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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

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.

Value Proposition

Drives Revenue and Increases Profit MarginsThrough 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 OverheadsWith 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 ChurnBy 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.

Talking SIP Core ComponentsTalking 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 ManagerThe 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.

Features & BenefitsPerformance 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.

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

The Voucher 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. The Voucher 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.

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.

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

The Voice 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.

Optional Add-On ModulesMAP 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.

Account Management

Context Sensitive Help

Intuitive Web Browser Styled Wizards with Tooltips

Graphical Toolbar

Context Sensitive Help

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.

System Management

Context Sensitive Help

High level forms interface with helpful wizards and direct table access

Graphical Toolbar

Context Sensitive Help

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

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 ManagerRedundant OptionMicrosoft SQL (Standby) Server/Secondary License Manager

Contact InformationContext Sensitive Help

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(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) 634-1523sales@ivr.comwww.ivr.com