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Intelligent Session Control

in the EnterpriseRibbon Communications Edition

by Walter Kenrich and Lawrence C. Miller


Intelligent Session Control in the Enterprise For Dummies®, Ribbon Communications Edition

Published by John Wiley & Sons, Inc. 111 River St. Hoboken, NJ 07030-5774 www.wiley.com

Copyright © 2018 by John Wiley & Sons, Inc.

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Table of Contents iii


Table of ContentsINTRODUCTION ............................................................................................... 1

About This Book ................................................................................... 1Foolish Assumptions ............................................................................ 2Icons Used in This Book ....................................................................... 2Beyond the Book .................................................................................. 3Where to Go from Here ....................................................................... 3

CHAPTER 1: Understanding Intelligent Session Control and Policy ....................................................................... 5Business Communications Are Ever Evolving ................................... 5Policy and Session Management Concepts ...................................... 8Relating Policy and Routing to Security, the Network, and Unified Communications ............................................................. 8Understanding the Role of an SBC ................................................... 11Realizing the Benefits of Intelligent Session Control ..................... 11

CHAPTER 2: Implementing Access Policies for RTC Security ........................................................................... 13Call Admission and Identification ..................................................... 13Toll Fraud and Policy Controls .......................................................... 16

CHAPTER 3: Network Routing Methods for Real-Time Communication ................................................ 19What is Routing? ................................................................................. 19Route Prioritization ............................................................................ 20Value-Based Routing .......................................................................... 22Additional Routing Methods ............................................................. 24

CHAPTER 4: Exploring Intelligent Session Control Use Cases .......................................................................................... 27Centralized Policy and Routing ......................................................... 27Dial Plan Interworking ....................................................................... 29International Toll Avoidance ............................................................. 30Rerouting Traffic ................................................................................. 31Microsoft Active Directory Routing and Translation ...................... 32

iv Intelligent Session Control in the Enterprise For Dummies, Ribbon Communications Edition


CHAPTER 5: Ten Reasons to Choose Ribbon’s PSX for Intelligent Session Control ................................................. 35Reduce PBX Management and Maintenance Costs ....................... 36Reduce Long-Distance Costs ............................................................. 36Route Calls Based on Value ............................................................... 36Increase Employee Productivity ....................................................... 37Sweat Your Existing Assets ................................................................ 37Make Your Network More Reliable .................................................. 37Improve Security with Access Control ............................................. 37Virtualize Infrastructure and Network Functions ........................... 38Integrate with Third-Party Solutions ................................................ 38Leverage Extensive Features and Capabilities ................................ 38

GLOSSARY .......................................................................................................... 39

Introduction 1


Introduction

Today’s enterprises strive to maintain and improve customer service while enabling more efficient employee collabora-tion. They can achieve this by optimizing the use of network

resources and simplifying network operations with intelligent session and routing control solutions.

As Session Initiation Protocol (SIP) session traffic continues to grow — a trend accelerated by the rapid adoption of multime-dia devices such as smartphones and tablets — service provid-ers and enterprises must find ways to effectively manage, route, and control this traffic in their networks. Additionally, most real-time communication (RTC) networks today are multivendor, multiprotocol environments with fragmented policy information spread out across individual geographies and individual network elements.

The most effective way to manage and control SIP traffic and multivendor, multiprotocol environments is through a robust centralized policy management solution. A centralized model allows service providers and enterprises to easily manage and update their SIP policies from a master server in their core network and quickly copy this policy intelligence to local policy servers deployed nationally or internationally.

About This BookIntelligent Session Control in the Enterprise For Dummies, Ribbon Communications Edition, consists of five brief chapters that explore the following:

» Which policies are important and the benefits of intelligent session and routing control (Chapter 1)

» How to enable access control and security policies (Chapter 2)

» How to configure various intelligent routing features (Chapter 3)

» Real-world use cases and scenarios in the enterprise (Chapter 4)

» The benefits of Ribbon’s intelligent session control solution (Chapter 5)

2 Intelligent Session Control in the Enterprise For Dummies, Ribbon Communications Edition


There’s also a convenient glossary at the end of the book — in case you get stumped on any technical acronyms or concepts.

Foolish AssumptionsIt’s been said that most assumptions have outlived their useless-ness, but we assume a few things nonetheless!

Mainly, we assume that you are a CIO, CTO, IT director, or man-ager responsible for RTC in your company. We also assume that you’re working for an enterprise or telecommunications service provider supporting multiple locations for your organization. As such, this book is written primarily for technical readers with some understanding of networking, RTC, and security.

If any of these assumptions describe you, then this book is for you! If none of these assumptions describe you, keep reading anyway. It’s a great book, and when you finish reading it, you’ll know a few things about intelligent session control!

Icons Used in This BookThroughout this book, we occasionally use special icons to call attention to important information. Here’s what to expect:

This icon points out information you should commit to your non-volatile memory, your gray matter, or your noggin — along with anniversaries and birthdays!

You won’t find a map of the human genome here, but if you seek to attain the seventh level of NERD-vana, perk up! This icon explains the jargon beneath the jargon!

Tips are appreciated, never expected — and we sure hope you’ll appreciate these tips. This icon points out useful nuggets of information.

These alerts point out the stuff your mother warned you about (well, probably not), but they do offer practical advice to help you avoid potentially costly or frustrating mistakes.

Introduction 3


Beyond the BookThere’s only so much we can cover in 48 short pages, so if you find yourself at the end of this book, thinking “where can I learn more?” just go to https://ribboncommunications.com.

Where to Go from HereWith our apologies to Lewis Carroll, Alice, and the Cheshire cat:

“Would you tell me, please, which way I ought to go from here?”

“That depends a good deal on where you want to get to,” said the Cat — err, the Dummies Man.

“I don’t much care where . . . ,” said Alice.

“Then it doesn’t matter which way you go!”

That’s certainly true of Intelligent Session Control in the Enterprise For Dummies, Ribbon Communications Edition, which, like Alice in Wonderland, is also destined to become a timeless classic!

If you don’t know where you’re going, any chapter will get you there — but Chapter 1 might be a good place to start! However, if you see a particular topic that piques your interest, feel free to jump ahead to that chapter. Each chapter is written to stand on its own, so you can read this book in any order that suits you (though we don’t recommend upside down or backwards).

We promise you won’t get lost falling down the rabbit hole!

https://ribboncommunications.com



CHAPTER 1 Understanding Intelligent Session Control and Policy 5


Understanding Intelligent Session Control and Policy

In this chapter, you learn about real-time communication (RTC) policy management, how it relates to intelligent routing and session control, and the benefits of a centralized policy and

routing server.

Business Communications Are Ever Evolving

Business communications are always evolving. With Session Ini-tiation Protocol (SIP) well over 15 years old, we’ve gone from old touch-tone services to voice over IP (VoIP), and now to cloud RTC as a service. And with mergers and acquisitions seemingly hap-pening every day, workers are now communicating with their colleagues around the world across many disparate platforms.

Chapter 1

IN THIS CHAPTER

» Recognizing the evolution of business communications

» Defining real-time communication policies

» Controlling SIP sessions and routing

» Understanding the role of the SBC

» Looking at the benefits of intelligent session control



With SIP, new services and modalities — everything from voice and video to collaboration, WebRTC, Wi-Fi Calling, and more — for RTC are enabled. Bring your own device (BYOD) users are empowered to use their mobile devices as they see fit, which often means they’re not used with any consideration for how the enter-prise as a whole might be impacted. Furthermore, as BYOD end-points become more distant from the core network, the harder it is to control access and costs.

SIP has become just another application on the network, and without appropriate policy and routing measures in place, costs and complexity increase substantially. More importantly, RTC security threats — including financial threats, network threats, and compliance issues — become real risks for your business and can impact the underlying technology of your enterprise. So, while business communications are ever changing — which is good for businesses — it comes with some challenges.

To end-users, a SIP session doesn’t look any different than a traditional voice call: It confirms the availability of a particular endpoint, establishes a connection between two endpoints, and controls the exchange of media during the session.

In reality, SIP sessions are much more complex than that; they consist of multiple policy decisions such as security settings, routing paths that might be based on lowest cost and/or highest quality, media transcoding options, and signaling interworking. Decisions like these may be made multiple times during a single SIP session, and the intelligence for these decisions may reside in dozens of different network elements such as soft-switches, session border controllers (SBCs), and least-cost routing (LCR) engines. Thus, what appears to be a simple VoIP call may actu-ally be a complex series of communications between a call routing server, a policy database, multiple SBCs, an Electronic Numbering (ENUM) server, and other network devices.

For enterprise communications networks, the complexity of SIP traffic creates challenges which include the following:

» It consumes more bandwidth.

» It can introduce quality issues (for example, latency and dropped packets).

» It may be prone to errors because the policy information is often manually entered and updated on each device.



To minimize the complexity of SIP networks and provide more robust management of SIP sessions, a centralized policy and rout-ing server can be deployed to address some of the most important challenges facing enterprise multimedia networks.

A centralized policy and routing server enables the following capabilities and benefits:

» Consistency in policy and routing decisions across custom-ers, employees, and applications, regardless of location, which dramatically reduces operating expenditures (OPEX) costs and management complexity while allowing for easier and more efficient session management

» Simplicity in the management and provisioning of network-wide policy and routing information, including moves/adds/changes (MACs), overload controls, least cost routes, and number portability lookups

IT no longer has to manually update and manage local policy within multiple applications, which is prone to errors.

» Eliminates national and international long-distance toll charges between employees anywhere in the world by leveraging IP-based enterprise wide-area networks (WAN) for VoIP calls

» Avoids downtime from a cut line or hardware failure by rerouting traffic automatically over the IP-based WAN in the event of a single SIP trunk failure at an office or call center location

» Seamlessly integrates multivendor private branch exchange (PBX) environments to create centralized dial plans and call routing

» Intelligently manages communication sessions routing paths, priority, and admission control based on granular policies, such as media type, source/destination, and time of day/week



Policy and Session Management Concepts

A centralized policy and routing server can provide enterprises with the following types of business logic:

» “Strategic” business logic: For example, which SIP applica-tion takes precedence, and under which circumstances is the network allowed to share bandwidth or access resources with other applications attempting to use those facilities?

» “User-based” business logic: For example, how are individual users (executives, branch users, high-profile customers, and so on) allowed to interact with network resources?

» “Application-based” logic: For example, is the application at its peak of handling based on total sessions or capacity, what interworking is required, what disaster recovery routing should be used, and so on?

Relating Policy and Routing to Security, the Network, and Unified Communications

A centralized policy and routing server provides a complete view of your network and is the heart of any intelligent routing and session control solution. The policy and routing server works in conjunction with SBCs and other multivendor equipment in a network to provide a cost-effective, simple-to-use, and easy-to- implement policy solution (see Figure 1-1).

A centralized policy and routing server in the enterprise needs to be an intelligent session and routing control solution that applies specific rules to IP-based communications, such as VoIP calls, video conferences, unified communications (UC) sessions, third-party applications, and messaging systems. Enterprises can control multimedia communications to customers and employees



with flexibility and ease. For example, enterprises can centrally manage communications coming into contact centers based on specific business rules in the form of policies, in order to route the top-priority calls to a particular center of expertise. Or enterprises could prioritize (or not) the delivery of bandwidth-intensive video applications based on different business rules.

Intelligent session and routing control solutions can also help enterprises significantly lower their long-distance fees, use fewer mobile minutes, simplify network administration, optimize bandwidth utilization, and reduce network downtime — saving as much as 55 to 75 percent in recurring communications costs.

Intelligent session and routing controls also provide a view of policies applied across your network and allow you to consolidate and centralize dial plans and integrate multiple vendors to look like a single communications system. With this type of solution enterprises can improve their communication flows by applying various types of policies including the following:

» Security/access control: Enforce security policies such as authentication, identification, authorization, call screening, fraud control, and more. Read Chapter 2 to learn more about access control and security policies.

FIGURE 1-1: A centralized policy and routing solution provides comprehen-sive, network-wide intelligent session and routing control.



» Network: Manage actions of network elements such as PBXs, session managers, call servers, and SBCs. Examples of network policies include routing based on destination; time of day (ToD) or time of week routing; session blocking based on pre-defined criteria (for example, authorized/unauthorized access); and level of quality of service (QoS) to ensure high-quality voice and video sessions. Read Chapter 3 to learn more about network policies.

» UC: Bring together multivendor environments to create centralized dial plans, centralized call routing, and more, and determine when and how different types of communications flow across the network (see Figure 1-2). Examples of a RTC policy could include assigning a lower priority to video applications and routing high-definition (HD) calls through HD paths.

FIGURE 1-2: Normalize multivendor, multidial plan networks with centralized dial plans.



Understanding the Role of an SBCAn SBC controls a SIP network by admitting (or not admitting) and then directing communications between two or more end devices on the network. These communications are called sessions.

SBCs are deployed at the network perimeter (or border), so they can be implemented by enterprises to control and secure RTC sessions. An SBC performs the following functions:

» Secures the network: An SBC hides the topology of an enterprise’s internal IP network, encrypts the sessions for privacy and compliance, and protects and secures RTC from various threats, such as spoofing and denial-of-service (DoS) attacks.

» Enables SIP trunking: An SBC provides you with a secure point of demarcation or termination for a SIP trunk connec-tion into your communications network.

» Interworking and protocol translation: An SBC provides a smooth experience in terms of interconnecting and inter-working between different applications, networks, and the protocols running over them.

» Acts as session traffic cop: The SBC is the gatekeeper to SIP-based services in an enterprise network. In this role, SBCs perform session admissions control, which is the process of determining who has access to the network.

» Media services: With SIP, different codecs are used to support bandwidth and/or quality concerns. An SBC will know which codecs are supported and required on each side of the network border. It will decode and then re-encode the voice or video signal to the required codec as it crosses the network border.

Realizing the Benefits of Intelligent Session Control

In large SIP deployments where multiple SBCs are installed at multiple network borders, the task of individually configuring routing and policies on all SBCs can be tedious, error-prone, and



expensive. To address these issues, a centralized policy and rout-ing server with a single set of policy rules can be used by each SBC on the network.

Centralized policy and routing management provides value to the enterprise in four key areas:

» Intelligent session management: Policies can include how to handle routing, priority, and admission control of sessions, for greater savings and improved call quality, based on attributes such as media type, source and destination, ToD, routing path, and others.

» Multi-PBX dial plan interworking: Bring multiple devices and signaling protocols together in a multivendor environ-ment to coexist with centralized and simplified dial plans and call routing.

» Greater operational efficiencies and savings: Reduce OPEX with centralized management on a single database, international toll avoidance, and more efficient use of network bandwidth.

» Enhanced business continuity: Set up alternate routing paths to protect against carrier and switch failures by intelligently routing traffic around network failures.

CHAPTER 2 Implementing Access Policies for RTC Security 13


Implementing Access Policies for RTC Security

In this chapter, you discover how a centralized policy and rout-ing server can help you implement advanced access policies to promote security in an enterprise real-time communication

(RTC) network. An example is screening or prohibiting all calls from an originating entity that is attempting to initiate a tele-phony denial-of-service (DoS) attack. Centralizing a policy and routing server can also help prevent toll fraud by blocking or pro-hibiting specific calls based on call type, country code, destina-tion, or leading digits.

Call Admission and IdentificationAs with any network, you want to be sure that users have proper privileges to access the network and that they are who they say they are. The same is true for calls that are initiated by using Ses-sion Initiation Protocol (SIP) in an RTC network. That’s why it’s important to ensure that calls are properly identified, and identi-fication is authenticated and authorized prior to allowing access to the network. Enterprise and cloud RTC deployments require a centralized policy and routing layer between your existing peers and applications, as shown in Figure 2-1.

Chapter 2

IN THIS CHAPTER

» Controlling who has access and is authorized to use your RTC network

» Using policies to enforce conditions of a call session

» Preventing toll fraud



RTC policy is based on relevant conditions in a session. For exam-ple, call admission looks at parameters such as resource access permissions, as well as capacity and bandwidth policies. Resource access permissions include traffic policing, DoS protection, IP access lists, and various policers. Capacity and bandwidth policies look at conditions such as rate limiting, various bandwidth capac-ity controls based on the media (for example, voice or video), and ingress session admission control. Figure 2-1 shows a high-level view of a session — a caller from one location calling/accessing an application in another location.

Identification, authentication, and authorization services in a RTC network include the following:

» Blacklist and whitelist policers: A blacklist specifies prohibited numbers and/or IP addresses, whereas a whitelist specifies allowed numbers and/or IP addresses in an RTC network. Generally, any blocking criteria (for example, 900 numbers, IP addresses, Uniform Resource Identifiers or URIs, called parties, and so on) can be a blacklist service. Calls not explicitly blocked by the blacklist are permitted to proceed to

FIGURE 2-1: Enterprise and cloud RTC deployments with a centralized policy and routing layer.



their destinations. On a blacklist, you provision one or more of the following types of blocking rules:

• Business group blocking profiles: Each profile specifies business groups, service groups, and business group subscribers that must be blocked.

• Blocking label profiles: Each profile specifies countries, national numbers, and call types that must be blocked.

To bypass generalized blocking rules, you can create a whitelist by defining certain service exceptions. For example, all international calls are blocked or blacklisted, but a service exception allows calls with country code 44 (United Kingdom) to proceed. An explicit whitelist service eliminates the need to provision both blocking and exceptions, thus simplifying the provisioning process.

On a whitelist, you provision blocking label profiles to always permit calls into and out of the RTC network. Each profile specifies countries, national numbers, and call types that must be allowed. Calls not explicitly allowed by the whitelist may be blocked if they meet the conditions of the blacklist.

A centralized policy and routing server will use these two policers to determine if either of them will be used to define how the call must be handled. Execution precedence between blacklist and whitelist processing should be available, and the order profiles are examined within each list.

Whitelist processing should always be executed before blacklist processing.

» Remote user access: In this scenario, remote users dial into a service access number (SAN). If their calling number isn’t known in the RTC network, they’re prompted for a personal identification number (PIN) for authentication and access to the network.

If the user is pre-registered in the centralized policy and routing server, remote access can easily be used by offsite users at fixed locations and small enterprise sites. Although these users must dial a SAN to gain virtual private network (VPN) access, they aren’t required to enter a PIN because their calling number has been pre-registered in the central-ized policy and routing server’s database.



» Authorization code services (or class of service): Authorization code services allow you to provision a different set of services to users who have a valid authorization code. For example, you can provision authorization code services such that only employees that present the authorization code are permitted to make calls to specific area codes or countries. As another example, you can provision authoriza-tion code services for employees to use in public venues, such as airports, whereby they need to enter an authoriza-tion code before making an out-of-network call. This is done to prevent malicious users from calling pay-per-call services, which lead to theft of services and fraud in those networks.

Authorization code services generally require you to create a two-stage call processing script. The script answers the call, prompts the caller for an authorization code, and collects the digits. Then, the centralized policy and routing server determines if the authorization code is valid by comparing the digits entered by the caller to valid authorization codes associated with the user.

Toll Fraud and Policy ControlsOne common type of fraud that can be mitigated with a policy server is robocalling. In this type of fraud, computers are used to make short duration calls to keep lines busy at a contact cen-ter. Some smart programs may use the contact center’s interac-tive voice response (IVR) to go into a loop, keeping the lines busy for a long time. This results in real customers either not getting connected, or getting delayed in connecting to customer support phone numbers. Customers usually call customer support phone numbers with issues that they want resolved. Busy signals only exacerbate their frustration, resulting in higher customer churn. The policy server can be used to block calls from potential robocall originating numbers.

Toll fraud is one of the oldest forms of computer crime. Toll fraud involves a malicious user gaining unauthorized access to voice services on an enterprise network, for example, to make international calls or use other toll services. In other cases, a



cybercriminal might gain access to special classes of numbers to extract revenue, such as repeatedly calling a duplicitous premium rate number (perhaps a “1-900 psychic hotline”) that the cyber-criminal operates.

The global impact of toll fraud has soared to more than $46.3 bil-lion annually, or slightly more than 2 percent of all global telecom revenues, according to a recent Communications Fraud Control Association (CFCA) Global Fraud Loss survey. To put that into per-spective, credit card fraud was around $14 billion dollars over the same period. More ominously, toll fraud losses are growing at a faster rate than global telecommunications revenue.

Dial-through fraud (DTF) is the most damaging form of toll fraud, in which an IP private branch exchange (PBX) is compromised in such a way that an attacker using a simulated call generator can dial in to the PBX, get a dial tone, then hairpin dial out to an inter-national premium number to generate fraudulent revenue that’s charged to the target enterprise. See Figure 2-2 for an example of DTF.

Attackers may use DTF themselves to generate revenue directly, or they may sell access to the compromised PBX to other cybercrimi-nals to generate calls. DTF calls are usually short and variable in duration, and are usually generated outside of normal business hours to avoid detection.

FIGURE 2-2: An example of DTF.



A centralized policy and routing server can help an enterprise protect against various forms of toll fraud by implementing cer-tain access and security functions, such as the following:

» International number blocking: Block all outbound international calls.

» High fraud country blocking: Specify a list of countries to which you want to block calls.

» Blocking: Block calls based on the called number and call type.

» Limit the number of minutes based on destination numbers: Set fixed limits on the number of minutes permitted over a period of time (for example, monthly) to specific destination numbers to limit the potential impact of toll fraud.

The goal in centralizing access and security policies is to ensure no gaps exist in the enterprise’s security posture. After a malicious user finds an opening, toll fraud comes into play. Thousands of calls can be made to an unsuspecting organization, driving fraud costs sky high. In this case, a centralized policy and routing server can be configured to allow only a small num-ber of calls — or no calls — through to restricted countries. The centralized policy and routing server will detect the toll fraud, thereby triggering appropriate security measures. It will respond and instruct the SBC to not accept any more call attempts from that malicious IP address. And the SBC, in combination with the centralized policy and routing server, will now prevent any fur-ther fraudulent calls from that IP address to specific and costly destinations.

CHAPTER 3 Network Routing Methods for Real-Time Communication 19


Network Routing Methods for Real-Time Communication

In this chapter, you discover different routing options and features that can be configured in an enterprise real-time communication (RTC) network.

What is Routing?Routing is a process that determines how a session (or call) selects a path to its destination in a network, or between or across mul-tiple networks. One way to understand routing is to think about public or private transportation. For instance, when you’re going out to dinner, you determine your route to get to the restaurant by using the system of streets, roads, and highways. Many times, you take the fastest route, but there may be instances in which the roads are congested or closed, so you need to know alternative routes that are available.

Chapter 3

IN THIS CHAPTER

» Defining network routing for real-time communication

» Prioritizing network routing

» Using least-cost routing to reduce costs

» Exploring other routing mechanisms



RTC routing is generally performed in the same manner. In IP networks, routing is the higher-level decision making that directs the RTC flows from their source toward their destination through Session Initiation Protocol (SIP) trunks and intermediate network nodes by using specific routing mechanisms.

Entities involved in basic standard routing scenarios include a standard route, routing criteria, element routing priority, and the routing label (see Figure 3-1).

Essentially, incoming routing requests to a centralized policy and routing server contain many bits of information about the call/session. A centralized policy and routing server will compute information about the call, based on these inputs and its own internal databases.

Route PrioritizationIn an RTC network, you can use numerous methods to prioritize how sessions will be routed in the network. For example, the fol-lowing methods can be used for RTC route prioritization:

» Sequence: Each trunk group (TG) is assigned a specific sequence number. In sequence-based prioritization, the

FIGURE 3-1: The standard routing process with associated post-processing functions.



priority of the route is determined by the sequence number assigned to a particular TG. TGs with lower sequence numbers have the highest route priority for the session to reach its destination. If the lower numbered TGs are saturated or the session doesn’t conform to specific policies, higher sequenced TGs will be used as the routing path.

» Proportion: Proportion-based prioritization attempts to allocate sequential calls across various TGs so that over time the average number of calls sent to each TG will be propor-tional to the provisioned numbers. For example, if the numbers assigned to TGs A, B, and C are 50, 30, and 20, respectively, the goal is to route 50 percent of the calls to TG A, 30 percent to TG B, and 20 percent to TG C. This enables enterprises to route sessions to specific SIP trunks for more efficient use of their network.

» All proportion: Proportions are used to determine all the routes in the route list on a call-by-call basis. For the first route, a policy and routing server follows the process described in the preceding bullet for the proportion option, by generating a random number. For the second route, the policy server recalculates the associated random number range for the remaining routes and then generates a new random number. The policy server continues this process to populate each of the remaining routes in the route list.

» Round robin: In the round robin method, the centralized policy and routing server scrolls through the list of available TGs in sequential order, such that each TG will be returned as the first route at some point in time. Round robin is most effective in enterprises when the desired traffic distribution is equal across all routes.

» Least-cost routing (LCR): In the LCR method, the priority of the route to a particular TG is determined based on cost with the least (lowest) cost TG being the first route selected. An example of LCR is shown in Figure 3-2.

Each TG in the network can be configured with a cost value. The centralized policy and routing engine uses that information to calculate a list of TGs in ascending order of cost, with the lowest cost TG being the first. In the case of equal cost TGs, a secondary route prioritization type can be selected. In larger enterprises, this enables them to cost optimize their network.



Value-Based RoutingValue-based routing provides the most advanced intelligent routing based on combining the least cost with quality of service (QoS) metrics of that TG. This type of routing solution allows enterprises to reduce costs and increase the quality of service for their customers and employees by selecting the most appropriate TGs based on a variety of criteria, including LCR rates, service level agreements (SLAs), voice quality, and network capacity. Because of the complexity of LCR calculations and the number of variables involved, only a centralized policy and routing server can efficiently aggregate all the information and calculate the optimal routes.

Enterprises will generally use “value-based” routing to lower costs and maintain a high QoS for their users when they have multiple SIP trunking service providers in multiple geographies. Value-based routing options include the following:

FIGURE 3-2: An example of LCR.



» LCR: In the LCR method, the priority of the route to a particular TG is determined based on cost (see the preceding section for more details).

» QoS-based routing: QoS is a combination of various measured parameters that impact the call and voice quality, including

• Answer Seizure Ratio (ASR): The relationship between the number of seizures that result in an answer signal and the total number of seizures.

• Average Call Hold Time (AHT): The average length of time between the moment a caller finishes dialing and the moment the call is answered or terminated. This is primarily used in call centers and is a key factor when determining staffing requirements of call centers.

• Post Dial Delay (PDD): The length of time that’s con-sumed when a phone user dials the last digit in a phone number and when the user hears the ring or busy tone of the phone number being called.

• Mean Opinion Score (MOS): The arithmetic mean value of user opinion for rated voice quality (1 — Lowest, 5 — Highest).

• Jitter (in egress side): The variation in the time between packets arriving, caused by network congestion, timing drift, or route changes in voice over IP (VoIP). Jitter is typically measured in microseconds or nanoseconds and should be less than 100 milliseconds between the starting and final point of the communication.

• Average Packet Loss Percentage (PLP): The average number of lost packets to the total number of transmit-ted packets.

• Minutes of Use (MOU): This is a volume commitment from the Buyer Side to the Seller Side to route the committed amount of calls in terms of volume, typically on a monthly basis.

• Network Effectiveness Ratio (NER): The percentage of calls successfully completed. Unlike Call Completion Ratio (CCR), NER includes calls terminated by Busy and No Answer signals.



QoS plays an important part in routing decisions for RTC. Without the actual QoS data, enterprises can only rely on the committed QoS data from its SIP trunk providers. Often the actual QoS met-rics in the network fluctuate from the committed ones and, as a result, SIP trunk providers can’t guarantee QoS commitments to their enterprise customers. In other words, it is important to take into consideration the actual QoS metrics of the SIP trunk when making routing decisions.

Additional Routing MethodsAdditional routing methods provide flexible options for routing call sessions on enterprise RTC networks. These methods include the following:

» User-defined flexible routing: In general, SIP sessions have proprietary or standard SIP header and message body parameters that can be routed, based on the occurrence of configured patterns in the header or body of the message. In a centralized policy and routing server deployment, you will likely find a large multivendor network consisting of third-party private branch exchanges (PBXs), third-party session border controllers (SBCs), gateways, and feature servers. In this type of deployment model, there will be use cases where third-party elements may have proprietary headers that need to be parsed and routed according to some unique logic. These proprietary headers are typically used to convey information to a specific network element that needs to process the call in a special way. For example, call correla-tion, SIP Recording (SIPRec), archival, and so on. Similarly, you may encounter session flows in which there is a parameter in the Session Description Protocol (SDP) body of the SIP messages where there’s an expectation to route differently, based on SDP-defined parameters. These parameters could be related to the codec type or to the Codec Definition Parameters. In such cases, a centralized policy and routing server can offer user-defined flexible routing, in which IT staff can define the parameter patterns that could occur in call flows, and then define rules that will provide special routing for those calls.



» Emergency (911) routing: An emergency call must always receive top priority. Therefore, it’s imperative that any centralized policy and routing server can recognize calls to 911 (the North American variant) as an emergency call. The centralized policy and routing server doesn’t check for any provisioned services for the call; it simply routes the call, prioritized over any other call traffic. To ensure that the call is handled expeditiously, you should configure multiple routes for 911 calls.

» Enhanced emergency routing: Because of the mobile workforce that exists today, enterprises must be able to effectively route emergency calls to the nearest public service answering point (PSAP). Undoubtedly, the biggest challenge to routing 911 calls with a mobile workforce is knowing where the calling person is located.

The Enhanced 911 (E911) emergency routing V5 interface specified in the National Emergency Number Association (NENA) i2 architecture defines the requirements. E911 is mostly associated with service providers and wireless providers. It enables location services for SIP and/or wireless callers that access the network from various remote locations, or may be otherwise using a number that isn’t associated with the geographic location in which they’re actually located. To make emergency services available to callers not tied to fixed geographic locations, E911 tracks caller location and the PSAP (that is, the emergency services dispatcher) nearest to them by means of a voice over IP (VoIP) Positioning Center (VPC). A VPC maintains a database of the current locations of callers.

» Time-based routing: With time-based routing, enterprises can create routing profiles based on time of day (ToD), day of week (DoW), holidays, special days, and for time zones. For example, a business operating under a “follow the sun” customer support model might create ToD profiles to automatically route incoming calls to different call centers around the world, corresponding to normal business hours in that region, to provide 24/7 support. Similarly, routing profiles can be created to route calls differently based on the day of the week (for example, weekend calls might be sent to an answering service or voicemail), holiday schedules, and other special days, such as Mother’s Day.



» Business continuity: Enterprises can ensure business continuity in their RTC networks by configuring alternate routing paths (for example, over different SIP trunks or service provider networks) in case of outages or telephony denial-of-service (TDoS) attacks. Options include the following:

• Overflow routing: Each TG in a network can be assigned an overflow number. In addition, a list of overflow routes can be provisioned by using nested routing labels. Overflow routing may be appropriate for some emer-gency call situations as well as other routing scenarios.

• Temporary alternate routing (TAR): TAR allows you to add and/or replace routes. This can be used to temporar-ily change the normal routes to alleviate network congestion or handle network outages.

» Calling forced routing: Calling forced routing executes routing to the internal number based on the originating number and inbound rules set on the system. When calling forced routing is applied, all calls are routed using the provisioned routing label, independent of the called number.

CHAPTER 4 Exploring Intelligent Session Control Use Cases 27


Exploring Intelligent Session Control Use Cases

In this chapter, you explore real-world use case scenarios for deploying a centralized policy and routing server in an enterprise.

Centralized Policy and RoutingToday’s enterprises strive to improve customer service and make employee collaboration more efficient. Optimizing the use of net-work resources and simplifying network operations with intel-ligent session and routing control solutions can help enterprises reach this goal.

A centralized policy and routing server (see Figure 4-1) brings together heterogeneous multivendor environments and enables multiple private branch exchanges (PBXs) to communicate, thereby simplifying and centralizing the routing infrastructure.

Chapter 4

IN THIS CHAPTER

» Centralizing call routing

» Showing the interworking multivendor dial plans

» Reducing long-distance toll charges

» Avoiding SIP trunk failures with failover routes

» Leveraging Microsoft Active Directory



With this deployment model, you can improve your enterprise real-time communication (RTC) infrastructure by applying vari-ous network and unified communications policies. Advanced fea-tures include E.164 Number to URI Mapping (ENUM) and quality of experience based routing.

Key benefits of centralized policy and routing include the following:

» Consolidate and centralize policy, routing, and other information in a single master server, dramatically reducing operating expenses (OPEX) and management complexity, while allowing for easier and more efficient dial plans.

» Leverage IP-based enterprise networks for Voice over IP (VoIP) calls, eliminating national and international long- distance toll charges between employees anywhere in the world.

FIGURE 4-1: Centralized policy and routing for Session Initiation Protocol (SIP) trunking.



» Avoid downtime from a cut line or hardware failure, by rerouting traffic automatically over the IP-based wide area network (WAN) in the event of a single SIP trunk failure at an office or call center location.

Dial Plan InterworkingMedium- to large-sized enterprises have evolved in a similar manner to service providers. The flurry of merger and acquisi-tion activity within large financial institutions and healthcare companies has resulted in networks with a diverse variety of net-working elements that provide security and call processing for real-time voice calls. For example, you might see a network with SBCs and IP PBXs from multiple vendors. The network may have multiple different session control devices as well. In addition to all of this, the enterprise may be planning to deploy Skype for Business, Slack, Jabber, or WebEx — either on-premises or in a hosted environment.

It is an operational nightmare to manage and provide new ser-vices on this sort of “un-unified” network. Service-level agree-ments (SLAs) are a constant problem where users’ issues aren’t resolved quickly and new business services can’t be launched on time due to incompatibilities between the different networks.

Enterprises need to centralize their architectures to handle this complexity. A centralized policy and the routing server provide a unified view of the network and make it easy to manage and monitor the network. New services can be launched quickly, as the centralized routing engine takes care of normalization, and varied dial plans can be seamlessly integrated. No “forklift” replacements are needed on the dial plan or the devices and the enterprise’s past investments are protected. You can see an example of the centralized dial plan in Figure 4-2.



International Toll AvoidanceA centralized policy server allows enterprises not only to create a centralized dial plan but also to leverage their own IP-based enterprise WAN for VoIP calls, eliminating national and interna-tional long-distance toll charges between employees anywhere in the world (see Figure 4-3).

A properly provisioned policy and routing server will have network-wide visibility in terms of high-quality routes, least cost routes, on-net and off-net routes, alternate routes, and failed routes. Using this visibility, the policy and routing server can decide, on a per-call basis, what the best way is to route a call based on the configured policy.

For enterprises that span multiple countries, a centralized policy and routing server can help reduce toll charges by keeping calls on-net and only breaking out to the public switched telephone network (PSTN) if necessary. Typically, enterprises save about 55 to 75 percent on toll charges with a centralized policy and routing server.

FIGURE 4-2: A centralized dial plan.



Rerouting TrafficA centralized policy and routing server also enables network fail-ure avoidance for higher reliability in the enterprise network. The policy server quickly reroutes calls around SIP trunk failures to ensure service availability (see Figure 4-4).

Advanced centralized policy and routing servers have a variety of traffic management controls and routing redundancy features such as

» Traffic controls that include

• Calls to and from destination and origination numbers or trunk groups controlled or throttled based on the traffic situations in the network

• New calls to and from trunk groups gracefully terminated to relieve congestion

• Call bandwidth reserved directionally so network conges-tion can be handled in a fair and graceful manner in both directions of the network element under congestion

FIGURE 4-3: Reduce toll costs by minimizing call routes over expensive service provider networks.



• Adaptive overload controls available on the policy server that kick in when the policy server itself is congested; the system can be set up such that calls can be processed even during extreme congestion situations

» Routing redundancies to include

• Temporary alternate routes configured in the system; routes used when there are issues with the primary routes

• Overflow routes setup to be used when the primary routes can’t handle the increased traffic volume

• Proportional routing used to balance traffic across multiple egress points

Microsoft Active Directory Routing and Translation

Microsoft Active Directory (MSAD) is used by most enterprises as a centralized provisioning point for their employees’ access and

FIGURE 4-4: A centralized policy and routing server.



use of the network. When combined with a centralized policy and routing server, enterprises can create a powerful RTC solution for their employees.

Enterprise user information can be retrieved by a centralized policy and routing server from MSAD to provide the following services:

» Calls to the employee/user can be forwarded to another number (for example, home, mobile, or office) of the user based on configured policy.

» Caller display name can be populated based on MSAD attributes, such as common name, display name, or user principal name.

» Determine the users of Skype for Business/Lync voice lines and route calls appropriately.

In enterprise deployments, a centralized policy and routing server retrieves the user information from the MSAD database and uses this information for call routing and policy decisions. This reduces the administrative overhead of provisioning and managing user information in two places.

Figure 4-5 shows the MSAD support function in a centralized policy and routing server deployment.

In this example, the centralized policy and routing server retrieves the MSAD user data (such as common name, display name, user-PrincipalName, msRTCSIP-Line, telephoneNumber and mobile) from the domain controller and stores it in a database. During call processing, Lightweight Directory Access Protocol (LDAP) queries are made against the local cached user data. The following are translation and routing examples:

» Forwarding the calls to other numbers of the user based on policy: The routing engine translates the dialed number with the user’s home, mobile, or office phone numbers for completing the calls, based on policy configuration.



» Populating caller display name: The centralized policy and routing server queries the display name based on the MSAD user’s mobile or home phone number and sets it in the policy response. Therefore, the called party can see the caller name, even when caller name isn’t set by the public switched telephone network (PSTN).

» Determining if the MSAD user has a Lync or Skype for Business voice line: The centralized policy and routing server queries user information to determine whether the user has Lync voice (msRTCSIP-Line) or is being served by another private branch exchange (PBX) infrastructure. This information is then used to route the calls appropriately to Microsoft Lync, Skype for Business, or a third-party PBX.

FIGURE 4-5: Leveraging MSAD.

CHAPTER 5 Ten Reasons to Choose Ribbon’s PSX for Intelligent Session Control 35


Ten Reasons to Choose Ribbon’s PSX for Intelligent Session Control

Centralized intelligent session and routing control solutions enable enterprises to substantially reduce both their com-munications capital and operating expenditures (CAPEX

and OPEX, respectively) while still improving the richness and quality of their communications services. In this chapter, we explain ten benefits of Ribbon’s PSX for intelligent session and routing control.

Chapter 5

IN THIS CHAPTER

» Lowering maintenance, management, long-distance, and routing costs

» Getting the most out of your employees and equipment

» Improving reliability in your network

» Simplifying migrations from legacy equipment

» Looking at virtualization and access control security



Reduce PBX Management and Maintenance Costs

Two-thirds of enterprises have private branch exchanges (PBXs) from more than a single vendor. Large enterprise networks may contain a dozen or more PBXs spread out around the world at var-ious office locations, with each PBX requiring a specialized engi-neer to provision and update dial plans, routing, and subscriber information for that office.

With Ribbon’s PSX, enterprises can consolidate and centralize dial plans, routing, and subscriber information in a single master server, dramatically reducing both the OPEX and the management complexity associated with PBX-based information manage-ment while allowing for an easier and more efficient dial plan to manage.

Reduce Long-Distance CostsRibbon’s PSX allows enterprises to leverage their own IP-based enterprise network for Voice over IP (VoIP) calls, eliminating national and international long-distance toll charges between employees anywhere in the world.

Ribbon Communications policy solutions have reduced recurring long-distance phone charges by 55 to 75 percent for a multina-tional enterprise customer.

Route Calls Based on ValueValue-based routing allows an enterprise to intelligently route calls based on lowest cost, combined with the highest quality of service (QoS) in the network. Value-based routing techniques, such as least cost routing (LCR), have been proven to reduce OPEX costs by over 20 percent and improve margins by 10 percent.

Ribbon’s PSX enables faster loading and analysis of daily rate sheets that can be combined with QoS-based real-time metrics in a centralized provisioning and distributed processing architec-ture, which enables fast and simple network deployment.

CHAPTER 5 Ten Reasons to Choose Ribbon’s PSX for Intelligent Session Control 37


Increase Employee ProductivityCentralized Session Initiation Protocol (SIP) policy management is the key to truly unified communications (UC). Ribbon’s PSX allows enterprises to effectively consolidate messaging systems and unify business applications so employees can get more done more quickly. On average, employees can save 1.21 hours per day by using UC to communicate and collaborate.

Sweat Your Existing AssetsWith Ribbon’s PSX solution, enterprises need not spend money upgrading PBXs and other legacy systems to stay compatible with the latest SIP stacks and applications. The robust signaling inter-working of the Ribbon PSX takes care of the compatibility for you by providing a single routing and policy system for both legacy (for example, Time-Division Multiplexing, or TDM, and H.323) and newer SIP systems.

Make Your Network More ReliableRibbon’s PSX can act as a SIP proxy and redirect server, allowing enterprises to reroute traffic automatically over their IP-based network in the event of a single SIP trunk failure at an office or call center location. This feature prevents an outage at an office or call center due to a cut line or a hardware failure, allowing enterprises to offer communications services with “five-nines” (99.999 percent) reliability.

Improve Security with Access ControlAs organizations move to SIP for their UC systems, the opportu-nity for malicious users to access and steal services (such as toll fraud) increases, unless a Ribbon PSX is deployed. With Ribbon Communications, enterprises can focus on their core competen-cies without fear of theft on their networks, by limiting access to the UC platforms and stopping toll fraud.



Virtualize Infrastructure and Network Functions

The Ribbon PSX can be deployed as a fully virtualized, cloud-native solution that provides exceptional scalability and centralized policy/traffic control as a virtualized network function (VNF) format. The Ribbon PSX supports Intel-based hardware and third-party hypervisors to fit seamlessly into any service provider’s or enterprise’s Network Function Virtualization (NFV) strategy.

Integrate with Third-Party SolutionsWith an extensible design, the Ribbon PSX can be deployed with third-party session border controllers (SBCs), gateways, and SIP application servers, to provide the “gold standard” for intelligent network policy and routing management services.

Leverage Extensive Features and Capabilities

Ribbon Communications has well over a decade of intelligent policy and session management principles for enterprises. With the Ribbon PSX solution built around a centralized policy and dis-tributed architecture, it supports a wide range of services to intel-ligently process calls, including toll bypass, voice virtual private networks (VPNs), Electronic Number Mapping System (ENUM)/Intelligent Network Application Part (INAP) lookups, blacklisting/whitelisting, and regulatory services, such as E911, Government Emergency Telecommunications Service (GETS), Multi-Level Precedence and Preemption (MLPP), and lawful intercept.

The Ribbon PSX functions as a centralized call routing engine in heterogeneous voice networks including SIP, H.323, TDM, and so on. Its highly redundant architecture and live upgrade capa-bilities deliver 99.999 percent availability — even during peak loads of millions of calls per hour.

Glossary 39


Glossary

2G: Commercially introduced in 1991 and based on GSM, the second generation of wireless telecommunications technology enabled digital data services for mobile, notably SMS text messages. See also Global System for Mobile Communications (GSM).

Adaptive Multi-Rate Wideband (AMR-WB): A wideband speech audio codec standard that provides improved speech quality due to a wider speech bandwidth (50 to 7000 Hertz) compared to narrowband speech codecs (300 to 3400 Hertz). See also codec.

Answer Seizure Ratio (ASR): The relationship between the number of seizures that result in an answer signal and the total number of seizures.

Average Call Hold Time (AHT): The average length of time between the moment a caller finishes dialing and the moment the call is answered or terminated. This is primarily used in call centers and is a key factor when determining staffing requirements of call centers.

Average Packet Loss Percentage (PLP): The average number of lost packets to the total number of transmitted packets.

Breakout Gateway Control Function (BGCF): A SIP proxy that processes requests for routing from an S-CSCF when the S-CSCF has determined that the session cannot be routed using DNS or ENUM/DNS. See also Session Initiation Protocol (SIP), Serving Call Session Control Function (S-CSCF), and Domain Name System (DNS).

Call Completion Ratio (CCR): The ratio of successfully completed calls to the total number of attempted calls.

Calling Party Category (CPC): An ISUP parameter that defines the origination of a call (such as pay phone, data terminal, ordinary subscriber) and a language indicator that lets operators know which language to use. See also Integrated Services Digital Network User Part (ISUP).



Class of Service (COS): Provides a way for service providers to define a common set of services that can be applied to various call processing entities.

Code Division Multiple Access (CDMA): A channel access method used by various telecommunications technologies to enable multiple transmitters to simultaneously send traffic over a single communication channel.

codec: A device or computer program for encoding or decoding a digital data stream or signal.

denial of service (DoS) attack: A cyberattack in which the objective of the attacker is to make a system or network resource unavailable to authorized users.

dial-through fraud (DTF): A form of toll fraud in which an IP PBX is compromised in such a way that an attacker using a robocall generator can dial in to the PBX, get a dial tone, then hairpin dial out to an interna-tional premium number to generate fraudulent revenue that is charged to the target enterprise. See also Internet Protocol (IP) and private branch exchange (PBX).

direct inward dial (DID): A feature offered by telephone companies to subscribers who operate a PBX for multiple telephone numbers over on or more analog or digital phone circuits to the PBX. See also private branch exchange (PBX).

Domain Name System (DNS): A decentralized hierarchical database for computers, services, and other resources connected to a network or the Internet that provides mapping of numerical IP addresses to domain names, as well as other information. See also Internet Protocol (IP).

E.164: An ITU Telecommunication Standardization Sector (ITU-T) recommendation that defines a numbering plan for the worldwide PSTN and some data networks. See also International Telecommunication Union (ITU) and public switched telephone network (PSTN).

E.164 Number to URI Mapping (ENUM): A system of unifying the international telephone number system of the PSTN with Internet addressing and identification name spaces. See also E.164, Uniform Resource Identifier (URI), and public switched telephone network (PSTN).

European Telecommunications Standards Institute (ETSI): An independent, not-for-profit standardization organization in the telecommunications industry.

Glossary 41


G.722: An ITU Telecommunication Standardization Sector (ITU-T) standard for a 7 kilohertz wideband audio codec operating at 48, 56, and 64 kilobits per second. See also International Telecommunication Union (ITU) and codec.

G.722.1: An ITU Telecommunication Standardization Sector (ITU-T) standard audio codec providing high quality, moderate bit rate (24 and 32 kilobits per second) wideband (50 hertz to 7 kilohertz) audio band-width, 16 kilo-samples per second audio coding. See also International Telecommunication Union (ITU) and codec.

G.729.1: An ITU Telecommunication Standardization Sector (ITU-T) standard for an 8 to 32 kilobit per second embedded speech and audio codec. See also International Telecommunication Union (ITU) and codec.

Global System for Mobile Communications (GSM): An ETSI standard to describe the protocols for 2G digital cellular networks used by mobile phones. See also 2G and European Telecommunications Standards Institute (ETSI).

H.248: An ITU Telecommunication Standardization Sector (ITU-T) specification of the media gateway control protocol architecture for providing telecommunication services across a converged internetwork consisting of the traditional PSTN and packet-switched networks. See also International Telecommunication Union (ITU) and public switched telephone network (PSTN).

H.323: An ITU Telecommunication Standardization Sector (ITU-T) specification that defines the protocols to provide audio-visual commu-nication sessions on any packet network. See also International Telecommunication Union (ITU).

Integrated Services Digital Network (ISDN) User Part (ISUP): Part of SS7 used to set up telephone calls in the PSTN. See also Signaling System No. 7 (SS7) and public switched telephone network (PSTN).

International Telecommunication Union (ITU): A United Nations (UN) agency responsible for coordinating global cooperation for information and communication technology issues.

Internet Protocol (IP): The principal communications protocol in the TCP/IP communications suite for routing across network boundaries (routers) and the Internet. See also Transmission Control Protocol (TCP).

IP version 4 (IPv4): The version of the Internet Protocol (IP) most in use today based on a 32-bit IP address. See also Internet Protocol (IP).



IP version 6 (IPv6): The version of the Internet Protocol (IP) that is replacing IP version 4. IPv6 is based on a 128-bit IP address that will allow an infinitely higher number of devices than IPv4 to be connected to the Internet. See also Internet Protocol (IP) and IP version 4 (IPv4).

IP Multimedia Subsystem (IMS): An architectural framework for delivering IP multimedia services over an IP packet-switched network. See also Internet Protocol (IP).

Jitter: The variation in the time between packets arriving, caused by network congestion, timing drift, or route changes in Voice over IP (VoIP). Jitter is typically measured in microseconds or nanoseconds and should be less than 100 milliseconds between the starting and final point of the communication.

least cost routing (LCR): The process of selecting the path for out-bound communications traffic based on cost.

Mean Opinion Score (MOS): The arithmetic mean value of user opinion for rated voice quality (1 — Lowest, 5 — Highest).

Minutes of Use (MOU): This is a volume commitment from the Buyer Side to the Seller Side to route the committed amount of calls in terms of volume, typically on a monthly basis.

Network Effectiveness Ratio (NER): The percentage of calls success-fully completed. Unlike CCR, NER includes calls terminated by Busy and No Answer signals. See also Call Completion Ratio (CCR).

private branch exchange (PBX): A multiline phone system used by businesses.

Post Dial Delay (PDD): The length of time that is consumed when a phone user dials the last digit in a phone number and when the user hears the ring or busy tone of the phone number being called.

public switched telephone network (PSTN): The aggregate of the world’s circuit-switched telephone networks operated by national, regional, and local telephony operators.

service-level agreement (SLA): A contractual agreement between a service provider and its customers (internal or external) that defines the level of service to be provided in terms of performance, latency, reliability, availability, and other objective metrics, as well as the remedies available to the customer if the service provider fails to meet its service level commitments.

Glossary 43


Serving Call Session Control Function (S-CSCF): A SIP server that performs session control and is the central node of the signaling plane in an IMS core network. See also Session Initiation Protocol (SIP) and IP Multimedia Subsystem (IMS).

session border controller (SBC): A device deployed in a VoIP network to control signaling and media streams involved in the setting up, conducting, and tearing down of telephone calls and other interactive media communications. See also Voice over IP (VoIP).

Session Initiation Protocol (SIP): A signaling protocol designed to carry voice, data, and video transmissions over IP networks rather than on public switched telephone networks (PSTNs).

Signaling System No. 7 (SS7): A set of telephony signaling protocols used to set up and tear down most of the world’s PSTN telephone calls. See also public switched telephone network (PSTN).

SILK: An audio compression format and audio codec developed by Skype Limited. See also codec.

Special Access Code (SAC): A service offered by the PSX that causes calls to be routed based on 700, 800, and 900 number access codes (different from call types) for routes provisioned on a class of service (CoS) entity.

Temporary Alternate Routing (TAR): A feature that allows you to add and/or replace routes to temporarily change the normal routes to alleviate network congestion or circumvent a network outage.

time-division multiplexing (TDM): A method of transmitting and receiving independent signals over a common signal path by means of synchronized switches at each end of the transmission line so that each signal appears on the line only a fraction of time in an alternating patter.

Transmission Control Protocol (TCP): One of the core protocols of the Internet Protocol suite, TCP is one of the two original components of the suite, complementing the Internet Protocol (IP), and therefore the entire suite is commonly referred to as TCP/IP. TCP provides reliable, ordered delivery of a stream of bytes from a program on one computer to another program on another computer. TCP is the protocol that major Internet applications such as the World Wide Web, email, remote administration, and file transfer rely on. See also Internet Protocol (IP).

trunk group (TG): A group of communication trunks with the same purpose. For example, multiple DID trunks are commonly grouped together in a trunk group. See also direct inward dial (DID).



unified communications (UC): Integrated communication services such as audio/video/web conferencing, desktop sharing, instant messaging, mobility features, presence information, and voice, among others.

Uniform Resource Identifier (URI): A string of characters used to identify a resource.

voice over IP (VoIP): Technology that enables delivery of voice commu-nications and multimedia sessions over IP networks. See also Internet Protocol (IP).

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