Clearspan Enterprise Guide

ENTERPRISE SOLUTIONS GUIDE
2739-001
Release 14.0

2811 Internet Blvd.
Frisco, Texas 75034-1851
Tel +1 469 365 3000
Tel + 1 800 468 3266
WWW.AASTRAINTECOM.COM

Document Revision History

Release Version Reason for Change Date Author
14.0 1 Aastra Intecom Rebranded April 26, 2007 D.Woelfle

CLEARSPAN ENTERPRISE SOLUTIONS GUIDE AASTRA - 2739-001
2007 AASTRA NTECOM PRELIMINARY PAGE 2 OF 55

Copyright Notice

Copyright © 2007 Aastra USA Inc.
All rights reserved.
Any technical documentation that is made available by Aastra USA Inc. is proprietary and
confidential and is considered the copyrighted work of Aastra USA Inc.
This publication is for distribution under Aastra USA Inc. non-disclosure agreement only.
No part of this publication may be duplicated without the express written permission of
Aastra USA Inc., 2811 Internet Blvd, Frisco, TX 75034.
Aastra USA Inc. reserves the right to make changes without prior notice.

Trademarks

Aastra® is a registered trademark of Aastra Technologies, Ltd.
BroadWorks® is a registered trademark of BroadSoft, Inc.
Printed in the United States of America.


Table of Contents

1 Purpose and Audience .......................................................................................... 9

2 Overview ................................................................................................................ 9
2.1 What is Clearspan? ........................................................................................9
2.2 Why choose Clearspan?................................................................................9
2.3 What is in the Clearspan Enterprise Solution? .......................................... 11
2.4 Key Business Applications.......................................................................... 12
3 Standard Deployment Models.............................................................................13
3.1 Single Site Deployment Model.................................................................... 13
3.2 Multi-Site Deployment Model...................................................................... 15
4 Key User Applications .........................................................................................20
4.1 Clearspan Communicator Family ............................................................... 20
4.1.1 Clearspan Communicator ................................................................. 20
4.1.2 Clearspan Communicator - Multimedia............................................ 21
4.2 Clearspan Assistant Family......................................................................... 22
4.2.1 BroadWorks Assistant – Enterprise.................................................. 22
4.2.2 Clearspan Assistant - Mobile ............................................................ 24
4.3 Personal Web Portal.................................................................................... 25
4.4 Unified Messaging ....................................................................................... 27
4.5 Contact Centers........................................................................................... 28
4.6 Auto Attendants ........................................................................................... 30
4.6.1 Clearspan Receptionist ..................................................................... 30
4.7 Web Conferencing....................................................................................... 32
4.8 Clearspan Deployment Studio .................................................................... 34
5 Integrating with Legacy Equipment.....................................................................34
5.1 PBX Solution (SIP Trunking)....................................................................... 34
5.1.1 Network Layout.................................................................................. 34
5.1.2 Services.............................................................................................. 35
5.1.3 Services for PBXs without DID ......................................................... 36
6 Clearspan Service Delivery Platform ..................................................................39
6.1 Network Elements ....................................................................................... 39
6.1.1 Application Server.............................................................................. 39
6.1.2 Network Server .................................................................................. 40
6.1.3 Media Server...................................................................................... 41
6.1.4 Web Server ........................................................................................ 42
6.1.5 Conference Server............................................................................. 43
6.2 Call Scenarios.............................................................................................. 44
6.2.1 Registration ........................................................................................ 44
6.2.2 Call Origination................................................................................... 44


6.2.3 Call Termination................................................................................. 44
6.2.4 Entry / Exit Point................................................................................. 45
6.2.5 Media Resources............................................................................... 45
7 Access Devices ...................................................................................................45
7.1 Access Gateways........................................................................................ 46
7.2 SIP Phones .................................................................................................. 46
7.3 Soft clients.................................................................................................... 46
8 Network Gateways ..............................................................................................46

9 Network Infrastructure Components ...................................................................47
9.1 LAN Requirements ...................................................................................... 47
9.2 Using DHCP................................................................................................. 49
9.3 Using DNS ................................................................................................... 49
9.4 Using TFTP.................................................................................................. 50
10 Operations, Administration, Maintenance and Provisioning .......................50
10.1 Configuration and Provisioning ................................................................. 50
10.1.1 Initial System Configuration............................................................. 51
10.1.2 Creating an Enterprise..................................................................... 53
10.1.3 Creating a Group ............................................................................. 54
10.1.4 Configuring Translations and Routing ............................................ 55
10.1.5 Bulk Provisioning ............................................................................. 55


Figure 1 Fully Integrated Communication Suite .................................................................................. 10
Figure 2 Enterprise Solution ................................................................................................................... 12
Figure 3 PSTN Connectivity ................................................................................................................... 14
Figure 4 Clearspan interface to PSTN ................................................................................................... 15
Figure 5 Distributed Call Control Model................................................................................................ 16
Figure 6 Clearspan Call Control Model.................................................................................................. 17
Figure 7 Clearspan Multi-site Deployment Model................................................................................ 18
Figure 8 Clearspan Communicator Multimedia Menu ........................................................................ 21
Figure 9 Clearspan Call Center - Agent ................................................................................................. 23
Figure 10 Clearspan Call Center - Agent Menu.................................................................................... 23
Figure 11 Clearspan CommPilot Personal Portal................................................................................ 25
Figure 12 Clearspan Call Manager ......................................................................................................... 26
Figure 13 Clearspan Contact Center Menu........................................................................................... 28
Figure 14 Clearspan Contact Center...................................................................................................... 29
Figure 16 Clearspan Conference............................................................................................................ 33
Figure 17 Clearspan Conference Call Menu ......................................................................................... 33
Figure 18 Clearspan Network Layout .................................................................................................... 35
Figure 19 Application Servers................................................................................................................. 40
Figure 20 Network Server ........................................................................................................................ 41
Figure 21 Media Server............................................................................................................................. 42
Figure 22 Web Server ............................................................................................................................... 43
Figure 23 Conference Server .................................................................................................................. 43


Part I – Solution Overview


1 Purpose and Audience

The Clearspan Enterprise Solutions guide is intended for enterprise telecom
administrators or those who support enterprise communications networks (system
administrators and system integrators). This document describes how Clearspan is
deployed as a converged communications platform in the enterprise. It is designed to
advise implementers about the advantages, processes and requirements of deploying
centralized voice services architectures.
At the core of the Clearspan solution is BroadWorks, an open, standards-based, highly
resilient and easily managed software suite, capable of elegantly scaling to millions of
users. Deployed on industry standard server platforms, the solution delivers lower total
cost of ownership (TCO) by lowing implementation, operating, support and maintenance
costs. IT and Telecom managers will enjoy an easy-to-provision and easy-to-manage
solution that is extensible with many 3rd party applications, such as CRM and Microsoft
LCS.
In addition to offering a converged environment, Clearspan addresses an additional set of
problems inherent in traditional networks. Using the web, Clearspan provides a high
servicing component between the enterprise administrator and the user. User control and
configuration is provided via a standard web portal that augments basic phone
functionality to make existing services (for example, conference calling and call
forwarding) easier to use, and new services easier to deploy. For example, a user can
selectively forward calls from an office phone to a cell phone with a few clicks on a web
page or wireless PDA.
The Clearspan platform operates on a standards-based, modular architecture that uses
common protocols (such as SIP), open interfaces, and scalable, industry-standard
hardware. The open environment enables enterprises, as well as third-party service
developers, to rapidly introduce new features and launch new applications.

2 Overview

2.1 What is Clearspan?
Clearspan is an enhanced unified communications system that allows large enterprises to
offer advanced communications applications over a Voice over IP (VoIP) network. VoIP
networks consist of network elements that provide packet conversion, call control, and
applications. Clearspan provides call control and enhanced services for enterprise users. It
is developed on a standards-based, modular architecture utilizing common protocols,
open interfaces, and scalable, industry-standard hardware.

2.2 Why choose Clearspan?
There are many compelling reasons why an enterprise should choose Clearspan IP-PBX
communications platform:
A Fully Integrated Communications Suite
Clearspan provides a complete suite of communications applications. It offers not only
classic PBX-like services, but also a rich set of enhanced services including unified
messaging, interactive voice response, business conferencing and contact center. These
services are integrated into a single platform with a common web-enabled end-user


interface, eliminating the need to deploy and integrate dedicated communications
platforms for each application.

N-Way Conferencing
Vendor D

Unified Messaging IVR - Vendor B
Vendor C

PBX - Vendor A

• Software-based solution
• Hardware-based multi-vendor solution
• Proprietary vendor hardware
• Off-the-shelf hardware
• Custom platform for each application • Single platform for all applications
• Poor user experience • Integrated user experience
• Proprietary graphical user interfaces • Web-based access
• Limited extensibility • Extensible service platform
• Proprietary legacy CPE • Standard SIP-based CPE

Figure 1 Fully Integrated Communication Suite

A Scalable Enterprise Communications Solution
Clearspan is a flexible unified communicatoins platform designed to meet the demanding
requirements of today’s large enterprises. At its core is a highly scalable call processing
architecture capable of serving enterprises as small as a thousand users, as well as the
ability to grow to a network serving hundreds of thousands of users without sacrificing
ease of management or reliability.
Single Site or Multi-Site Deployment Models
With Clearspan, service delivery is not bound to a physical site. Users can be deployed in
a simple single site network, or distributed across a multi-site enterprise without impact to
service functionality, capacity or performance. Clearspan can be deployed at a single site,
yet support users distributed across hundreds or even thousands of branch locations. To
better manage calls across a large multi-site enterprise, Clearspan includes a variety of
powerful site-based call routing and virtual private networking policies.
Disaster Proof Reliability
All Clearspan network elements can be deployed across geographically distinct locations
throughout the corporate communications network. This allows enterprises to deploy a
fully redundant data center in two geographically separate sites. In the event that a major
disaster disables the primary data center, end-users will still have full access to their
service profile through the redundant data center. With Clearspan, enterprises are
provided with carrier level five 9’s reliability.

An Open, Standards Based Platform


Clearspan does not use any proprietary interfaces. When interfacing with access
equipment, Clearspan uses the Session Initiation Protocol (SIP). This allows enterprises
to choose from a broad range of standards-based phones, terminals, gateways and
peripherals that best meet their budget and end-user requirements. Finally enterprises
can choose a communications solution without fear of “vendor lock-in”.
Clearspan is also an open services platform, offering a rich set of application programming
interfaces that enable real-time computer telephony integration with enterprise
applications. Unlike legacy communications platforms, these interfaces leverage simple
XML-based transaction models allowing enterprises to rapidly integrate both real-time call
control and service management applications.

2.3 What is in the Clearspan Enterprise Solution?
At its foundation, a Clearspan based enterprise unified communications solution includes
the service delivery platform built around BroadSoft BroadWorks. This platform is
comprised of four separate network elements: the network server, the application server,
the media server and the web server. If conferencing is required, a conferencing server is
also included. These elements work together to deliver core call control, enhanced
applications and service logic. Typically Clearspan is deployed in the enterprise data
center. For high availability and disaster recovery solutions, the network elements can be
deployed across two physically separate data centers without impact to performance,
manageability or scalability.


Figure 2 Enterprise Solution

To deliver these services to the end user, Aastra also incorporate session border
controllers, access devices and network gateways. End-user access devices include
access gateways, SIP phones and soft clients. Network gateways include PSTN
gateways using either FXO, CAS, or PRI based interfaces.
Aastra has assembled a set of “enterprise configuration kits” that itemize vendor-specific
packages of access and network equipment that can be reliably deployed with Clearspan.
Each kit is tested and validated so that it integrates seamlessly, providing a complete
communications solution.

2.4 Key Business Applications
The flexibility of the Clearspan solution, coupled with Aastra’s technology and partners,
enables enterprises to address a wide range of applications. Along with the following


applications, enterprises may easily develop their own applications, features and
integrations with a commercially available software development environment.
The applications most often considered are:
Converged Communications – maximize end-user productivity by providing them a
fully integrated communications environment, enabling smooth interaction across
individuals and groups. With its tight integration with Outlook, click-to-dial, voice and
video, unified messaging, call center and conferencing, Aastra provides a ubiquitous
communication experience, independent of the endpoint device used.
Fixed/Mobile Convergence (FMC) - Whether users are using mobile or stationary
phones, they share the same experience. Because the features are separated from
the endpoints, Aastra can easily integrate any type of device, without additional user
licensing expense. With Aastra’s FMC solution, there is no reason to involve the
cellular carrier as all features supplied to the mobile client are independent of carrier
used.
Business (SIP) Trunking - allows enterprises to provide “IP Trunking” connectivity to
existing PBXs. Enterprises may deploy next-generation trunking and overlay features
while preserving legacy PBX investments and slowly migrating all locations to VoIP.
Aastra provides a best-in-class set of integrated features and end-user clients that anchor
each of these applications. Users experience a unified service regardless of location or
endpoint used.

3 Standard Deployment Models

Clearspan supports a variety of deployment models – from single site deployments to
geographically dispersed multi-site deployments.

3.1 Single Site Deployment Model
The picture below depicts a typical Clearspan single site deployment. In this example, a
large single building enterprise has deployed Clearspan to offer all voice services in the
network. The building has a standard LAN configuration with one or more wiring closets
on each floor. In each wiring closet are rack mounted Fast-Ethernet (100Base-T) layer 2
access switches – enough to serve all stations on the floor.
Access Device Connectivity
Not all desktops have to have SIP phones; legacy analog equipment is also supported. In
this example, a line access gateway is deployed in wiring closets on floors that have
analog handsets. The line access gateway serves to convert standard analog phones to
SIP signaling and RTP media traffic over the IP network. The line gateway connects to
the local Fast Ethernet switch in much the same way as an IP phone. If only a few analog
phones exist, an Integrated Access Device (IAD) may also be used.
Each wiring closet is interconnected to the core enterprise LAN using Gigabit Ethernet
where traffic is backhauled to the service infrastructure.
Service Infrastructure Deployed in Centralized Server Room
A fully redundant Clearspan system is deployed in a datacenter with a high-capacity IP
connection to the core LAN. Often this is the same room that houses all of the enterprise
data applications.
PSTN Connectivity


Also in the datacenter are redundant PSTN gateways. The gateways connect to the local
PSTN switch using standard TDM interfaces – typically one or more T1s in North America,
or E1s outside of North America. The PSTN gateways connect to a layer-2 switch using
either a Fast Ethernet or Gigabit Ethernet connection. Clearspan controls all signaling to
and from the PSTN gateways.

Figure 3 PSTN Connectivity


Clearspan Manages All Call Control
The picture below shows the system interfaces for a typical outgoing call to the PSTN. In
this example, the user employs a web-based click-to-dial application to originate the call. .

Figure 4 Clearspan interface to PSTN

It is important to note that all call control is handled directly by Clearspan. Signaling with
the SIP phones is carried over the IP network using the standards-based Session Initiation
Protocol (SIP). Clearspan also uses SIP to signal with the PSTN gateways to setup the
outbound call. Once Clearspan has established the call, RTP media is exchanged directly
between the SIP phone and the gateway.

3.2 Multi-Site Deployment Model
Issues with Large Multi-Site PBX Solutions
Although Clearspan easily supports large single-site enterprises, it is often chosen over a
classic PBX or IP-PBX solution for its advantages in large multi-site deployments. Many
traditional PBX solutions suffer from a number of issues when deployed in multi-site
enterprises:
• Capacity – multi-site enterprises are usually very large enterprises often as big as
a 100,000 users. Some traditional PBX solutions can typically handle up to 2500
users per system. Once this capacity limit has been reached, another system
must be deployed.
• Scalability – when a traditional PBX system has reached capacity, another
system must be deployed in the network. Adding multiple call control elements to
a voice network makes routing and translations rules complicated – and often it
requires deploying a full mesh network between all the PBX systems.
• Feature Limitations – some traditional PBX solutions are not designed to
seamlessly deliver features to users across multiple sites. Often features across
multiple sites have limitations or simply are not supported.
• Manageability – legacy PBX solutions are hard to manage as they often lack a
centralized multi-system management interface.
• Availability – some traditional PBX solutions have very weak redundancy
solutions – especially when applied to a multi-site enterprise. Usually,


redundancy requires spares for each line card, so a fully redundant PBX solution
would require separate spares per site. This is an extremely expensive solution,
and is also difficult to inventory and manage.
• Cost – the cost of deploying and managing multiple PBX systems in a distributed
multi-site network is high.
Disadvantages of a Distributed Call Control Model
This approach of adding PBX systems to the network in order to meet capacity can be
characterized as a “distributed call control model”. Often these systems have geographic
limitations that require the PBX system to be deployed close to the devices that it controls.
Consider a retail enterprise case study consisting of 1600 sites as depicted in the diagram
below:

Figure 5 Distributed Call Control Model

In this example, a popular IP-PBX solution was analyzed. To meet the requirements, the
solution required 33 separate call control data centers, each with their own distinct PBX
system. Each of the 33 PBX systems then needed to be interconnected and provisioned
with routing and translation logic about each of the other 32 PBX systems in the network.
The solution looked more like a network of 33 disparate enterprises, than one large
integrated solution.
Clearspan is Ideal for Multi-Site Enterprises
Clearspan was designed to address all of these multi-site issues not only for classic PBX
services, but also for the complete suite of voice applications including conferencing,
unified messaging, IVR and contact center services. This makes Clearspan an ideal
choice for large multi-site enterprises.
Clearspan can be characterized as having a logically centralized call control model. The
diagram below depicts how the same retail enterprise described above would be served
using Clearspan:


Figure 6 Clearspan Call Control Model

With Clearspan, the number and complexity of sites may be greatly reduced. Centralized
call control enabled this enterprise to share a geographically redundant platform, reducing
maintenance costs, support costs and total cost of ownership.

Unprecedented Capacity and Scalability
In this deployment, all 1600 retail sites and hundreds of thousands of employees have
their services delivered through a single pair of Clearspan servers in geographically
redundant data centers. This is possible because, unlike traditional PBX solutions,
Clearspan has no limitations on the number of sites or the number of subscribers in each
site. The diagram below takes a closer look at a typical multi-site Clearspan deployment
model:


Figure 7 Clearspan Multi-site Deployment Model

Voice Enabled Wide Area IP Network
Traditional multi-site PBX solutions require dedicated trunks between each site. These
solutions often require a fully meshed set of interconnects in order to optimize call routing.
Because Clearspan is a voice over IP solution, it can leverage the existing core wide area
IP network that already interconnects enterprise sites for data applications, as shown
above. It is important to note that the quality of service over the IP network must be
managed, as it will be used for both call signaling and media traffic between sites.
Precedence must be given to voice traffic over data traffic entering the network.

Geographic Data Center Redundancy
Voice service is a critical business application –enterprises cannot function if the voice
network is down. If all call control is provided from a single data center on the enterprise
network, there is a risk of total network outage in a disaster scenario. To meet enterprise
disaster recovery requirements, Clearspan supports geographic redundancy. Instead of
installing all network elements in one dedicated data center, Clearspan allows the
redundant elements to be installed in a separate geographically redundant enterprise data
center. Although these network elements are installed across two physically separate
sites, they are still managed as one logical system – configuration and provisioning
changes are automatically replicated between data centers. This approach provides an
enterprise with carrier grade five 9’s reliability.


Optimized PSTN Inter-working and Expense
Interconnections to the PSTN are an expensive recurring cost to the enterprise. In a multi-
site enterprise, it is important to minimize the number of direct connections to the PSTN
and leverage the wide area network for intra-site calls whenever possible. Clearspan will
always route calls over the core WAN, utilizing VoIP whenever possible.
Clearspan also has routing policies that allow enterprises to leverage their PSTN gateway
deployments. For instance, in the diagram above, Branch Office A has a PSTN gateway
in local calling area 1 (LCA-1). This PSTN gateway is not only used for local calls by
users in Branch Office A, but also Branch Office B. Similarly, Branch Office D in local
calling area 2 (LCA-2) also serves users in Branch Office C.
To optimize hop-off costs to the PSTN, Clearspan can choose the best gateway route for
a given caller and called number. For instance, if a user in Branch Office D (LCA 2) calls a
number that is LCA 1, Clearspan will route the call over the WAN and through the PSTN
gateway that is hosted in Branch Office A, where it will be delivered as a local call. This
provides for least cost routing of public calls over the private network.
Often a large enterprise will negotiate a reduced long distance rate with an inter-exchange
carrier. The rate will apply to all calls made out of the PSTN gateway hosted in the
primary data center (headquarters) or an alternative backup data center (division office).
Clearspan can be provisioned with routing policies that will select the PSTN gateways in
the data centers for all long distance calls from any of the enterprise sites.

Site Independent Feature Functionality
Unlike some traditional PBX solutions, Clearspan services are not bound by physical site
boundaries. In the network above, users in one site can belong to service groups with
users in any of the other sites. For instance, a user in Branch Office B can belong to a
hunt group with users in Branch Office C and users in Headquarters. The same user can
act as an agent with one or more call centers in the network – regardless of where the call
center pilot number is hosted. Relocating users from one site to another is a simple web
based operation. This capability allows enterprises to reduce employee redundancy and
improve responsiveness.

Smart Sites – Remote Survivability Solutions
Because Clearspan is based on the Session Initiation Protocol (SIP) standard (a peer to
peer protocol) as opposed to a master slave protocol like MGCP or SCCP, sites can be
deployed with remote survivability support. These “smart sites” are able to detect a WAN
outage. When an outage occurs, the local access equipment provides a set of basic
services including: intra-site extension dialing, call hold/retrieve, call transfer, and 3-way
calling. If a PSTN gateway is available on the site, then incoming and outgoing PSTN
calls can also be delivered.

Remote Worker Support
Clearspan allows remote workers or telecommuters to have access to their complete set
of voice services while working from home or on the road. Clearspan supports both
“broadband” and “narrowband” remote worker solutions.
With a broadband connection, users get service directly through their IP phone or soft
phone connected to the corporate network over a VPN connection. With a narrowband
connection, users get service through their local PSTN connection. Incoming calls are
forwarded to their “remote-office” number, while outgoing calls are initiated through a web


based call back mechanism. Both solutions allow enterprises to manage outgoing calling
plans and centralize billing for all remote users.

Ease of Management
One of the biggest advantages of Clearspan over some traditional PBX solutions is
manageability. While other PBX solutions often require routing policies and translations to
be replicated across multiple systems, routing policies and translations in a Clearspan
network only need to be provisioned once.
Clearspan also incorporates a multi-level web portal for user and service management.
This portal allows enterprises to administer company wide group services, define sites,
and manage resources such as devices, phone numbers, and services across the
corporation. The department level portal allows delegation of user moves/adds/changes
(MACs) to a designated department administrator. Enterprise administrators have the
flexibility to manage users across all sites remotely, or delegate user-management
responsibility to individual site administrators.
Finally, the user portal empowers end-users to take control of their service profile through
the web. Users can select their incoming and outgoing calling preferences directly on the
web. From classic services like call forwarding, call rejection, and 3-way calling, to
enhanced services like remote-office, voice mail and business conferencing, the web
portal makes it easy for end users to access their services anywhere on the corporate
network.

4 Key User Applications

All Clearspan features run on the same platform, use a common interface, and can be
added to the network at anytime. The sections below briefly describe a few of the key
applications provided by Clearspan. For a complete description of all of the features and
their functionality, refer to the latest “Clearspan Product Overview” for details. All of these
features are software based solutions and require no extra hardware to deploy.

4.1 Clearspan Communicator Family
The Clearspan Communicator family enables end users to convert their desktop computer
into a fully functional phone. This client offers advantages over other client softphones as
a result of the integration with the Clearspan system platform and tailored features-
functionality to Clearspan applications.

4.1.1 Clearspan Communicator
The Communicator client is an audio-only client that provides a feature-rich desktop
interface to make and receive calls as well as execute enhanced features. The
Communicator has access to Clearspan’ many advanced VoIP features including
screening and messaging/voice mail integration.
As an integrated device with the Clearspan platform, enterprise administrator are able to
auto provision the client, provide automatic updates, and maintain version control
seamlessly through the Application Server. The Communicator is capable of supporting
G.729a protocol. The Communicator allows users to utilize the client as a primary or
secondary phone device. Clearspan supports this via server features that enable both
devices to register to the same user account and directory number (DN)/direct-inward
dialing (DID) number.


The Communicator is available in English, Spanish, and Simplified Chinese, in addition to
having multi-language support through the Clearspan platform.

4.1.2 Clearspan Communicator - Multimedia
The Clearspan Communicator - Multimedia provides all the capabilities of the audio-only
Communicator and adds high quality multimedia transmission. With the Communicator -
Multimedia, a user’s desktop computer can function as a full-feature multimedia
communications device.

Figure 8 Clearspan Communicator Multimedia Menu

While Clearspan integration offers considerable value for the audio-only client, the benefits
are even further expanded with the addition of video. First, the video codec negotiation is
somewhat more complicated than audio-only codec negotiation. Clearspan pre-
integration limits the issues associated with video codec negotiation, simplifying
deployment and reducing support calls and customer issues. Second,
pre-integration with the video client enables separate routing of voice streams over
traditional handsets while the video signal is connected to the video client. Thus, users
can enjoy features of their desk handsets such as a high-fidelity speakers and
microphone, along with the multimedia experience available from the desktop client.


4.2 Clearspan Assistant Family
The Assistant family is a suite of products that helps, or “assists”, end users to manage,
control, and configure their phone. In a compact and extremely useful user interface,
Clearspan Assistant provides users with an easy way to answer, transfer, and dial calls,
search enterprise, personal, or Outlook contacts directories, and quickly enable/disable
call settings such as Call Forwarding, Do Not Disturb, and Simultaneous Ringing with a
click of a mouse.

4.2.1 BroadWorks Assistant – Enterprise
The BroadWorks Assistant is a feature-rich, integrated toolbar that enables users to make
and accept telephone calls, and change telephone settings, from within Internet Explorer
and Outlook. The toolbar delivers the following features and benefits to users:
Complete call control capabilities, including making, receiving, transferring and holding
calls
Right-click to dial on highlighted phone numbers within web pages
Easily change telephony service settings such as Simultaneous Ringing, Call
Forwarding All, Call Forwarding No Answer, Call Forwarding Busy, Do Not Disturb,
Remote Office, VM User, Logged-in Status
Add Microsoft Outlook contact vCards for new incoming calls
Outbound click-to-dial from Outlook contacts
Group, enterprise and LDAP directory browsing
Call notifications with caller ID from Clearspan
See who is calling based on a contact list in various Clearspan directories
View call history
User help functions available during any operation with an HTML user guide built into
the application


Figure 9 Clearspan Call Center - Agent

Figure 10 Clearspan Call Center - Agent Menu


4.2.2 Clearspan Assistant - Mobile
Clearspan Assistant - Mobile is a feature-rich, easy-to-use mobile application that is fully
integrated with the Clearspan platform. It is an integrated Nokia E-series mobile handset
application that enables users to originate Clearspan-routed telephone calls, modify their
telephone settings, view their corporate directory, and have access to their call logs.
Clearspan Assistant - Mobile is a carrier class, lightweight mobile application for everyday
users that delivers the following features and benefits to users:
Provides access to a corporate directory
Addresses the needs of corporate, small business and small office/home office users
Receives voice mail notifications
Easily change telephony service settings such as Simultaneous Ringing, Call
Forwarding All, Call Forwarding No Answer, Call Forwarding Busy, Do Not Disturb,
and Remote Office
View call history
Place calls from a mobile phone but have it appear like as if the user is calling from
their office line
Manage only one phone number and voice mail account
Create international profiles


4.3 Personal Web Portal
The Personal web portal provides individual users with the ability to configure and manage
a host of traditional and advanced telephony services. Each user is empowered with the
control and flexibility to easily configure their services to meet their unique needs. Users
no longer have to remember any star codes or complex procedures to configure their
services, as is often the case with legacy systems. Rather, Clearspan improves personal
productivity by leveraging the web to make services understandable and actually useable.
Users can customize their services to follow them anywhere, whether at work, at home or
on the road. For example, the Call Notify service (shown below) enables users to indicate
which incoming calls they want to be notified of, and during which hours of the week.
They can also choose to have their notifications sent to their mobile phone or e-mail
address.

Figure 11 Clearspan CommPilot Personal Portal
After logging in to the personal web portal with his/her user identity and password, a user
can activate, deactivate, and modify the parameters of his/her own services, including:
Call Forwarding (Always, Busy, No Answer, Selective), Simultaneous Ring, Do Not
Disturb, Selective Call Acceptance and Rejection, Anonymous Call Rejection, Priority and
Distinctive Ringing, Calling Line ID Blocking, Voice Messaging Notification, Voice
Messaging to E-mail, Personalized Name Recording, and Remote Office. The left
navigation area lists services and features available to the user.
The Web Portal updates features and configurations in real-time, providing utility and
convenience. Convenience is important for features that are frequently updated by users
such as the call-forwarding suite. The web interface also provides a means for making
sophisticated entries and configurations, very difficult to replicate through a standard voice
portal. For example, in addition to specifying phone numbers for service treatments,
Clearspan enables users to provide SIP URLs, feature access codes, and/or speed
codes. Selective call treatments can be defined for parameters that include line ID status
such as PRIVATE, and UNAVAILABLE, along with the more common IDs such as
ANONYMOUS.
The following features can be provided with the CommPilot Call Manager:
Dial/Redial (click-to-dial) – Enables a user to enter and dial a number, dial directly
from a drop-down Phone List (Group, Personal, or Call Log) or from Outlook or LDAP
tabs, or click the Redial button.


Transfer – Enables a user to redirect a ringing, active, or held call to another number
or directly to voice mail. Before transferring the caller, the user can choose to consult
with the third party first or establish a three-way consultation.
Send to VM – Enables a user to transfer calls to voice mail.
Talk – Enables a user who is already engaged in call to answer another waiting call.
When available, Calling Line ID is displayed with caller’s name and number.
Hold – Enables a user to place an existing call on hold for an extended period of time,
and then retrieve the call to resume conversation. While the calling party is held, the
user can choose to make a consultation call to another party.
Conference – Enables a user to establish a three-way call involving two other parties.
Hang Up – Enables a user to disconnect a call that has been answered.
Hot Links – Buttons are provided to enable a user to turn on/off frequently-used
services such as Call Forwarding Always and Do Not Disturb. Alternatively, if
CommPilot Express has been configured, the user can change their CommPilot
Express status (for example, Available, Busy, Unavailable) by choosing from a drop-
down list.
System Buttons – Buttons are provided to enable a user to send an e-mail for
technical support, get context-sensitive online help, and configure the CommPilot Call
Manager.

Figure 12 Clearspan Call Manager

The Call Manager is used to augment the functions of a standard phone. It provides an
alternative user interface for enhanced services, replacing flash-hook and star codes. It is
not intended to replace the phone or be a stand-alone substitute for basic call functions.
Calls can be initiated and manipulated using the phone or the web interface. The audio is
transported through the phone, not via the user’s computer.


4.4 Unified Messaging
Clearspan messaging provides all of the features of a traditional voice messaging solution,
plus:
• Message delivery to any specified email account.
• Message waiting notification delivered to the phone and to any specified mail or
Short Message Service (SMS) account (e.g. cell phone).
• Integration of messaging capabilities with Clearspan services (call back, transfer,
find me / follow me services, escape to extension, voice portal, etc.).
Incoming calls to the user are sent to voice mail upon reaching a busy or no-answer
condition. The caller is then played a greeting. There can be different greetings for busy
and no-answer conditions and all greetings can be partially or fully customized by the user:
• Default busy greeting
• Default busy greeting with name
• Custom busy greeting
• Default no-answer greeting
• Default no-answer greeting with name
• Custom no-answer greeting
The caller can then leave a message or press 0 to transfer to an attendant. The attendant
is configurable by the user and can be any valid phone number. If the caller leaves a
message, he/she has access to the following functions:
• Long message warning tone
• Set the message status to urgent and/or confidential
• Review the message and erase, record it again or deposit it
• Users can also configure their voice mail service to serve other phones, like a cell
phone. With this capability, users can forward any phone to the CommPilot Voice
Portal phone number and have the calls be sent directly to their mailbox greeting.
This powerful application allows employees to use a single Clearspan voice mail
system for their office phone and mobile phone – they do not need to manage
multiple voice mail accounts separately.
Voice messages are stored on standard email servers (POP3, IMAP4, Microsoft
Exchange Server) as WAV audio files attached to emails.
Users can retrieve their emails from their main location, from a third-party location or from
any standard email client. When retrieving emails from their location, users simply dial the
CommPilot Voice Portal phone number (or extension). The system prompts the user for
their passcode. After entering the passcode, the user is informed of the mailbox status
(how many urgent, new, expired, and saved messages) and can review the messages
through a menu. While reviewing the messages, users can play the message envelope,
jump to next or previous message, skip ahead, skip back, pause, repeat, erase, save,
reply, call back, forward, compose and send to a user or a distribution list.
When retrieving emails from an email client, the user simply configures the client to collect
email from the email server where the messages are stored. Messages are retrieved as
WAV attachments to emails and can be listened to with standard audio software.
Messages received as email can be manipulated like any other email (stored, forwarded,
replied to, etc.). Some enterprises prefer to keep “carbon copies” of all voice mail


messages received, for regulatory or security reasons – this can be done easily on
Clearspan with just a few clicks.
When the user receives new messages, they can be notified by standard message waiting
indication mechanism (stutter dial tone and message waiting lamp). Users can also
request a notification to be emailed to a specific location, like a cell phone, when a voice
message is received.

4.5 Contact Centers
Clearspan provides support for basic contact centers, allowing business agents to receive
incoming calls from a central phone number. Using this service, a business can establish
technical assistance lines, customer support numbers, or order-taking centers. Multiple
contact centers can be supported per business. Incoming calls to a contact center are
presented to the next available agent.
The Contact Center is based on Automatic Call Distribution (ACD) service to provide a
complete, business-ready application. Hence, contact centers inherit all of the
characteristics of the Hunt Group service and are also provided with sophisticated call-
handling features like queuing, music on hold, etc.

Figure 13 Clearspan Contact Center Menu

Clearspan expands the capabilities of legacy contact centers by delivering a true remote
agent solution, allowing call center agents to be geographically distributed. Therefore
agents can attend calls from home, a satellite office, or any other location served by
Clearspan in a transparent fashion


.

Figure 14 Clearspan Contact Center

Clearspan Contact Center functionality can be combined with other Clearspan Call
Completion services to ensure that all incoming calls are serviced expeditiously under any
network condition and at anytime.
Voice mail – If there are no agents to handle an incoming call or the call goes
unanswered for a specified amount of time, the call can be forwarded to a contact
center voice mailbox.
Night service – Calls received after-hours or on non-business days receive a service
menu of options allowing a caller to leave a voice message or transfer to an
emergency number.
Multiple call distribution policies – Incoming calls are handled according to the
selected policy, which include uniform call distribution, linear hunt group, circular hunt
group, and simultaneous ringing.
Call queuing – When all call center agents are busy, incoming calls can be queued
until they can be presented to an available agent.
Queue escape – Callers who are queued can press a key to be sent directly to the
contact center voice mailbox instead of waiting for an available agent.
Overflow – When a contact center cannot accept any more calls, incoming calls can
be forwarded to an overflow phone number.
Statistics – Statistics are generated for each contact center group and can be viewed
by the group administrator via the web portal.
Service integration – Any Clearspan personal service can be assigned to a contact
center phone number to customize the contact center group. This includes services
such as Call Forwarding, Call Notification, Call Screening, and Voice Messaging.
Queue flushing – When all agents in the contact center group log out, queued calls
are automatically sent to the contact center group voice mailbox.


Outlook contact integration – vCards from the agent’s Outlook or Exchange contact
database pop up for incoming calls.
Agent log in/log off – Agents can log in and out from the group so that calls are only
presented to agents who are on duty.
Screen pop-ups – Incoming calls pop up on a web screen showing information
associated with the incoming call. A group-specific URL is accessed for each call.
Configurable Music on Hold – The queued callers are provided with music or
advertisements between periodic announcements.

4.6 Auto Attendants
The Auto Attendant provides enterprises with a powerful and flexible tool to field inbound
calls and deliver them to the intended destination through interactions with the caller. The
Clearspan Auto Attendant is an integral part of the product offering and does not require
any third-party equipment.
The Auto Attendant is reached by dialing an associated phone number or an extension.
Once connected to the Auto Attendant, the caller is played a greeting that provides a
menu of options to complete call routing.
The menu, which is configurable by an enterprise administrator, can provide up to nine
options to the caller, including:
One-key dialing – The caller presses a pre-defined DTMF key to reach a particular
phone number or extension within the group. This option is also used to build
multilevel IVR menus.
Operator dialing – The caller presses a pre-defined DTMF key to reach an operator.
Name dialing – The caller spells the name of the intended party using the numerical
DTMF keypad. Upon identifying a unique match, the caller is played the name of the
called party and is then transferred.
Extension dialing – The caller enters the extension of the intended party through the
numerical DTMF keypad. Upon collecting the full extension, the caller is played the
name of the called party and is then transferred.
The moves, adds, and changes for users in a group are automatically available for the
Auto Attendant name dialing and extension dialing functions. Access to users currently in
the group is always available.
The Auto Attendant uses the multi-location enterprise capabilities of the Clearspan
platform to transparently support geographically distributed groups. The Auto Attendant
supports users with a Direct Inward Dialing (DID) number as well as users without an
external public directory number. These users originate calls as usual and the Auto
Attendant allows them to receive external calls. Calls made to the Auto Attendant use the
routing capabilities described above to terminate calls to the appropriate user.
This support provides greater flexibility for a group administrator to create and delete users
and in many cases reduces the costs associated with obtaining DID numbers. Clearspan
can also offer speech-based Auto Attendants, which use a speech IVR instead of a
DTMF-based IVR (“call Joe on his mobile”).

4.6.1 Clearspan Receptionist
The Clearspan Receptionist is a carrier class IP telephony attendant console for use by
receptionists, or telephone attendants, who manage and screen inbound calls for
enterprises. The BroadWorks Receptionist is a feature-rich desktop application that is fully


integrated with the Clearspan platform. The application delivers the following benefits to
users:
Easy-to-use design that follows the natural work “flow” of a call from the top to the
bottom of the screen.
Intuitive business processes, as only “valid” options are presented to the attendant.
Professional call handling as critical information is available in real time.
Accurate delivery of messages via a one-step process when people are unavailable.

The BroadWorks Receptionist can be used under a number of different operational
scenarios. These scenarios include:
After Hours – Allows operators to automate switching from day to night mode.
Call Center Queue – Allows operators to monitor and control calls in a Call Center
queue, and to manage their availability status.
Hoteling – Allows multiple part-time operators to share a single log-in sequence when
they change shifts.
Low Traffic – Single receptionist answering one or more dedicated main line
numbers.
High Traffic – More than one attendant console managing multiple dedicated main
line numbers.
Network Attendant Console – Geographically dispersed operators supporting each
other in an enterprise configuration.
Multi-tenanted Offices – One or more operator answering calls on behalf of different
organizations.
Optional Voice Mail Transfer – Operator has the added ability to transfer calls to
voice mail for contacts in a group/enterprise that are busy or unavailable.


Figure 15 Clearspan Receptionist Menu
4.7 Web Conferencing
Clearspan supports a variety of business conferencing services, including meet-me
conferencing, reservationless conferencing, automated dial-out conferencing,
collaboration tools, and more.
Web Conferencing enables the set up, use, and monitoring of n-way conferences via a
web interface. Both internal and external participants can use a conference bridge once it
has been set up. A single
The Conferencing service includes the following features:
Audio and web conferencing
Scheduled, recurring, reservation-less, and ad-hoc
Meet-me dial-in numbers
Web collaboration
Share Microsoft PowerPoint, Excel, and Word files (extensions are automatically
checked by Clearspan for validity)
Secure SSL and password protection
Web browser viewable, no client required
Optional click-to-download of web presentations
Moderator control
Dial-out capability


Mute, hold, drop, and add participants
DTMF and web portal interfaces
Roll call (service can record and announce participants’ names as they join)
In-call functions
Hand raising (muted participants can request the conference leader to un-mute
them temporarily)
Optional leader
PIM integration
Automated e-mail invitations and Outlook calendar entries
Reporting
Web-based reporting
Department and project codes
Recording
Recording and playback of individual conferences (the CDR can account for
recording duration)
Access code generation
Automatic, pre-assigned, or user-defined

Figure 16 Clearspan Conference

Figure 17 Clearspan Conference Call Menu


4.8 Clearspan Deployment Studio
The Deployment Studio facilitates the customization, deployment, and management of
Clearspan Assistant, Assistant - Enterprise, Receptionist, and Call Center -
Agent/Supervisor desktop client software by enterprise providers. Providers can configure
a variety of client settings, such as branding, networking, localization, and additional
pre-defined settings, for Assistant, Assistant - Enterprise, Receptionist, and Call Center.
The Deployment Studio is required to deploy all Clearspan clients.
Enterprise providers proceed through an intuitive GUI to fully customize their client
applications to suit their specific requirements.

5 Integrating with Legacy Equipment

This section describes solutions for integrating Clearspan with existing enterprise voice
networks. This is often required in multi-site enterprises with a large investment in legacy
voice equipment. The services provided by Clearspan complement the services provided
by the existing solution. Equipment on the enterprise site communicates with Clearspan in
the enterprise data center. Solutions are described for enterprises with existing Private
Branch Exchanges (PBXs).

5.1 PBX Solution (SIP Trunking)
A Private Branch Exchange (PBX) is a private telephone system that provides on-
premises dial service and connections to the PSTN. A PBX offers call control features
such as call waiting, call transfer and conferencing. Most PBXs can interface with other
communications applications such as auto attendant, voicemail, automatic call distribution
and call accounting. Many PBXs use proprietary phones, since a significant portion of a
PBX sale involves handsets.
PBXs typically serve medium to large enterprises locations. Typically, each user has a
dedicated line. For outgoing calls, the PBX switches calls from a user line to a trunk
connected to the PSTN. For incoming calls, a PBX switches calls from a trunk connected
to the PSTN to a user line. PBXs provide both personal and group based services. They
are popular in enterprises where employees require direct dial access.

5.1.1 Network Layout
A PBX gateway is used to inter-work a legacy PBX with Clearspan. The gateway provides
trunk interfaces for connectivity with the PBX and Ethernet interfaces for connectivity with
the LAN. The PSTN interfaces are via CAS or PRI trunks. On the Clearspan system, the
address of the gateway connected to the PBX is setup as a routing element within a Voice
VPN enterprise group.


Figure 18 Clearspan Network Layout

As shown, the PBX connects to the GW via CAS or PRI trunks. The gateway is connected
to the corporate LAN via one or more Ethernet interfaces.

5.1.2 Services
To illustrate how the Clearspan services can be used to complement the services
provided by a legacy PBX, an example of a typical enterprise is used. In this example, an
enterprise has a PBX serving telephones and/or fax machines with a T1 PRI trunk
connected to a GW.
In this example, calls can be delivered to the GW with or without direct inward dialing
(DID). If DID is used, then individual phone numbers are assigned to each user. This
allows services to be provided on a user basis. If DID is not used, then there is a main
phone number for the PBX. Clearspan delivers all incoming calls to the main number,
which is associated with an attendant or auto attendant. The attendant then delivers the
call to individual lines connected to the PBX. Note, most PBXs typically support both DID
and non-DID access simultaneously. Some users will use DID and some will not.

5.1.2.1 Services for PBXs with DID
Some of the basic personal services provided by Clearspan are also provided by the PBX,
such as call transfer, call forwarding, extension dialing, etc. Clearspan complements these
basic services, providing additional enhanced personal services, including:

Service Usage
Anonymous Call Rejection Deny anonymous incoming calls before reaching the PBX.


Service Usage
Call Forwarding Selective Forward certain incoming calls before reaching the PBX.
Call Notify Log all incoming calls before reaching the PBX.
CommPilot Call Manager Provide for web based call control functions.
CommPilot Express Customized incoming call management.
LDAP Directory Integration Integrate with enterprise directory.
Outlook Integration Integrate with Outlook or Exchange Server.
Remote Office Provide support for tele-workers and remote offices.
Selective Call Acceptance Accept only certain incoming calls before reaching the PBX.
Selective Call Rejection Deny only certain incoming calls before reaching the PBX.
Simultaneous Ring Ring the PBX user in addition to multiple external numbers.
Voice Messaging Hosted, centralized voice messaging.

Group services can also be used with PBXs with DID support. Group services that are
typically assigned include:
Service Usage
Account and Authorization Track and/or authorize all outgoing calls made from the PBX.
Codes
Attendant Console Provide hook status for off network calls to and from the PBX.
Auto Attendant Provide automated attendants for PBX users.
Call Centers Provide call centers for PBX users.
Calling Group Identity Provide the enterprise identity, instead of the individual line identity, for all
Delivery outgoing calls made from the PBX.
Hunt Groups Provide hunt groups for PBX users.
Incoming Calling Plan Block certain types of incoming calls made to the PBX.
Outgoing Calling Plan Block certain types of outgoing calls made from the PBX.
Voice Portal Provide a dial in interface for service interaction and management.

5.1.3 Services for PBXs without DID
When DID is not available, services are provided on an enterprise wide basis (main phone
number), not on a per user basis. Clearspan provides hosted, enhanced services that
complement those provided by the PBX, including:
Service Usage
Call Forward No Answer Forward unanswered calls to an auto attendant or enterprise voice
mailbox.
Call Forwarding Selective Calls received after hours or on the weekend are forwarded to an auto
attendant, enterprise voice mailbox or remote location.
Call Notify Calls to the main number are logged via email notifications.
Calling Line Identity Delivery Provide caller number and name for incoming calls.
Selective Call Rejection Nuisance calls to the main number are blocked.
Voice Messaging Provide an enterprise voice mailbox or individual voice mailboxes. This
works in a similar manner to a KTS. See KTS Solution for more details.

Group services can also be used with PBXs without DID support. Group services that are
typically assigned include:


Service Usage
Account and Authorization Track and/or authorize all outgoing calls made from the PBX.
Codes
Calling Group Identity Provide the enterprise identity, instead of the individual line identity, for all
Delivery outgoing calls made from the PBX.
Outgoing Calling Plan Prevent certain types of outgoing calls made from the PBX.


Part II – Solution Components


6 Clearspan Service Delivery Platform

This section describes the Clearspan platform in more detail. It provides specifics on each
Clearspan server as well as the preferred hardware platforms. In addition, it provides call
flows showing how the components interact.

6.1 Network Elements
Clearspan is composed of the following servers:
Application Server – provides call processing, service logic, endpoint management
and service management
Network Server – provides policy based translations and routing, subscriber location
directory and media server selection
Media Server – provides specialized media resources, including media recording,
media playback, DTMF digit detection, three way conferencing and media relaying
Web Server –supports Clearspan system web interfaces and additional external
interfaces. The Web Server supports the Personal Web Portal and Call Manager via
HTTP, as well as the BroadWorks Open Client Interface for the Assistant and
Receptionist applications.
Conference Server (optional) – provides enhanced audio and web conferencing.
PM Server – monitors the health, performance and reliability of the Clearspan
system.
Client Management System (CMS) – provides the central point to control the
functions associated with the configuration, deployment, and activation of user profiles
and templates for endpoints or clients.
Together, these components enable a wide array of personal, group, and network
services including PBX equivalent services, messaging, attendants, interactive voice
response, call distribution, site networking, conferencing and desktop integration. Each
server is described in more detail below.

6.1.1 Application Server

The Application Server is a service delivery platform responsible for the execution and
management of enhanced personal and group services. Application Server functions
include management of network traffic, handling of signaling interfaces, and logical
execution and management of services. The Application Server comprises a database,
the ServiceOS™ abstraction layer, and protocol stacks.
The Application Server supports the Session Initiation Protocol (SIP) as the
communication protocol with VoIP equipment, including:
Access Equipment – SIP Phones, Analog Line Gateways, WiFi Phones, Soft Clients
Network Equipment – PSTN Gateways, PBX Gateways, Session Border
Controllers, Edge Elements
Other Servers – Network Servers, Media Servers, Conferencing Servers, Third Party
Servers
The Application Server provides secure access to subscribers and services for
management, administration, provisioning, and configuration. The associated web portals
can be customized for different customer groups, based on the subscribed-to services of


those groups. With various levels of security and accessibility (system provider, enterprise
administrator, group administrators, department and end-user), secure access is granted
not only to the system, but also to the correct level of privileges. Different users are
directed to the applicable screens with proper security and real-time information.
Information is retrieved from the database and displayed in real time.

Figure 19 Application Servers

Application Servers are deployed in pairs, where one servers acts as the primary server
and the other as the secondary server. Under normal operation, the primary Application
Server will manage all call processing and endpoint management. User modifications,
additions and deletions are replicated across the database of both the primary and
secondary Application Server, making all service and user profiles available on both
servers. Database replication is done in real-time, as additions and modifications are
made. Thus, both the primary and secondary servers have access to exactly the same
database.
In the event that the primary server fails, or is simply inaccessible from one or more
endpoints in the network, then those endpoints will be able to route their calls through the
secondary server. The failover time is the time it takes an endpoint to retry after a non-
response to the secondary server. This is typically engineered to be less than 1 second.
This type of redundancy may be geographically distributed; protecting against server
failures as well as against IP networking failures (e.g. router and circuit failures).
Application Servers are hosted on IBM servers, running the Linux operating system. Each
server connects to the VoIP network using 10/100BaseT Ethernet interfaces. Multiple
interfaces are available supporting a wide variety of IP network configurations. No
specialized hardware is required.

6.1.2 Network Server
The Network Server enables system providers to centrally manage network-related
applications within their network. This includes public translations and routing capabilities
such as least-cost routing, as well as enterprise-focused network applications such as
voice virtual private networks (VPNs).


The Network Server allows construction of massive next-generation voice networks by
assisting with the scaling of IP telephony networks and offloading subscriber-specific
routing functions, allowing softswitches to focus on core call state management for
trunking media gateways. The Network Server also acts as a platform for network-based
enterprise applications such as private dial plans, and supports passing of non-numerical
characters like * and # to the network so they can be used to trigger functionality in other
networks.
The Network Server optimizes network resource utilization by providing the capability to
selectively route calls to geographically dispersed resources, thus maximizing network
bandwidth utilization. For example, the Network Server can manage Media Servers as a
single network-wide pool of resources and, at the same time, select an appropriate Media
Server for the location of the requesting subscriber. The Network Server also performs a
central role in supporting Clearspan redundancy capabilities.
Provisioning new subscribers can occur via synchronization of the group and user data
between Application Servers and Network Servers. The Network Server functions with
the Application Server as shown in the following figure:

Network Devices

Subscriber S
Location I
P

S
I Translations
P & Routing

Figure 20 Network Server

The policy database provides public and private (that is, enterprise) policies, to manage a
system provider’s network services. It also maintains subscriber location information to
track where subscribers are hosted within the network. The policy database contains
translation processing and a routing engine that is driven by a flexible policy approach.
Dialing plans, call typing, route selection, and network services configuration are policy-
driven and can be updated “on-the-fly”, including the introduction of new policies within the
network. Operators can configure the precedence of their routing policies to match
requirements.
The ServiceOS manages the sessions, which are the network connections associated
with a user or network element.

6.1.3 Media Server
The Media Server functions together with an Application Server (AS) and enables
enhanced services, such as Auto Attendants, Video Auto Attendants, Meet-Me,
Conferencing, Multimedia Messaging, video advertisements in Call Center queues, play
tones and treatments to callers. The Media Server supports these services for wireline,


wireless, and/or converged networks, offering enterprises enhance productivity for end-
users. Media Servers can be geographically distributed, which can minimize call latency
and bandwidth utilization.

AS HTTP

S S RTP
I I RTCP
P P

NS

Figure 21 Media Server

In a stand-alone architecture, SIP and HTTP are used as standard protocols for
communication between the Application Server, Network Server, and the Media Servers.
The Application Server requests resources from Media Servers and controls the behavior
of those resources. Third-party media servers can also be deployed and controlled via a
standard SIP interface.
Enterprise providers can choose from multiple codecs that are supported on the Media
Servers. Configurable classes of multimedia service can also be assigned to enterprise
providers, groups, and individual users, restricting callers to the Codecs in their assigned
set. Enterprise providers can opt for lower bit-rate Codecs to increase the number of
simultaneous calls that can be provided on an access link to end users. G.729 is one of
the available Codecs supported.
In addition, the Clearspan system can be configured to route calls differently depending on
the Codecs they use, eliminating network elements that do not support included media, or
prioritizing network elements that offer better support for included media.

6.1.4 Web Server
System providers deploy Web Server farms to support Clearspan system web interfaces
and additional external interfaces. The Web Server supports the CommPilot Interface
using HTTP, and the Open Client Interface. System providers can choose to use either of
these interfaces or both concurrently. The following figure shows the functions within the
Web Server:


Web Portals,
End Users and
Applications
Administrators
and Plug Ins

O H
C T
I T
P

O L
Data C O Subscriber
Access I C Location

•

Figure 22 Web Server

6.1.4.1 Open Client Interface
The Open Client Interface (OCI) is an open interface that enables third-party applications
to leverage Clearspan call control and provisioning functionality. This interface operates
using both XML and HTTP/SOAP. Thus, Clearspan makes it easy for developers to
create new applications to augment existing Clearspan functionality. For example, the
OCI queries the Network Server to find the Application Server with the requested data
such as a user’s mobile number or department.

6.1.5 Conference Server
The Conferencing Server is a specialized media resource that provides a complete set of
enhanced business conferencing features, including web-based presentation and
collaboration. A single pool of Conference Servers can support multiple Application
Servers in a Clearspan system. The Conferencing Server functions as shown in the
following figure:

Access and Network
Devices

R
T
P

S H
Resource T Data
I T
Control P Retrieval
P

•

Figure 23 Conference Server

The Conferencing Server is fully integrated with the Clearspan back office and operates
on standard hardware, thereby allowing for cost-effective scaling with the rest of the
platform.


6.2 Call Scenarios
This section describes how the components of Clearspan work together to deliver a call
from and to a user.

6.2.1 Registration
When an access device, such as an IP phone or analog gateway, is powered on, it sends
a registration message to the Application Server. This message indicates the user and
their associated address. Clearspan stores this information in its location database. It is
used when calls are received for a user, so the call is delivered to the appropriate device.

6.2.2 Call Origination
When a user originates a call, a signaling message is delivered to the Application Server
to which the endpoint is registered. The Application Server provides call control and
service logic execution. This includes services such as call screening and extension
dialing.
If the call is to another user within the same group then a signaling message is sent to the
destination user’s endpoint, and the device is alerted. If the call is not to another user
within the group, a signaling message is sent to the Network Server. The Network Server
computes a destination route using the user’s phone number and group along with the
dialed number.
The Network Server computes a list of destination gateways, and returns this list to the
Application Server. The Application Server sends a signaling message to the first gateway
in the list. If the gateway accepts the call, then call setup is complete. If the gateway
denies the call, then the next gateway in the list is tried until successful call setup is
achieved.

6.2.3 Call Termination
When someone external to the group calls a user, the PSTN gateway delivers a signaling
message to the Network Server. The Network Server tracks which Application Server is
currently serving the user, and returns this address to the PSTN gateway. The PSTN
gateway then sends a signaling message to the hosting Application Server.
The Application Server receives the signaling message, and provides call control and
signaling logic execution. This includes services such as call screening, call forwarding,
find me/follow me and voice messaging. After the initial termination services are provided,
a signaling message is sent to the user’s endpoint and call setup is completed.


6.2.4 Entry / Exit Point
The Network Server is responsible for managing routing information associated with a
Clearspan system, and for managing location information for individual users. It acts as a
single point of access for calls coming from outside of Clearspan via a PSTN or PBX
gateway.
For example, when a call from the PSTN is destined to a user managed by Clearspan, the
call will be routed from the PSTN to a PSTN gateway. The PSTN Gateway will then send
a signaling message to the Network Server; the Network Server will then redirect the call
to the Application Server that is hosting the called user. If the Network Server is hosting
multiple Application Servers, it will redirect the call to the Application Server hosting the
user with a phone number matching the called phone number.
Since the Network Server is a centralized point of contact, it is the key to providing
scalability. Because the Network Server is a stateless, transaction-oriented server, it
scales easily by adding more Network Servers. As a system grows, a provider can add
Application Servers and Network Servers to increase system capacity. As the number of
supported users grows, the provider must also add Media and Conference Servers, since
the system will require more media resources.

6.2.5 Media Resources
When a service on the Application Server requires interaction with the user, the
Application Server requests a media resource. It does this by sending a media resource
request to the Network Server. The Network Server computes a list of available Media
Servers, which are closest to the point of entry. This is computed by comparing the calling
phone number with the phone number associated with the Media Server.
The Network Server returns a list of Media Servers to the Application Server. The
Application Server then uses this list to request a media resource. If the first Media Server
in the list is not available or does not have available resources, then the next server in the
list is tried. This is done until a media resource is successfully allocated.
When a resource is allocated, the Application Server requests that the media path be
reset so that the user or destination is directly connected to the media resource. Then, the
particular service on the Application Server orders the media resource to perform a
particular function. This can include things such as playing an announcement, recording
audio, detecting DTMF digits or conferencing multiple parties together.
When the service completes, the Application Server requests that the media path be reset
back to the original settings so that the user or destination is released from the media
resource. Then, the media resource is released and returned to the queue of available
resources.

7 Access Devices

Access devices provide terminal-access to the network by allowing an end-user to
originate and terminate calls. In this solution, all access devices use SIP (the Session
Initiation Protocol) as the common access signaling protocol. In addition to providing a call
signaling path into the network, the access device is also responsible for converting the
end-user analog voice transmission into a packet based voice over IP stream in real time.
This voice over IP stream is commonly referred to as the RTP (Real-Time Transport
Protocol) stream. In general, access devices can be further categorized into access
gateways, IP phones and soft clients. Access devices should all support the G.711
CODEC for u-Law and a-Law, as well as the RFC 2833 standard for in-band DTMF
transmission.


7.1 Access Gateways
Access gateways allow standard analog phones to interface to the VoIP network. On one
side they provide an FXS interface to a standard analog phone. On the other side they
provide a 10/100BaseT interface to the VoIP LAN. Access gateways allow existing analog
wiring to the desktop to be re-used in a VoIP network. Access gateways can be further
sub-divided into “desktop gateways” and “rack-mounted gateways”.
Desktop-gateways are usually one to four port units that are intended to be deployed near
the desktop or station (i.e. fax terminal). Rack mount gateways can support anywhere
from 8 to 48 ports and are intended to be deployed in a wiring closet within the same
enterprise site hosting the users – usually deployed along side other networking
equipment such as hubs and switches.
Access gateways will typically support a number of local or “built-in” services. These
include call-hold, retrieve, transfer, call-waiting (up to 2 call appearances), calling
name/calling line id, and sometimes even 3-way calling. These basic built-in services are
an important building block when designing remote site survivability solutions in the event
that access to both of the redundant data centers is lost.

7.2 SIP Phones
SIP Phones are digital handsets that interface directly to the VoIP network without
requiring an intervening access gateway. SIP phones have a scalability advantage over
access gateways as they require no extra rack space as new users are added to the
network. SIP phones can be further categorized as basic and enhanced.
A basic SIP phone provides a core set of services similar to those provided in an analog
gateway: hold, transfer, calling name/caller id, and usually 3-way calling. Basic SIP
phones usually offer only one line and only provide up to two simultaneous call
appearances. They should include a built-in Ethernet bridge so that users can easily
share the port on their desktop with their phone and their PC. The bridge should support
in-line power as well as VLAN and TOS bit tagging.
Enhanced SIP phones, such as Aastra’s 3- and 5-series, build on the capabilities of the
basic SIP phone, but include a comparable handset experience to that found in traditional
PBX offerings. Features like multiple call appearances and bridged line appearances are
supported. An enhanced SIP phone also includes a built-in hands-free speaker phone.
These phones typically include a large screen LCD display that allows access to
directories, contacts and other hosted content through their micro-browser interface.

7.3 Soft clients
Soft clients allow end-users to access the communications network using software on a
general purpose desktop computer, laptop or personal digital assistant. These clients
leverage the soundcards commonly found in computer platforms. Users can simply plug
in a headset and a microphone to the system and use the software to originate and
terminate calls. Features and quality tend to vary greatly from one vendor to another.
Usually a soft client is deployed as a secondary device and used only when a standard IP
or gateway-based handset is not available.

8 Network Gateways

PSTN gateways are used to inter-connect Clearspan with the PSTN. The gateway
provides trunk interfaces for connectivity with the PSTN and Ethernet interfaces for
connectivity with the LAN. The PSTN interfaces are via CAS or PRI trunks, or FXO analog
lines. On the Clearspan system, the address of the gateway connected to the PSTN is


setup as a routing element. Public routing policies are used to route calls to the
appropriate gateway.
For incoming calls from the PSTN to a Clearspan user, the following information is
delivered:
Trunk Type Calling Party Called Party
CAS Trunks Provided Provided
(Based on Gateway or Connected Switch) (User Identity = Number)
PRI Trunks Provided Provided
(Number + Optional Calling Name) (User Identity = Number)
FXO Lines Not Provided Not Provided
(Must Utilize an Attendant or Auto Attendant)

For outgoing calls from a Clearspan user to the PSTN, the following information is
delivered:
Trunk Type Calling Party Called Party
CAS Trunks Provided Provided
(Based on Gateway or Connected Switch) (Normalized Destination Number)
PRI Trunks Provided Provided
(User Identity = Number) (Normalized Destination Number)
FXO Lines Not Provided Provided
(Normalized Destination Number)

PSTN gateways can be placed at any site within an enterprise. Using the sophisticated
translations and routing policies used by the Network Server, calls from users within an
enterprise can utilize any PSTN gateway at any site within the enterprise. Calls to users
can be received from the PSTN gateway co-located with their site, or can be relayed on-
network to a remote site. A variety of routing policies are available, based on the call
origin, destination or call type, allowing a high degree of flexibility when configuring a
system.

9 Network Infrastructure Components

9.1 LAN Requirements
An obvious key element in the delivery of VoIP services over a data network is the
architecture and performance related to that data network. The architectures and
configurations available and diverse types of equipment deployed make it impractical to
outline a recommended data network configuration. In general, some basic considerations
must be given to how that IP network will perform. The enterprise must ensure that the
network is able to meet quality considerations as VoIP traffic is very sensitive to various
network characteristics. Characteristics to keep in mind are:

Delay — The amount of time it takes a packet to move from one point in the network to
another. VoIP can typically tolerate delays up to approximately 150ms before the quality of
the call becomes unsatisfactory with less than 70ms providing for excellent voice quality.

Jitter — The variation in delay over time. If the delay varies too widely (i.e., jitter is large) in
a VoIP call, the call quality is degraded. The amount of jitter that can be tolerated in the
network depends on the depth of the jitter buffer on the network elements in the voice
path. The greater the jitter buffer, the more the network, in general, can reduce the effects
of jitter.


Packet loss — The loss of packets along the voice path can severely degrade the quality
of VoIP calls. VoIP networks should achieve packet loss of less than 1% for satisfactory
voice quality, closer to zero provides for excellent voice quality. The enterprise must
understand the services being offered that could affect overall network design (i.e.
converged voice and data) while still allowing these metrics to be achieved. Mechanisms
such as QoS—which is dependant on what is supported by the end points (i.e. IADs
and/or Network Gateways)—will need to be employed. In addition, voice CODECs that
are enabled in the network will play a part in network design as they impact the amount of
bandwidth required, and some CODECs are more susceptible to network problems.

10/100 BaseT Switched Ethernet Infrastructure -- All VoIP endpoints should be
connected to the LAN through layer 2 IP switches. This puts each endpoint onto a
separate segment allowing maximum bandwidth at the edge of the network. The “hub and
spoke” topology of a switched layer 2 network also provides excellent security against
“snooping” attacks within the enterprise.

Virtual LANs -- Ideally, VoIP endpoints should be deployed on separate physical LANs
from data endpoints. This gives voice traffic and data traffic their own layer 2 broadcast
and collision zones. This eliminates the possibility of large bursty data packets (like a file
transfer application) from colliding with streaming voice packets on the local network
segment. However, in some enterprises this may not be practical, as it would require
rewiring every station with two Ethernet ports – one for voice and one for data.
Fortunately, virtual LAN technology has been widely adopted by all mainstream layer 2
switching manufacturers. The accepted standard in the industry is IEEE 802.1q –
commonly called “VLAN”. VLANs allow network administrators to easily group endpoints
into logical LANs, regardless of their physical location. This also simplifies moving
endpoints throughout the network. Note that when employing VLANs, it is important to
make sure all network elements conform to the same VLAN standard.

Traffic Classification -- Eventually all packets in a converged network must converge.
Typically, this happens at a layer 3 switch or router. If all traffic passing through the router
were treated equal, then voice over IP service would degrade during peak office hours as
both data and voice contended for the same routing resources. Congestion at the router
would cause both voice and data packets to be equally lost and delayed. Lost data
packets can be easily retransmitted without the user noticing, but too many lost voice
packets will become very noticeable to the user. Variably delayed voice packets will result
in an annoying jitter in the speech. To avoid degradation of voice service, a router must be
able to route voice packets with a different priority than it routes data packets. To do this, a
number of traffic classification technologies have evolved. The predominant ones to
consider are:

Class of Service -- Also known as IEEE 802.1p, or CoS. This is a layer 2 traffic
classification technology that extends and is dependant on the VLAN IEEE specification
802.1q. Using 802.1p to classify packets on VLANs allows network administrators to
prioritize traffic between layer 2 switching elements.

IP Precedence -- This is a layer 3 traffic classification technology that the Type of Service
byte in the IP header to classify traffic. Layer 3 switches or routers can then use this
classification to apply different routing priorities and queuing policies to traffic.

DSCP -- Differentiated Service Code Points, or DiffServ is a layer 3 traffic classification
technology. It uses the Type of Service byte in the IP header. Packets in a DiffServ
network are classified on a per hop basis – as opposed to IP precedence which is
intended to specify a network wide classification. Network wide classification is not
practical in large networks managed by different groups or organization – which makes
DiffServ a more practical choice for wide area network traffic classification. Regardless of


the traffic classification technology, it is important that packets are classified as close to the
edge as possible. This means that traffic classification should ideally be performed at the
VoIP endpoints.

Traffic Prioritization -- Once packets have been classified, it is important that the network
infrastructure transport the packets with the appropriate priority. This is achieved in routers
and switches by defining different queues for passing traffic. Each queue is given a
different routing priority and mapped to a corresponding class of traffic. Then a queuing
algorithm determines how each packet is de-queued for routing. There are many queuing
algorithms to choose from, but a common queuing algorithm for converged voice over IP
networks is called “Low Latency Queuing”. This algorithm defines a strict priority queue for
low latency traffic like voice. Voice traffic gets priority and all other traffic is distributed
using a class-based weight fair queuing (CBWFQ) model.

Inline Power -- Inline power delivers DC power to endpoints over the same physical CAT-
5 wiring used for the layer 2 network. Not only does this simplify deployment of IP phones,
it also allows network administrators to use centralized battery backup solutions to keep
the voice network operational in the event of a power loss. It is important to choose a
matching inline power technology. Inline power, often call Power over Ethernet (POE), is
standardized through the IETF.

9.2 Using DHCP
DHCP is used to dynamically allocate and assign IP addresses. DHCP allows you to
move network devices from one subnet to another without administrative attention. If using
DHCP, you can connect Aastra SIP IP phones to the network and become operational
without having to manually assign an IP address and additional network parameters.

Aastra SIP IP phones comply with the DHCP specifications documented in RFC 2131. By
default, Aastra SIP IP phones are DHCP-enabled.

Mandatory DHCP settings are:
• IP Address
• Subnet Mask
• Default Route
• DNS Server
• Domain

Optional settings are (one of the following)
• DHCP option 150 (TFTP server IP address)
• Standard DHCP option 66 (TFTP server name)

TFTP servers are used to provide specific and general configuration information for all IP
phones on the network. Specifying TFTP server address via DHCP eliminates any
manual phone based configuration. For more information, see the Using TFTP section.

9.3 Using DNS
Clearspan, and Clearspan hosted endpoints, use DNS for the location of service
resources. Service resources consist of:
• Proxy/Registrar Location
• Network Routing Policy Location
• Gateway Location


All network devices use DNS SRV (RFC 2782) to locate service nodes on the network.
DNS SRV specifies resource location using predicable cost and weights for each service
request. Request resolutions are based on protocol, and return weighted device locations
that are cached by each device. Examples of weighted service locations include:
• Active/Hot Standby Application Servers
• Load Balanced Network Servers
• Load Balanced Gateway Interfaces
Implementing DNS SRV for voice services enables administrators to modify network
addresses for entire networks, from a single location.

9.4 Using TFTP
There are two configuration files that you can use to define a phone’s SIP parameters; the
default configuration file and the phone-specific configuration file. These configuration files
must be stored in the root directory of your TFTP server.
You can use the default configuration file to define values for SIP parameters that are
common to all phones. You can use the phone-specific configuration file to define
parameters that are specific to a phone. Phone-specific parameters should only be
defined via a phone-specific configuration file or manually configured. Phone-specific
parameters should not be defined in the default configuration file.
Configuration File Guidelines
When modifying the default configuration file and creating the phone-specific configuration
files, adhere to the following guidelines and requirements:
• Except for parameters used to define the lines and users on a phone, all other SIP
parameters can be defined in either the default configuration file or the phone-specific
configuration file. However, for network control and maintenance purposes, we
recommend that you define the parameters that you want to apply to all phones in the
default configuration file.
• SIP parameters specified in the default configuration file will override those
parameters stored in Flash memory. Parameters specified in a phone-specific
configuration file will override those stored in Flash memory or specified in the default
configuration file.
• The name of each phones' phone-specific configuration file is unique and is based on
the MAC address of the phone.

10 Operations, Administration, Maintenance and Provisioning

10.1 Configuration and Provisioning
Clearspan is managed through an intuitive web portal. It provides separate web portals for
end-users, department administrators, site administrators, enterprise administrators, and
system administrators. Each level of the management hierarchy can configure and
manage their respective resources in real-time. All service activation and configuration
changes are implemented immediately. Users and administrators are empowered with
more control and responsiveness. The functions of each level are described below:
System Provider (System Administrator)


Monitors and maintains Clearspan
Access to all levels of administration
Sets and manages system parameters
Adds, modifies, or deletes enterprises, groups, and users
Enterprise Administrator
Adds, modifies, or deletes groups and users
Adds groups and group administrators
Provisions phone numbers and access resources
Provisions services/Customizes web branding
Group Administrator (Site Administrator)
Adds, modifies, or deletes users
Assigns phone numbers and access resources
Sets and manages services for group
Department Administrator
Adds, modifies, or deletes users
Sets and manages services for department
End User
Customizes individual phone services and features
Makes calls using the CommPilot Call Manager
In addition to the web portal, the system administrator also has access to the command
line interface (CLI). The CLI allows access to system configuration functions, as well as
monitoring and management functions.

10.1.1 Initial System Configuration
Upon initial installation, the following functions can be performed on the Application Server
using the command line interface:
Application Server Aliases - Set all the names that identify the Application Server as
a proxy or call agent, including the FQDN for the redundant pair
Network Server Identity - Set the Network Server address
Enable Media Server Selection – Enable to perform Media Server selection using
the policies on the Network Server
Enable AS/NS Sync API – Enable to automatically propagate group and user phone
number additions and deletions to Network Server
Enable Accounting – Enable to generate call detail records for each call, logging the
records into a flat file in XML or CSV format with automatic FTP transfer to an external
server
Configure SNMP – Add destination addresses for traps to access control list
Intra-Group/PSTN Routing – Set to route all inter-group calls to the Network Server
Voice Messaging SMTP/POP3 Setup – Set SMTP addresses for email dispatch,
and POP3 address for email retrieval


System Domain – Set to the default system domain can be configured, which allows
users belonging to the default domain to login with username only
SIP Realm – Set to use the domain as the realm for SIP authentication challenges
The following functions can be performed on the Application Server using the system
provider web portal:
Administrators – Add, modify or remove system administrators or provisioning
administrators (only required if there is a need for multiple system administrators)
Announcements and Tones – Select and upload the messages to be played for
system announcements and the tones (only required if the defaults are not sufficient)
Password Rules – Define the password rules that users and administrators must
follow to create or update passwords (only required if the defaults are not sufficient)
Web Branding – Modify the look and feel of the web portal for all administrators and
users (only required if the default is not sufficient)
Services – Set the system parameters for services (only required if the defaults are
not sufficient)
Group
Account/Authorization Codes - Select the messages to be played for
Account/Authorization Code prompts and failures
Call Park - Select the messages to be played for Call Park prompts and
failures
Customer Originated Trace - Select the messages to be played for
Customer Originated Trace prompts and failures
Emergency Zones - Select the messages to be played for Emergency
Zones call failures
Incoming Call Plan - Select the messages to be played for Incoming Call
Plan prompts and failures
Instant Conferencing - Select the messages to be played for Instant
Conferencing prompts and failures
LDAP Directory - Configure a system-wide LDAP Directory
Outgoing Call Plan - Select the messages to be played for Outgoing Call
Plan prompts and failures
Voice Messaging - Configure the system-wide Voice Messaging settings
User
Anonymous Rejection - Select the messages to be played for Anonymous
Rejection prompts and failures
Call Forwarding Always - Select the messages to be played for Call
Forwarding Always prompts and failures
Call Forwarding Busy - Select the messages to be played for Call
Forwarding Busy prompts and failures
Call Forwarding No Answer - Select the messages to be played for Call
Forwarding No Answer prompts and failures
Call Notify - Select the messages to be played for Call Notify prompts and
failures


Call Return - Select the messages to be played for Call Return prompts and
failures
Do Not Disturb - Select the messages to be played for Do Not Disturb
prompts and failures
Speed Call 8 - Select the messages to be played for Speed Call 8 prompts
and failures
Speed Call 100 - Select the messages to be played for Speed Call 100
prompts and failures
Upon initial installation, the following functions can be performed on the Network Server
Network Server Identity - Set the Network Server address and aliases
NNACL and LCA Files – Install Telcorida NNACL and LCA files, used for public
routing
Public Policy Defaults – Verify and modify policies for call typing, call screening,
subscriber location, near end routing, far end routing and service center routing
Media Server Selection – Verify and modify settings for media server selection
Upon initial installation, the following functions can be performed on the Media Server
Signaling Interfaces – Configure the addresses used for signaling
RTP Interfaces – Configure the addresses used for media
Enable RFC2833 – Enable DTMF via In-band RTP packets
Codecs – Set the preferred codec for the media server

10.1.2 Creating an Enterprise
After the initial system configuration is performed, an enterprise must be created. The
enterprise represents the entire organization to be served or a significant subset. Multiple
enterprises may exist on a system. Each enterprise has a virtual system, which is
completely independent of other enterprises.
The system administrator uses a wizard which requests the information necessary to
create and populate a new enterprise. The administrator can add, modify, delete, and
manage groups, resources and additional enterprise administrator accounts.
The following functions can be performed on the Application Server using the enterprise
web portal:
Profile Information – Add or modify enterprise profile information (name, contact
information, location, etc.)
Enterprise Administrator – Add, modify or remove enterprise administrators
Domains – Add domains associated with an enterprise (used to scope user names
for login, for SIP authentication realms and for SIP URL identification and calling)
Phone Numbers – Authorize the list of phone numbers and/or number ranges which
the enterprise is allowed to use
Services – Authorize the list of services and service quantities which the enterprise is
allowed to use
Conference Ports – Allocate conference ports which the enterprise is allowed to use


Password Rules – Define the rules that users and administrators must follow to
create and update passwords (only required if the default is not sufficient)
Web Branding – Modify the look and feel of the web portal for users and
administrators (only required if the default is not sufficient)

10.1.3 Creating a Group
After an enterprise is created, a group can be created. The group represents a site or
location of an enterprise. Multiple groups may exist per enterprise. Each group has a
virtual system, which is completely independent of other groups.
The enterprise administrator uses a wizard which requests the information necessary to
create and populate a new group. The group can add, modify, delete, and manage
departments and users, manage group services and manage group administrators.
The following functions can be performed on the Application Server using the group web
portal:
Profile Information – Add or modify group profile information (name, contact
information, location, etc.)
Group Administrator – Add, modify or remove group administrators
Departments – Add, modify or remove departments within the group
Department Administrator – Add, modify or remove department administrators
Policies – Configure group level access and authorization policies
Devices – Add, modify or remove IP phones, Integrated Access Devices, analog line
gateways, etc which the group is allowed to use
Assign Domains – Assign a domain from the enterprise to a group (used to scope
user names for login, for SIP authentication realms and for SIP URL identification and
calling)
Assign Phone Numbers – Assign the list of phone numbers and/or number ranges
from the enterprise which the group is allowed to use
Assign Services – Assign the list of services and service quantities from the
enterprise which the group is allowed to use
Assign Conference Ports – Assign conference ports from the enterprise which the
group is allowed to use
New User Services Template - Add or remove services for the user template, which
is applied when a new user is created
Calling Line ID - Set the name and/or number of the group for outgoing calls, which is
used instead of a user's name and number (optional)
Common Phone List – Setup common phone numbers for the group phone list
(optional)
Feature Access Codes - Specify feature access codes (also known as star codes)
and feature code prefixes associated with the group's services (optional)
Password Rules - Define the password rules that users and administrators must
follow to create and update passwords (optional)
Configure Device - Load or modify the default configuration file for an access device
(optional)
Digit Collection - View or modify the digit map for the group (optional)


Extension Dialing - Configure extension dialing for the group (optional)
LDAP Directory - Configure the LDAP directory for the group (optional)

10.1.4 Configuring Translations and Routing
The translations and routing policies for an enterprise are configured. This includes the
routing of private calls as well as public calls. The following functions can be performed on
the Network Server using the command line interface and web portal:
Hosting Network Elements - Configure the application servers as hosting network
elements (hosting users)
Routing Network Elements – Configure PBX and PSTN gateways as routing
network elements (inter-connection points with other networks or devices)
Media Servers – Configure media servers list, used for selecting a resource when
requested by the application server
Enterprise Entity – Setup enterprise entity for entire enterprise (corresponds to a
specific enterprise)
Voice VPN – Setup voice VPN dialing between different sites, including sites with
legacy PBXs (routing network elements)
Least Cost Routing – Setup far end hop-off routing (use gateway closest to
destination for public calls) or enterprise gateway routing (use gateway closest to
origin for public calls)
Service Center Routing – Setup translations and routing for emergency, operator
and directory assistance calls

10.1.5 Bulk Provisioning
Bulk provisioning can be used with the command line interface. The CLI can parse and
execute files containing commands, allowing a batch style processing mechanism. This is
useful for creating templates for commonly used configurations.


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