Talking SIP Sales Presentation that provides an overview of Talking SIP, the industry's award winning and leading Application, Media and Billing Server that powers the top next-generation networks around the world.

1. ™

2. Corporate Overview
๏ Industry Leadership
๏ First commercial vendor to offer a SIP based real-time billing platform
๏ Leader in the SIP space, enhanced services, and real-time billing
๏ Experienced Management Team
๏ Built one of the fastest growing technology companies in the US (Inc. 500)
๏ Developed entire prepaid calling card system for one of Canada’s largest
incumbent telecom carriers
๏ Customer Focused
๏ Ranked in the highest tier for customer satisfaction in a Microsoft
sponsored survey
๏ Customer feedback driven product development

3. What Can Talking SIP Do?
Talking SIP is the industry leading application, media and billing server that powers some
of the largest and most recognized networks around the world. Designed specifically for
next-generation carriers and service providers both small and large alike, Talking SIP
helps them drive revenue to their networks, increase their profit margins and cultivate
subscriber loyalty through innovative in-demand services, creative billing options, and a
consolidated network architecture that greatly reduces the CAPEX and OPEX costs
normally associated with running an intelligent network.
Why Choose Talking SIP?
In today’s highly competitive telecommunication landscape it is critical that service
providers deliver differentiated offerings of in-demand applications that can attract
subscribers while providing innovative and integrated services that are easily managed
and controlled in order to retain them. The core focus of Talking SIP is just that – provide
you, the service provider, with the ability to offer innovative bundled services that attract
and retain customers while driving revenue and profitability to your network while
reducing your operating and administrative overhead.
How Does it Work?
By combining the function of multiple network appliances into a single, extensible and
scalable solution, Talking SIP reduces the level of integration, configuration and touch-
points required to get you to market and the overhead required to keep these level of
services operational. Talking SIP delivers market-ready, proven and hardened turnkey
applications that can be deployed in hours rather than weeks or months like our
competitors. In addition, Talking SIP’s architecture is designed to continually increase its
value proposition and your return on investment by allowing existing services to be
customized and allowing new services and features to be added at any time.

4. Value Proposition
Drives Revenue and Increases Profit Margins
Through innovative and in-demand enhanced services Talking SIP gives
service providers the ability to differentiate themselves from their
competition with leading, innovative, and high-margin services that empower
the end user and drive revenue to the next-generation network.
Reduces Operating Cost and Administrative Overheads
With Talking SIP’s consolidated network architecture, which combines an
application, media, billing, registration and location server into a single
product, service providers are able to reduce their capital costs along with
their network and carbon footprints while significantly lowering
administrative overhead.
Increases Customer Loyalty and Reduces Churn
By providing leading and extensible service applications, ubiquitous mobile
access and a rich and empowering management portal Talking SIP allows
service providers to bundle services, cross-subsidize service offerings and
consolidate billing to increase subscriber loyalty while building inertia to
migration.

5. Talking SIP Core Components
Talking SIP™
Talking SIP is the core call processing engine that provides enhanced services
and real-time billing to the network. Talking SIP is used to manage call
state, authentication, authorization, billing, call processing, media
processing (voice prompting, digit collection and media recording). Talking
SIP can deployed in a centralized, de-centralized or hybrid configuration
where multiple Talking SIP nodes can be deployed to support over 10,000
simultaneous callers.
Telephony Management Console™
The Telephony Management Console is the management console used to
perform account and system management to one or more Talking SIP nodes.
The Telephony Management Console is a fully secure, multi-threaded, user
group rights managed interface that provides intuitive, centralized and
simplified management for the entire Talking SIP network.
License Manager
The License Manager is used to manage the centralized license pool of
session licenses as well as to read and update license information stored in
the hardware lock to allow just-in-time upgrades. The License Manager can
allow session licenses to be partitioned on a node-by-node basis for hosted
configurations and allows for a redundant configuration for mission-critical
environments.

6. Features & Benefits
Performance and Scalability
Supports up to 2,000 sessions in a single server or blade
(approx. 60,000,000 minutes/month (based on 5 minute average
call duration and 5% blocking at the busy hour)).
Supports thousands of sessions by easily and seamlessly
integrating multiple server chassis/blades in a unified network.
Ability to mix multiple services on a single server or dedicate
servers to specific services based on network/business
requirements.
Provides simultaneous access to services for subscribers located
anywhere in the network.
Based on the SIP protocol standard -- the most flexible and
scalable call setup mechanism for supporting enhanced services.
Fully integrated application, billing and media server for
streamlined deployment and management.
Completely software based, no costly DSP resources or third-
party hardware required.
Supports the leading voice codecs (all compression and
decompression are performed in the edge device (e.g. IAD, IP
Phone, Gateway, etc.).
No additional licensing cost to deploy additional servers in the
network for centralized, decentralized or hybrid networks.
Centralized license pool to ensure the most efficient allocation
of globally deployed network resources.
Multi-Vendor SIP Device Support
IP Phones and Softphones
Integrated Access Devices (IADs)
Enterprise and Trunking Gateways
Session Border Controllers (SBCs) and softswitches
Proxy and Registrar Servers
International Support
Supports an unlimited number of languages with language overrides
to compensate for language specific number syntax and grammar.
Supports international date formats.
Supports universal time for globally deployed networks.
Languages may be assigned to a DNIS, Device,Account or manually
selectable by an unlimited number of user-definable language
selection menus.
Scalability and High Availability
Uses state-of-the-art redundancy and load balancing technology.
Database mirroring for data redundancy and high availability.

7. Enhanced Services (Included)
Debit/Calling Card Module
The Debit Card Module provides calling card functionality to
Talking SIP™. Prepaid and postpaid accounts are authenticated and
authorized by the Talking SIP™ platform and then prompted for the
desired destination number. The caller’s destination is rated and
routed, where Talking SIP™ then connects the caller through to the
desired destination. The call is tracked to detect when the called
party disconnects or the calling party signals that they wish to make
another call without having to be re-authenticated. As the calling
party’s balance nears depletion a low water mark is played and then
the callers are gracefully disconnected. Once the called and calling
parties are disconnected the account balance is immediately
updated and a call detail record and billing record are written.
Tandem Module (Class 4 Switching)
The Tandem Module allows Talking SIP™ to provide Class 4 tandem
switching, where the platform receives the inbound call request,
authenticates, rates, routes and rapidly terminates the call through
the platform based on the calling party’s dialled number. The
Tandem Module also utilizes Talking SIP™’s billing engine to
maintain billing and CDRs for each call on a prepaid or postpaid
basis. This module also supports some Class 5 Features as well.
800/900/DID Termination Module
The 800/900/DID Termination Module allows the termination of
toll or toll-free calls to a local phone number or IP-based device.
This module allows prompt and efficient re-direction of toll and
toll-free numbers to various termination numbers or IP devices
without the traditional delay and overhead. The 800/900/DID
Termination Module also provides full billing and accounting of all
traffic that passes through each inbound access number.
Voucher Recharge Module
TheVoucher Recharge Module allows service providers and
telecom operators the option of providing their customers with
the flexibility to transfer balances from one account or voucher to
another account. The module can be configured in conjunction
with the Debit Card Module, whereby the caller is provided with
the option of transferring a balance whenever he/she logs into the
system. TheVoucher Recharge Module helps to increase customer
retention by providing customers with the option of being able to
transfer balances to a familiar account number that may already be
pre-programmed into their wireless or wired telephone.
Service Charge Server
The Service Charge Server is used to process Billing Packages
(Billing Packages are used to automatically perform certain billing
or replenishment operations on accounts) to allow certain charges
(e.g. first use charges, immediate charges and reoccurring monthly
charges) to be applied to accounts. This module also supports the
optional Payment Gateway Server, to allow charges to be
automatically billed to account holders' credit cards.

8. Advertising Module
The Advertising Module is used in conjunction with the Debit
Module to allow advertising/content messages to be streamed to
callers, using a third party Advertising (or In-Call Media) Server like
Voodoovox (www.voodoovox.com), as an additional source of
revenue and/or to subsidize calling for end subscribers to
encourage usage and/or to help provider’s remain competitive.
Conference Module
The Conference Module is an optional module that brings
reservation-based and reservation-less conferencing to Talking
SIP™. Conferencing helps to bring friends, family, colleagues,
partners, suppliers and customers together by bridging time zones
and geographic boundaries in a timely, natural and convenient
manner.When face-to-face collaboration and communication is just
not feasible, Conferencing provides a great alternative that drives
inbound revenue and increases profit margins while cultivating
customer loyalty and increasing subscriber retention.
End User Web Interface
The End-user web interface allows providers to give their
customers the convenience of empowerment - the ability to access
account information, call detail records, and their billing history
whenever they desire. Used in conjunction with our Payment
Gateway Server, the Web Based Interface gives customers the
ability to offer an e-commerce distribution channel for self-service
account provisioning and account recharge to extend the reach of
an operation without consuming the costly resources.
Find Me Module (with Hosted Call Center )
The Find Me Module, also referred to as One Number Locator,
Follow-Me or Simultaneous Ring, is an optional module that
provides the convenience of a single telephone number for callers
to use to reach subscribers where Talking SIP™ will automatically
call multiple phones simultaneously (e.g. to simulate a PBX hunt
group) or sequentially (e.g. office, then mobile, and then voice mail)
to connect them. This module also supports the optional Hosted
Call Center module to allow call center or PBX queuing to local or
remote call agents.
Optional Add-On Modules
Callback Module
The Callback Module provides callback services to Talking SIP™. A
caller dials a shared or dedicated access number and then hangs up
after a certain number of rings before incurring any toll charges.
The system authenticates the caller based on the incoming ANI or
the dialed number and then calls the caller back at a preset
telephone number or from where the initiating call was placed.
Callback requests may be configured by an inbound call, a web page,
an e-mail message, an SMS message, a click to call link, as well as by
a simple but secure API.
Intelligent E-mail Agent
The Intelligent E-mail Agent allows system reports such as traffic
analysis, call summaries or billing records to be extracted from
Talking SIP™ and automatically e-mailed at predefined intervals.
System administrators can also use this agent to execute stored
procedures or custom SQL statements to assist in the maintenance
of the network and/or database.All tasks and reports can be
scheduled on a daily, weekly, monthly or an explicit date basis.

9. Optional Add-On Modules
Reminder Module
The Reminder Module provides appointment, reminder and wake-
up services to Talking SIP™. The caller is prompted for the desired
reminder time, which is confirmed and then the call is
disconnected.When the reminder's date and time arrives Talking
SIP™ places an outbound call to deliver the message, and may be
configured to navigate through a far-end IVR system, auto-attendant
or PBX. If the message cannot be delivered the system will
automatically queue the reminder to try to re-deliver the reminder.
Registration and Location Services
This option allows Talking SIP™ to act as a Registrar Server and a
Location Server. With this option SIP endpoints are able to register
with Talking SIP™ as well as be challenged when utilizing services
for the most secure authentication method. In addition, nomadic
SIP endpoints as well as SIP endpoints residing behind dynamic IP
addresses may be located in order to facilitate PSTN to IP as well
as IP to IP calling plans.
Voice Mail and PBX Module
TheVoice Mail and PBX Module is an optional module that brings
voice mail, PBX/MBX, auto-attendant (ACD) and audio-text
services to Talking SIP™. Seamlessly integrated into the other
enhanced services, this module's features include distribution lists,
message delivery to mailboxes, e-mail addresses and telephone
numbers, support for message waiting indication and stutter dial-
tone, directory dialling by first name and/or last name, auto-
forwarding, auto-carbon copying, partitioning, toll saver mode,
hands-free message retrieval, message callback, classes of service,
optional unified messaging, distributed or centralized message
storage, and unlimited time of day/day of week greetings.
Credit Card Recharge Module
The Credit Card Module allows account balances to be monitored
for prepaid debit cardholders. After an account passes a certain
threshold the caller is provided with the choice of having their
account recharged online and in real time via a credit card. This
module offers great convenience to the cardholder and generates
revenue on the platform without requiring costly human
intervention.
Report Designer
The Report Designer is an external tool that is seamlessly
integrated into the TMC to allow users the flexibility to create new
reports or extend existing reports. All reports created in this
report designer can be loaded into the database and immediately
made available to users. The report designer is a standard drag and
drop report designer to graphically and visually construct a report.
Payment Gateway Server
The Payment Gateway Server provides an interface between Talking
SIP and merchant processors to facilitate credit card transactions
for account subscription and/or recharge. The Payment Gateway
Server supports leading merchant processors and also has an XML
interface to support third-party integration with other processors.
This Add-on module may be integrated into the Credit Card
Recharge module, the Service Charge Module (for automated top-
up and service charges), as well as the optional end user web
interface.

10. Optional Add-On Modules
MAP Portal Server
The MAP Portal Server allows Talking SIP to integrate into the
mobile core network, communicating with a MAP Portal to provide
authentication, authorization, billing and services to MNEs, MNOs,
and/or MVNEs and their mobile subscribers. The MAP Portal
Server provides support for processing requests for callback,
CAMEL, SMS, Data, feature codes, voice mail, and Class 5 call
forwarding.
Web Services API
The Talking SIP Web Service API is a secure, scalable and extensible
interface that has been developed around a RESTful
(representational state transfer) interface and the lightweight JSON
(JavaScript Object Notation) data-interchange format to allow
service providers and carriers to seamlessly integrate Talking
SIPTM into any application, web site, CRM, portal, or operational
support system (OSS) while having the flexibility to be able to
leverage the development tools they are most comfortable and
productive with.

11. Account Management
Context Sensitive Help
Intuitive Web
Browser Styled
Wizards with
Tooltips
Graphical Toolbar
Microsoft Outlook
Bar for Easy and
Direct
Access
Multi-threaded
Management
Interface for Time
Tasks
Account Management in the Telephony Management Console™ is modeled after a Microsoft Outlook™
type interface with Outlook™ bars and web styled wizards with commonly accessed items towards the top
and left. Account Management encompasses Account/Customer Management to Invoicing, Customer
Relationship Management, Rating and Reporting.
Context sensitive help with a menu simulator is only an F1 key press away. The Telephony Management Console™
is a fully secure interface that allows all of the functionality within the system to be granted and/or revoked on a
User Group basis to ensure that users are only provided with just enough access to perform their duties within
the network.

12. System Management
Context Sensitive Help
High level forms
interface with
helpful wizards
and direct table
access
Graphical Toolbar
Centralized Multi-
Node
Management
Multi-threaded Management
Interface for Time Intensive
Tasks
System Management in the Telephony Management Console™ is modelled after a Microsoft Management
Console™ type interface that system administrators and technicians are familiar with, as it is a common interface
used to manage Windows based machines.
The management of database tables, currencies, an unlimited number of languages and language groups, as well as
the provision of SIP based devices, mappings to applications by Device and/or DNIS and route management are all
performed within System Management. The Telephony Management Console™ provides the ability to manage
multiple communications nodes centrally, as well as propagates setting changes across multiple instances of an
application and/or across one or more communications nodes in real-time, through a single operation, without
caller interruption.
Microsoft
Management
Console Interface
for Direct Access

13. Network Diagram
Network
Firewall/
Session Border
Controller (SBC)
Origination/Termination
SIP Trunking Providers
and Endpoints
Router/Switch
End User Web
Interface
Telephony
Management
Console (TMC)
Telephony Management
Console (TMC)
Talking SIP
Communications Node 1
Talking SIP
Communications Node N+1
Microsoft SQL
Server/License Manager
Redundant Option
Microsoft SQL (Standby)
Server/Secondary License Manager

14. Contact Information
555 West Fifth Street
31st Floor
Los Angeles, CA 90013
Phone:
Fax:
Support:
E-mail:
Website:
(866) 856-0301 Toll Free U.S. and Canada
(213) 634-1522 Direct
(310) 943-2722
(866) 856-0303 Toll Free U.S. and Canada
(213) 634-1523
sales@ivr.com
www.ivr.com