taken big steps to transform their networks with VoIP technology to deploy new services while heightening efficiency.

But business customers, by still using trunking gateways for their carrier links, are held back by traditional TDM technology. Employing PRI or analog connections to provider networks limits available features to the "lowest common denominator" those connections will support. Connection quality is meanwhile downgraded when converting traffic back and forth between IP and TDM.

The SIP Forum--an IP communications industry association--is trying to address these problems with SIPconnect, a technical recommendation designed to facilitate direct "peering" between SIP-enabled networks of service providers and enterprise-based IP PBXs.

While manufacturers and service providers have largely settled on Session Initiation Protocol (SIP) as the way to realize full IP connectivity, the protocol by itself does not resolve the dilemma.

Under SIP, there are often many ways to achieve the same interconnection tasks, complicating interoperability. VOIP network interconnection involves issues beyond signaling, such as security, for example, which must be addressed to fully define a predictable interface model, and this requires building further on SIP.

SIPconnect

SIPconnect is a recommended set of interoperability guidelines that the SIP Forum is driving to establish as industry norms; it is not a new protocol. It. The SIPconnect Technical Recommendation defines a set of rules for seamless interconnection between SIP-enabled IP PBXs and SIP-enabled service providers, specifying required VoIP protocols and features to be supported, a reference architecture (see Figure 1, below) and implementation rules when protocols leave multiple options.

SIPconnect was conceived by Cbeyond in 2005 with the support of technology vendors. It was later turned over to the SIP Forum--the industry organization focused specifically on adoption of SIP-based products and services--for wider review and development, and an updated, improved version resulted.

In January, 2008, the SIP Forum formally ratified version 1.0 of the SIPconnect Technical Recommendation, validating that the proposal had survived credible and lengthy peer review, is stable and well-understood, and is believed to have resolved known design issues. To download the ratified SIPconnect Technical Recommendation Version 1.0, click here. At the same time, the SIP Forum Board of Directors announced formation of the SIPconnect v.1.1 Task Group to further enhance and update the recommendation. For an overview of this work in progress, please click here.

The reference diagram (Figure 1) outlines common functional elements required to support SIPconnect. SIPconnect treats the elements in the diagram as separate physical components for illustration only, as equipment manufacturers may combine functions in single physical devices.

For example, one vendor may integrate the SIP Proxy Server function with the IP PBX function, while another may integrate these two functions with the Firewall function as well. Both, as well as other combinations, are conformant as long as they adhere to the specific rules governing each integrated function.



FIGURE 1: SIPconnect Architecture



IP TRANSITION: THE CURRENT SITUATION

The growth of IP-based customer premise equipment (CPE) has helped businesses upgrade and converge network infrastructures and deploy exciting new IP-based features and capabilities while saving on recurring charges for analog lines and PSTN transport.

VoIP-based services (including IP Centrex and Hosted IP PBX) and equipment offer a range of new features otherwise unavailable, including voice-data integration, simple Web-based system management and desktop integration for presence-based features.

VoIP further allows service providers to make available new kinds of capabilities such as wireless-wireline integration, click-to-dial, teleworker/remote office applications and softphone support. To extend these next-generation carrier offerings most efficiently and seamlessly over the wide area to remote points, providers need to be able to directly connect, or "peer" over networks to IP PBXs at distant customer sites.

In a typical business network, the IP PBX serves as the interface to the LAN, enabling IP phones, PCs, conferencing devices, wireless equipment and other communications endpoints. At the same time, it also interfaces to the PSTN, enabling essential conversion of IP packets to traditional analog or digital signals and the reverse. This has typically been accomplished through adjunct IP telephony gateways, either at customer premises or in provider networks (see Figure 2, below).



FIGURE 2: A Typical Customer Premises Setup using an IP PBX and VoIP Gateway

But this packet conversion to TDM introduces delay (latency) that often degrades voice quality. Advanced IP-based signaling information and features meanwhile are often also stripped from the transmission by conversion, eroding the ability to deliver IP-based features. The bottom line: TDM routing of VoIP traffic is clearly an inferior, "Band-Aid" approach to next-generation communications that fails to support full VoIP capabilities.

Enabling IP PBXs rather to connect directly with VoIP service providers, eliminating the need for gateways and TDM traffic routing, is a far better approach, enabling the full capabilities of packet-based communications. To accomplish this peering, however, equipment and service providers must use common standards. Hence: SIPconnect.

While Session Initiation Protocol (SIP) adoption enables direct packet peering between compliant IP PBXs and compliant VoIP service providers, SIPconnect offers a well-defined method for applying SIP to connecting provider networks with IP PBXs. Figure 3, below, illustrates direct IP peering.



FIGURE 3: Direct IP Peering between IP PBX and VoIP Service Provider



SIP AND ITS DISCONTENTS

SIP is a text-based protocol designed to start, modify and end interactive communication sessions using voice, video, messaging or other multimedia applications. Its standards are overseen by an Internet Engineering Task Force (IETF) working group. SIP covers signaling, location and registration, allowing support for other features through separate protocols, making it "lighter" and more efficient than its early rival H.323, and one decentralized within the network--pushing intelligence to phones and other end devices. While H.323 was widely deployed as a protocol in the early days of IP telephony, SIP has largely replaced it in this decade.

While SIP is now almost universally considered the preferred means for routing IP traffic, its own gaps have been widely noted. The protocol defines multiple methods for interconnecting, which complicates interoperability. SIP fails to inherently resolve issues beyond signaling. And it has problems in managing hierarchical logical identities, a requirement for effective peering.

Thus, additional consensus on accomplishing interconnection with SIP is required--most significantly, a specification that chooses when multiple options are available. The SIP Forum has taken, as its primary work, definition and refinement of a peering architecture to link IP PBXs to service providers. The SIP Forum has also created a "SIPconnect Compliant Program" allowing hardware and software vendors as well as service providers to demonstrate their compliance with the SIPconnect Technical Recommendation and gain certification.

BENEFITS OF DIRECT IP PEERING

Virtually all major IP PBX players support SIP. But many vendors don't appear to have yet given full attention to the benefits of direct IP peering. Support by a product of an industry-accepted standard that successfully addresses quality and security issues reduces equipment and transport costs, expands availability of features and functionality, and eliminates time and cost needed to set up proprietary interfaces, offering a significant competitive edge.

Such peering meanwhile helps service providers offer higher-quality services with advanced features specifically tailored to IP PBX users. The ability to develop relationships with IP PBX vendors based on this interconnectivity can also help win customers and establish new relationships in various distribution channels, including interconnects, VARs and system integrators.

Business customers, the ultimate beneficiaries of IP peering, seek feature-rich yet affordable communication systems. Many have hesitated to deploy VoIP because of quality issues and other concerns with the emerging technology.

Many advanced features meanwhile enabled by IP-based systems--presence, video conferencing, one-click dialing--may not be enough to persuade customers to give up their old systems. Typically more urgent reasons are needed to make customers change. Direct peering can make the transition to IP more of a "slam dunk."

Businesses peering directly with providers obviate their need for costly TDM gateways and increase the efficiency with which they use local access lines. Another easily overlooked feature enabled through VoIP-based systems is the ability to provide Direct Inward Dialing (DID) capabilities for less cost. While many small businesses can't afford the full T1 or PRI line typically used for DID, direct IP peering lets VoIP providers offer many direct phone numbers on a single connection.

This means a small customer can use multiple "numbers" without paying recurring charges for separate analog lines or costly digital circuits. The ability to offer discrete “DID” numbers to individual employees within a small business--in preference to extensions off a main number--is a major benefit taken for granted by larger businesses.

Presence-based applications and other enhanced features remain an added bonus of direct IP peering, enabling carriage of end-user information intact to other VoIP-enabled destinations across networks.

IP peering meanwhile also benefits distributors and other channel partners. In response to quality of service problems arising from VoIP to TDM conversions, these players often need to perform costly, complex custom configurations for individual customers. Direct peering offers relief from these burdens, making gateways unnecessary and eliminating much time-consuming configuration and troubleshooting. Direct IP peering also allows security-related functions to be "off-loaded" to the VoIP provider network.

SIPconnect offers, in sum:

  • A Universal Approach: Clear, mutually agreed guidelines, to accelerate adoption and reduce development costs;

  • Customer Cost Saving: Obviating need for gateways and extending benefits and savings of IP systems;

  • Transparent Feature Transport: End-user information is passed on intact, not stripped out, enabling IP features and capabilities to all connection points;

  • Quality of Service: Defining key transport layer issues; and

  • Improved Security.

    BEHIND THE CURTAIN: THE SIP FORUM

    The SIP Forum is a non-profit industry organization whose members represent more than 45 IP communications companies and thousands of individuals from around the world. Its mission is to advance adoption of SIP-based products and services. The SIP Forum directs technical activities to achieve enhanced product interoperability, provides information on SIP benefits, and highlights successful applications and deployments. The SIP Forum is also the key "meeting place" for developers of commercial SIP-based services and related Internet technology, including IP phones, SIP servers, IP telephony gateways and PC clients.

    The SIP Forum seeks to facilitate SIP integration with other Internet-based technology developments including, for example, security, QoS and wireless internetworking. The SIP Forum is explicitly not, however, a standards-setting body, as the IETF continues to define the core SIP protocol.

    The SIP Forum is open to anyone accepting the architectural model on which SIP relies and willing to contribute to disseminating information about SIP. Individual membership is free. Organizational members pay an annual fee to cover Forum administrative costs. Membership has grown over 100% in 2008 and includes over 5,000 individuals. Organizational members include, for example, 3Com, Alcatel-Lucent, Avaya, Broadsoft, Dialogic, Ericsson, IBM, HP, Microsoft, Oracle, Siemens and Wipro as well as service providers Bandwidth.com, Cox Communications, Cbeyond and McLeod. Visit www.sipforum.org for more information.

    Marc Robins is managing director of the SIP Forum.