IP Telephony

Contact Centers

Mobility

WHITE PAPER

Services

Delivering on the Promise of SIP Networking

June 2005

Enterprise SIP Trunking:

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Table of Contents

Executive Summary ........................................................................................................... 1

Introduction ........................................................................................................................ 1

SIP Networking Across the WAN ..................................................................................... 2

Service Providers Are Rebuilding Their Networks Around SIP ................................... 4

Exploring the Benefits of SIP Networking for the Enterprise ...................................... 5

Phase 1: Traditional Networking Model Using PSTN .................................................................... 5

Phase 2: Transition to IP Telephony Today .................................................................................. 6

Phase 3: VoIP Transit Using SIP ................................................................................................ 7

Phase 4: Future Path Incorporating Universal Directory Services .................................................. 9

Call Center case study .............................................................................................................. 9

Avaya: Building the SIP-Connectivity Ecosystem ........................................................ 11

SIP Trunking ......................................................................................................................... 13

Instant Messaging (IM)/Presence ............................................................................................. 13

SIP Telephony ....................................................................................................................... 13

User Mobility ......................................................................................................................... 13

SIP Applications .................................................................................................................... 13

Conclusions ....................................................................................................................... 14

Footnotes .............................................................................................................................. 14

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Executive Summary

Converged networks have been evolving since their initial deployments, with simple Voice over IP transport services giving way to next-generation networks supporting intelligent multimedia communications. The focus of the enter-prise has evolved as well, moving from simple cost savings towards embedding communications into business pro-cesses, improving worker productivity, and enhancing customer interactions. Yet even the most advanced enterprises have typically limited their IP Telephony deployments to their own intranets, and not across the public network. Multiple integration challenges have prevented the full realization of end-to-end IP telephony communications.

However, solutions are emerging that will reduce these challenges, and at the same time deliver an additional wave of financial and operational benefits. New solutions are possible due to a number of factors including:

• IP Telephony is reaching a critical mass within enterprise networks.

• Service Providers are leveraging their mature IP backbones to offer voice trunking services to enterprises.

• Growing acceptance of Session Initiation Protocol (SIP) by both Service Providers and Enterprises as the primary protocol for multi-vendor interoperability.

Session Initiation Protocol (SIP) is an open industry multi-media signaling standard loosely modeled after existing, mature standards like SMTP and HTTP. The openness of SIP allows carrier voice equipment to interoperate with enterprise voice equipment over an IP network. The logical voice channel established between them is known as a SIP Trunk. SIP Trunks enable enterprises to create a single, pure IP connection to carrier clouds.

SIP Trunks offer both hard dollar and productivity improvements including:

• Elimination of the need for and cost of separate PSTN & data circuits at the enterprise.

• Reduced cost of communications between an enterprise and its partners and suppliers.

• Support for multiple forms of real-time communication including voice, video, and instant messaging, along with the benefits of user-based presence.

• Reduced toll charges from SIP origination/termination services to the PSTN for external long-distance and local access calls.

• Reduced hardware costs.

Avaya is leading the transformation of enterprise communications applications to take full advantage of the economic rewards promised from SIP networking. Avaya has dramatically expanded its offering of SIP enabled platforms and applications and has established an aggressive SIP interoperability program to ensure SIP remains open among a large array of application and service providers.

Introduction

For years, converged network visionaries have been promising a single communications circuit to connect the enterprise to a service provider network. Voice, data, e-mail, video, and other services would traverse this single integrated link, efficiently connecting the enterprise to its remote sites, customers, and partners.

Incremental steps toward this unified network have been taken in enterprise and service provider networks. Many years ago, carriers began routing voice calls over their private IP networks, to lower costs and improve network efficiencies. Meanwhile, enterprises were connecting IP-PBXs between sites over private IP networks to save toll costs. More recently, enterprises have begun to flatten, consolidate, and extend their communication networks by deploying remote IP-connected gateways off of central servers to reduce operating costs, leverage centralized applications, and enable management of multiple sites as a single system.

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But the vision of a single circuit to the cloud proved elusive. . . until now. Several critical factors have converged to bring Session Initiation Protocol (SIP) networking to the Enterprise market.

• Enterprise penetration of IP telephony has reached a critical mass. According to Nemertes research, over 71% of IT professionals surveyed now have IP telephony deployed in their network. In many cases, VoIP connectivity using SIP trunking can be readily introduced into these networks with minimal hardware and cost impact. This provides a ready market of significant size for service provider trunking services.1

• Enterprises are looking to expand the value of their IP telephony networks by extending the cost and deployment benefits to the partner/supplier eco-system side of their networks.

• Service Providers, looking for new revenue streams and competitive differentiators, are adapting their mature wholesale IP peering services into retail trunking services.

• Session border elements for SIP have become widely available, robust, and rich with the necessary security and management features.

• SIP has reached a required maturity level and is rapidly gaining acceptance as the preferred signaling protocol for real-time communications requiring multi-vendor interoperability.

This paper will explore the emergence of SIP Networking and what it can mean for achieving new network efficiencies and application deployment. It will also illustrate how customers can benefit now from the deployment of SIP trunks within their enterprise networks and explore how Avaya is supporting the effective deployment of SIP through its Converged Communications Server capabilities and its support of open interoperability for SIP networking and applications.

SIP Networking Across the WAN

The Session Initiation Protocol (SIP) is an open industry standard built on top of mature Internet standards, like TCP/IP and DNS. SIP’s openness allows carrier voice equipment to interoperate with enterprise voice equipment over an IP network. The logical voice channel established between them is known as a SIP Trunk.

SIP Trunks enable enterprises to create a single, pure IP connection to carrier clouds. An Enterprise SIP Proxy peers with a Carrier SIP Proxy, with the appropriate federations and security protections between them. The IP network continues to carry e-mail, Internet, and other corporate data traffic as it does today, and voice is simply layered on top of the network as another IP application. SIP sets up and tears down voice calls to and from the enterprise over this IP data network. Intra-enterprise or on-net calls are typically sent over the enterprise’s private IP or MPLS backbone, while off-net calls to the PSTN are handed off to the Service Provider via SIP. These off-net calls then ride the carrier IP network until the last mile, where a SIP gateway converts VoIP to TDM for calls to PSTN parties. Enterprises may also have the option of having on-net calls between facilities or locations traverse a Service Provider’s SIP VoIP backbone, which is typically dedicated to voice so that voice quality can be guaranteed.

SIP Trunks are transparent to the underlying data link topology. Since SIP is an application layer protocol, trunks can ride over any suitable Layer 2 protocol, such as MPLS or Metro Ethernet.

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SIP

PBX

PSTN

IPNetwork

LocalDIDs

TollFree800

PRI/DS-1 card at$3-$5K

Local Loop at$200-$500/month

VoIP Today

PBX

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Headquarters

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Usage charge~$0.029+/minute

PBX

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TollFree800

PBX

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LongDistance

Carrier SIP Network

Headquarters

SIP

Usage charge~20% less

Figure 1: SIP Trunking Changes How Enterprises Connect With Carriers

The introduction of SIP trunks can produce important new value to enterprises. In some cases, unique features like local number portability (LNP) are available. Calls destined for a local DID can be automatically re-routed over the SIP network to a central location such as a data center. This offers tremendous flexibility for contact center calls, among others. For instance, if a call is destined for a number associated with a remote sales branch, it can be re-routed to a central contact center agent, who could handle the call more efficiently and cost-effectively. If the caller needs to speak with someone directly at the branch, the agent can transfer the call to the branch using the enterprise WAN or a Service Provider SIP network, at a much lower rate than if the transfer were made over the PSTN.

SIP Trunks offer both hard dollar and productivity improvements. A single SIP link can support intra-enterprise connectivity and provide external long-distance and local voice access to the PSTN through the Service Provider SIP network. This results in “hard dollar” savings to enterprises as monthly recurring costs are lowered by eliminating all of the traditional PSTN circuits into the central PBX and by reducing the number of local trunks terminated by enterprise gateways at remote offices. Off-net calls to PSTN numbers are carried on the Service Provider IP backbone to the closest PSTN gateway, where the call is converted to TDM for the last mile. Per-minute toll charges on these calls are typically greatly reduced (approximately 20% less than similarly negotiated PSTN toll rates), since less tariffs are applied and fewer numbers of carriers have to handle the call.

The productivity and application benefits of SIP are becoming increasingly compelling. Full SIP networking, including the last mile connectivity of SIP Trunking, expands the flexibility and applications richness of roaming enterprise users back to communications services in the core of their corporate network. As cell phones and other mobile devices become SIP-enabled, SIP networking will provide the mobile user with connectivity to presence, Instant Messaging, and other application services. This will enable a mobile user with his cell phone to look “present” to a desk worker back at the office so the on-site user can IM or push-to-talk with the mobile worker.

SIP Trunks Offer

Cost Savings

and Productivity

Advantages

• Elimination of separate PSTN and data circuits at all enterprise locations

• Reduced external communication costs between an enterprise and its partners, suppliers, and customers

• Unification of multiple forms of real-time communications

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Service Providers Are Rebuilding Their Networks Around SIP

A diverse group of Service Providers are building SIP network capabilities including mobile network operators, large Inter-Exchange Carriers, Competitive Local Exchange Carriers, large cable operators, and Tier 2 service providers.

While initial offerings have tended to highlight new SIP services, many carriers have gone beyond offering SIP as a service by standardizing on it as the core signaling protocol within their networks. The two primary standards bodies (3GPP/3GPP2) that are defining specifications for third generation (3G) mobile systems have both specified SIP as the primary signaling protocol for IP-based voice and multimedia. Many new services being introduced on existing 2.5G wireless networks are already based on SIP.

Traditional carriers are also beginning to deploy SIP at the core of their network, using border elements such as signaling and media gateways to provide interworking with other protocols. Because SIP supports essentially any type of real-time communications, resources for security, authentication, billing, routing intelligence, etc. can be easily applied to any service the carrier develops. As a peer-to-peer protocol, much of the intelligence required for adding new SIP services can be distributed as modular components that can be easily added via application servers without having to make any changes to the underlying infrastructure.

Customer Network

Customer Network

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ia

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ignaling

SIP Signaling

Figure 2: Service Provider SIP Network

With SIP, carriers are rapidly replacing their hardware-based TDM networks with a software-centric model that provides greater flexibility, reduces costs, and provides a more open architecture. As Figure 2 shows, border gateway elements are required today at the edge of the network to interface with most enterprise networks. These border gateways represent a capital and operating cost to translate other protocols to a common SIP base. For the carrier, offering SIP trunk services allows them to avoid the border gateway complexity altogether. These cost savings can be passed to the enterprise in the form of very competitive pricing for services.

Service

Providers Are

Implementing

SIP Services

• Standardized interworking with a variety of other protocols

• Allows resources to be easily leveraged across multiple services

• Modular environment speeds new service deployment

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Exploring the Benefits of SIP Networking for the Enterprise

As Service Providers re-architect their networks using SIP, enterprises are beginning to discover the substantial benefits of enabling their corporate networks to incorporate SIP.

Among the most fundamental impacts of SIP will be improving the efficiency of communications between enterprises and their partners, suppliers, and customers. The initial wave of VoIP benefits has been primarily limited to intra-enterprise communications. Communications between enterprises, even those that are VoIP-enabled, still require a circuit-switched hand-off that impacts voice quality, adds complexity, and introduces unnecessary expense through intermediate carriers.

This concept of “Islands of VoIP Connectivity”, in which VoIP traffic is possible within individual “enterprise” islands but voice communication between islands requires traversal through the PSTN, are due primarily to signaling incompatibility between systems and the lack of a common source of directory information.2 SIP and universal directory solutions such as Electronic Number Mapping (ENUM) promise to address these issues.

To demonstrate how these technologies can solve the current networking bottleneck, a manufacturing enterprise example can be used to illustrate the four phases of networking transformation. The manufacturing company has developed an extensive supply-chain ecosystem with a number of partners and suppliers. Both the manufacturer and its partners/suppliers require regular real-time communications between each location within the ecosystem.

Phase 1: Traditional Networking Model Using PSTN

The requirement for regular real-time communications within the eco-system is initially met through a separation of voice and data networking. Data inter-connection between members of the ecosystem leverages the Internet and is secured through IP VPN connections. Voice inter-connection, including intra-enterprise, is achieved through the PSTN.

This traditional networking configuration separates voice and data traffic, as illustrated in Figure 3. It produces high quality voice capabilities, but does so at a high cost. With separate PSTN and data networks, hardware infrastructure and management costs are duplicated. Since all voice calls travel through the PSTN, pricing is based upon duration, geographic distance, and rigid tariff structures of the PSTN.

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PSTN

Branch Sites

Partner Supplier

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LocalPSTN

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Figure 3: Traditional Voice Network Architecture

Phase 2: Transition to IP Telephony Today

IP telephony technology today addresses some of the cost and efficiency issues of traditional dual networks. Figure 4 depicts the consolidation of voice and data network traffic across an enterprise IP WAN for both the manufacturing firm and its suppliers/partners.

Traditional

Voice Network

Drawbacks

• Parallel TDM/IP infrastructure costs

• Multiple TDM PRI connections

• Toll/tariff charges

• Wasted bandwidth

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PSTN

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ice

via

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Figure 4: IP Telephony Trunking Today

Note that from a physical connectivity standpoint, almost nothing has changed! In this example the enterprise and its suppliers have each migrated their internal voice communication over to IP by routing intra-company voice traffic over a private WAN or managed MPLS network, allowing them to reduce their intra-company toll and tariff charges. Consolidation of voice and data over IP further reduces overall network costs by increasing the enterprise’s utilization of available network bandwidth. The deployment of a VoIP WAN makes new applications such as off-shore outsourcing of call centers cost effective to implement since the WAN can be extended to any global location.

Yet the deployment of IP Telephony over an enterprise WAN has not yet fully optimized the full potential of IP networking for communications between an enterprise and its eco-system of partners/suppliers/customers. This is due primarily to the complexity of inter-connecting the different systems often in place between enterprises and because of the lack of a universal directory service to map phone numbers to the IP addresses of telephony servers. Most external enterprise communications still require a circuit-switched hand-off; enterprises must continue to maintain a parallel IP/TDM infrastructure that adds cost and prevents them from taking full advantage of IP telephony.

Phase 3: VoIP Transit Using SIP

The introduction of SIP Trunks addresses many of the remaining network cost and flexibility issues. As depicted in Figure 5, SIP Trunks allow the enterprise to take advantage of SIP termination and origination services that effectively transition PSTN connectivity for long-distance and local access calls to the Service Provider. The enterprise now only needs a single IP pipe to the cloud. Circuit-switched hand-off still occurs for communications between enterprises, but it is now handled through the Service Provider SIP network at a lower cost. Off-net calls to PSTN numbers are carried on the Service Provider’s IP backbone to the closest PSTN gateway, where the call is converted to TDM for the last mile. Per-minute toll charges on these calls

Limitations

of Today’s

IP Trunking

• “Islands of VoIP Connectivity” con-nected via PSTN

• Residual toll/tariff charges to other ‘Islands’

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are typically greatly reduced, since lower local rates are applied and fewer intermediary carriers are involved in handling the call. In addition to reduced toll charges, enterprises also directly benefit with “hard dollar” savings through the elimination of monthly recurring costs associated with multiple PSTN circuits and the capital/operating costs of dedicated PSTN interface hardware. These “hard dollar” savings are even greater for enterprises with large numbers of branch sites, as each of these sites can now operate with fewer LEC or CLEC trunks. With the advent of local number portability, most calls destined for the branch can be routed via SIP from the Service Provider into the enterprise data center. These calls can then be delivered to the branch using the enterprise’s private WAN or MPLS network, reducing the number of traditional local trunks needed in the branches to just the number required for survivability or emergency services (911).

Branch Sites

Partner Supplier

Sites

Data/SIP VoIPData/SIP VoIP

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Intra-Enterprise Voice via SIP

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Router

Router

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Telephony Feature Server

SIP Server

Telephony Feature Server

SIP Server

LocalPSTN

LongDistance

PSTN

Gateway GatewaySIPSP

SIPSPGateway Gateway

LAN

LAN

Figure 5: SIP Trunking

A recent survey of a few Service Providers offering these types of SIP Trunking services revealed the following rates:

Service Provider 1 Service Provider 2

PSTN Termination On-Net $0.005/minute $0.0064-$0.0072/minute

PSTN Termination Off-Net $0.029/minute $0.0221-.0251/minute

As figure 5 shows, these savings can be extended even further by also deploying SIP telephony within the enterprise. By unifying on SIP, the need for dedicated hardware devices is reduced, enabling an open, software-based communications infrastructure in which telephony and other communication services become modular applications. The result is simplified, open, and flexible enterprise communications.

Benefits of

SIP Trunks

• Single IP link for voice/data

• Converged Infrastructure cost savings

• VoIP transit reduces toll charges

• Open software-based architecture

• One protocol (SIP) for all communications

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Phase 4: Future Path Incorporating Universal Directory Services

The second contributor to the islands of VoIP connectivity problem has been the lack of a universal directory service like ENUM (Electronic Number Mapping), which provides a method for Service Providers to encode telephone numbers (formally known as E.164 numbers) into the Domain Name System (DNS). ENUM is a scalable, centralized directory that translates E.164 telephony numbers into IP addresses (or their equivalent, such as resolvable host names). This directory system is based on the Internet’s DNS, which is a highly scalable, hierarchical, directory service for translating domain names into IP addresses. ENUM was designed as a database that a VoIP Service Provider could consult in order to determine the IP address of an E.164 telephony number.

With a common directory source such as ENUM in place, highly efficient SIP peering interconnections between communities of enterprises become a reality. IP-based voice networks finally take on the best characteristics of the IP internet; ubiquitous coverage and easy network inter-connectivity over IP for all communications, as illustrated in Figure 6. With a common directory source such as ENUM in place for use within a service provider network, highly efficient SIP peering interconnections between communities of enterprises become a reality. For enterprises, the use of this capability can be transparent; inter-enterprise calls are handed off to the Service Provider, who then uses ENUM to resolve the E.164 digits to an IP address for routing the call (Figure 6).

Branch Sites

Partner Supplier

Sites

Data/SIP VoIP Data/SIP VoIP

Data/SIP VoIP Data/SIP VoIP

Intra-Enterprise Voice via SIP

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a S

IP

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Router

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TelephonyFeature Server

SIP Server

Telephony Feature Server

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ENUM

SIPSP

SIPSP

LAN

LAN

Figure 6: SIP Peering with ENUM Deployment

Call Center case study

Another example, this one of a retail enterprise with a large number of store locations, will illustrate how SIP trunking can also provide extensive benefits in a call center environment. In this example, there is no direct connectivity between the stores and the company’s central call center. The existing retail environment has multiple lines and employees to handle sporadic incoming call traffic. In addition the store employees

Benefits of the

all IP network

• Eliminates circuit-switched hand-offs

• Single IP link for voice/data

• Converged infrastructure cost savings

• VoIP intercon-nections between Enterprises

• Open software-based architecture

• Unification of all real-time communications over SIP

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generate millions of “800” minutes per month calling into the call center for sales assistance. The company has no budget to upgrade the retail store phone systems.

The current network topology results in several problems. In addition to the high cost of the “800” traffic, the limited lines available in each store result in inconsistent customer service because of intermittent “Busy signals” at the stores. In-store customer service also suffers when employees spend too much time handling local calls.

SIP Networking enables a reconfiguration of communications to address the problem. By implementing SIP-based 800 number and Local Number Portability for DIDs, 800 number and local DID number calls are converted by the Service Provider to SIP and transported directly to the call center over IP. This allows the call center to handle the majority of calls, but still perform a call transfer to the local store (possibly over the private WAN) if localized help is needed. The end result is a reduction in local PSTN services, lower 800 number charges, fewer number of local trunks at the retail stores, the potential for reduced employee head count at the retail stores, and improved customer service.

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Local Store

Customers

Stores

Local PSTN

TDM

Local PSTN

Gateway

Local DID

SP SIP

Call center handles majority of calls. Agents transfer the call back to store if needed

SP takes back active call and re-routes over gateway to local PSTN

Customer receives one call service, including local store support

SP terminates call locally, transports to call center over SIP

Customer dials number for local store

Figure 7: SIP Networking for the Call Center

SIP Outbound local and long distance services can achieve similar savings levels for enterprises with extensive outbound calling operations. Outbound calls are transported over the Service Provider’s SIP network and terminated to the PSTN at the last mile, reducing the impact of toll charges.

Contact Center

Example of SIP

Benefits

• No expensive upgrade required for existing store phone systems

• Store employees freed up to better service in-store customers

• Improved call center improves customer service by eliminating busy signals at stores

• Reduction in store call traffic reduces the number of lines needed at each store allows the elimination of extra phone lines

• Reduction in toll-free 800 number charges

• Call center eliminates dozens of T-1 PRI, replacing them with a single IP SIP link

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A recent survey of a few Service Providers offering these types of SIP Trunking services for call centers revealed the following rates:

Service Provider 1 Service Provider 2

Inbound DID $0.029/minute $0.005-$0.02/minute depending on pre-payment

Toll Free Inbound $0.039/minute

PSTN Termination On-Net $0.005/minute $0.0064-$0.0072/minute

PSTN Termination Off-Net $0.029/minute $0.0221-.0251/minute

Avaya: Building the SIP-Connectivity Ecosystem

For Avaya, SIP is a catalyst for the evolution of enterprise communications as an embedded multi-modal communication core service that can be integrated with other business applications.

Avaya is committed to helping its Enterprise customers fully exploit the advantages of SIP Networking. Its commitment is reflected both in the extensive portfolio of SIP products and applications and its programs designed to foster interoperability within the SIP ecosystem.

Avaya is building an ecosystem requiring the integration and collaboration of information and processes from thousands of Developers, Service Providers, System Integrators, Customers, and BusinessPartners. Avaya has SIP interoperability testing and certification programs for both enterprise and service provider equipment/application vendors. An online testing facility is also available through the DeveloperConnection (DevConnect) program, enabling companies to remotely test and verify basic SIP interoperability. Avaya SIP trunking conforms to applicable SIP standards. Avaya is also participating in SIPconnect, an industry initiative to define a standards-based approach to direct IP peering between SIP-enabled IP PBXs and VoIP Service Provider networks.

Avaya is strengthening its commitment to an open call model through the combination of Avaya Communication Manager and the Converged Communications Server. This will simplify the programming interface for Avaya Communication Manager and enable developers to create web services-based applications enabled by Avaya communications capabilities that combine the power of Avaya Communication Manager’s feature functionality with the broad and open reach of SIP.

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User Access

Meeting Exchange Enterprise

SIP Enablement Services

SIP Softphone

IP Softphone

4602 4610SW 462X SWCisco

SIP-Enabled Applications

Expanded Meet Me Conferencing

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H.323

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Handle-Based Dialing LDAP

Personal Profile Manager

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H.323

SIP Personal Information Mgr

Figure 8: Avaya SIP Architecture

The Avaya SIP architecture begins with the Converged Communications Server, which establishes the foundation for the Communication Services layer within the Avaya Communication Architecture. This layer unifies all enterprise real-time communications over an open SIP-based infrastructure, and provides the “glue” that binds with Avaya MultiVantage™ Communications Applications, exposing them as Web service components that can be easily invoked through standards-based clients or business applications, or as open APIs that provide a secure, reliable and highly scalable application development platform for access to Avaya Communication Manager services.

The Converged Communications Server is a family of related product offerings that currently consists of two primary components:

SIP Enablement Services: Avaya SIP Enablement Services combine the standard functions of a SIP proxy/registrar server with SIP trunking support and duplicated server features to create a highly scalable, highly reliable SIP communications network supporting telephony, instant messaging, conferencing, and collaboration solutions. These SIP services are modular software components that run over Linux on a dedicated industry-standard server platform, and are designed to operate in a networked hierarchical topology for scalability.

Application Enablement Services: Avaya Application Enablement Services consolidate the Avaya existing application enablement assets — such as Avaya Communication Manager Application Programming Interface (CMAPI) and Avaya CT — into a single, Linux-based platform. This enables enterprises to leverage the tremendous variety of computer-telephony integration and interactive response applications developed for these interfaces. Application Enablement Services allow for powerful new applications to be written and deployed that fully leverage Communication Manager via standards-based APIs and Web service components.

SIP Enablement Services and Application Enablement Services are modular offerings that are ordered independently and implemented as needed by the enterprise on separate, dedicated industry-standard servers. In combination, the new services of the Converged Communications Server create an application environment that combine the loosely coupled multi-modal services and presence capabilities available via a SIP-

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based architecture with the open APIs that expose the full breadth of features and functionalities of Avaya Communication Manager.

SIP Trunking

In many cases, enterprises with Avaya Communication Manager solutions can start to take advantage of SIP by adding Converged Communications Server (SIP Enablement Services) to establish a SIP standard compliant (IETF RFC 3261) network. In this configuration, Communication Manager supports SIP for internal telephony communications and trunk connections to the Service Provider.

Instant Messaging (IM)/Presence

Once SIP Enablement Services are in place, a gradual step-by-step migration path allows the enterprise to maximize the benefits of SIP while fully preserving compatibility and investment protection with existing H.323, digital and analog endpoints and infrastructure. For example, SIP Enablement Services can be leveraged to introduce secure Enterprise Instant Messaging (IM) and user presence integrated with telephony through the Avaya IP SoftPhone and IP Agent client applications. Avaya IP SoftPhone and IP Agent combine H.323 telephony support with a SIP-based IM client and a presence-enabled contact directory that supports both voice and IM communications. This allows enterprises to extend the benefits of user presence and IM to all users without the need to make extensive changes to their existing voice infrastructure.

SIP Telephony

SIP supports both numerical (telephony) and alphanumeric addressing, providing a critical bridge for communications between PSTN and Internet networks. This allows users on either network to reach any other user without giving up existing devices or the advantages of each. When enterprises are ready to make the transition to SIP Telephony, Avaya SIP Enablement Services will allow them fully leverage the benefits of SIP while supporting full compatibility and use of their existing infrastructure and endpoints.

For enterprises that have already deployed H.323 IP Telephony, the migration process can be initiated immediately and transitioned at whatever pace is desired. Once registered and licensed on SIP Enablement Services, existing Avaya 4602SW/4610SW/4620SW/4621SW IP phones can convert their operation from H.323 to SIP through a simple firmware upgrade. Avaya IP SoftPhone users have a similar migration path to SIP telephony through the Avaya SIP SoftPhone, which supports SIP signaling for both IM and telephony communications.

User Mobility

SIP is well-suited for mobility requirements. When a user logs onto a SIP device, it registers the user and sends the SIP URI of the device to the registrar service, which is used to route calls to/from the user. Avaya Personal Profile Manager, a user mobility application within SIP Enablement Services, leverages this capability to create a custom work environment that follows the user. Personal Profile Manager is centralized service that resides in each SIP Home Proxy Server and communicates with its assigned SIP endpoints to receive, store, and distribute user profile information.

SIP Applications

Through SIP Enablement Services, a number of Avaya MultiVantage™ Business Communications Applications, including Communication Manager telephony and Meeting Exchange web/audio conferencing, become available to a wide range of standards-based user agents, web-based applications, and communication devices to create a new paradigm of Converged Communications that will lead to increased flexibility and cost efficiency.

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Avaya Communication Manager is the next generation of Avaya call processing software. Designed as an open, scalable, and highly reliable telephony solution, it effectively scales from under 100 users to as many as 36,000 on a single system — and to more than one million users on a single network. Designed to run on a variety of Media Servers, Avaya Communication Manager provides centralized call control for a resilient, distributed network of media gateways and a wide range of analog, digital, and IP-based communication devices. Avaya Communication Manager integrates with the Avaya Converged Communications Server (SIP Enablement Services) to become a telephony feature server that supports seamless communications between existing analog, digital, and IP phones and SIP-based endpoints. With Avaya Communication Manager Extended Access, standard SIP phones can even gain access to advanced SIP telephony features.

Avaya Meeting Exchange is an audio and web conferencing solution that is supported by the Avaya S6200 server. The standards-based S6200 scales from 24 to 240 IP ports. It uses SIP as the signaling protocol for conferencing participants, and as a communication protocol that allows the conferencing application to run on standard third party media servers. SIP is also leveraged for redundancy – redirecting calls to a back up system if necessary. The S6200’s standards-based architecture allows enterprises to leverage existing infrastructure investments, smoothing the migration from TDM to IP and delivering ample cost and efficiency benefits. For larger-scale deployments the S6200 can be deployed with an S6800 media server to support up to 10,000 ports.

Conclusions

SIP Networking Services and Trunking are an important new category of networking capabilities for enterprises seeking to achieve new efficiencies in their network design, cost reductions for WAN networking and enablement of a new wave of SIP enabled applications. SIP networking offers are appearing now in the Service Provider marketplace with a growing number of carriers bringing SIP WAN capabilities to your enterprise locations.

Avaya as the leader of enterprise business communications applications has taken a strong step in the development and deployment of SIP networking and applications to both exploit the power of SIP as well as insure its openness. It offers one of the most extensive portfolios of SIP applications with its SIP enabled Avaya Communication Manager and Converged Communications Server at its core.

Avaya has also made a large commitment to establishing openness and interoperability for SIP through its DevConnect and SIP Connect Programs, which supports software developers, systems integrators and Service Providers in developing interoperability with Avaya products and services.

Enterprise network and applications planners should be actively learning about and planning to take advantage of SIP networking. Avaya can help support your exploration and deployment SIP networking.

For more information on how Avaya can take your enterprise from where it is to where it needs to be, contact your Avaya Client Executive or Authorized Avaya BusinessPartner, or visit us at www.avaya.com

Footnotes1 Johna Till Johnson, “Technology Trends & Futures”, Nemertes Research, January 2005

2 Daniel Goldberg, Island Hopping: Interconnecting VoIP Systems and Services, Burton Group, Version 1.0, September 2, 2004.

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