Talking SIP
Application Server
Talking SIP™ is a leading application platform providing intelligence and revenue generating services to the next-generation network.
Media Server
Talking SIP™ is a leading media platform providing tones, voice prompts, recording and DTMF recognition to the next-generation network.
Billing Server
Talking SIP™ is a leading real-time billing platform providing pre-paid, post-paid, and multi-level billing and call cut off.
Product Summary
Talking SIP™
Talking SIP™ is an advanced software solution for next generation SIP based networks that brings the network to life. Communicating directly with industry leading softswitches, gateways, IADs and IP phones, Talking SIP™ functions as a fully integrated application, media and billing server, providing leading, in-demand enhanced service applications to the network along with real-time prepaid, postpaid and multi-level billing models with real-time call cut off.
Scaling from 30 sessions to over 9,600 sessions, Talking SIP™ is a readily extensible communications platform that can grow alongside your network over time and provides for the most efficient allocation of network resources. With a licensing model that readily supports centralized or scaled out deployment models, Talking SIP™ allows organizations the flexibility to decide how best to build out their network without additional licensing costs.
Features
Overview
Talking SIP™ is built on the Microsoft Windows® Server 2003 operating system and Microsoft SQL Server database. Talking SIP™'s low level IVR Communications Agent (IVR-CA), conforms to the IETF RFC 3261 and handles SIP signaling and media processing while supporting DTMF via RFC2833 and/or the SIP INFO method. The IVR-CA communicates directly with the Talking SIP™ application, media and real-time billing engine to provide in-demand, highly scalable and revenue-generating applications to the network. All communication between Talking SIP™ and the network members is via the SIP and RTP protocols over IP. Talking SIP™ supports the leading voice codecs (all compression and decompression is handled at the gateway/IAD/IP Phone level) and runs on readily available and inexpensive Intel Pentium based PCs configurable with RAID disk arrays and clustering.Multi-Vendor SIP Device Support
- Gateways
- Softswitches
- Session Boarder Controllers
- Proxy Servers
- Softphones/SIP phones
- IADs
Reporting
- Centralized Reporting Engine to allow reports to be managed from within the database so all additions, updates and deletions are immediately reflected on the agents' screen
- Includes over 95 configured and customizable reports
- Optional but fully integrated report designer
Customer Service Management
- Multi-level password protected access
- Audit logging of customer account transactions
- Account usage credit/debit with audit trail
- Billing and Invoice statement generation
Routing
- Real-time call processing, call cut off and routing
- Full digit manipulation of the inbound ANI and origin/carrier codes
- Full digit manipulation of the inbound DNIS
- Full digit manipulation to the terminating device with CLI/ANI manipulation
- Least cost routing and rating by time of day/holidays/CLI/number dialed
Open Architecture
- Microsoft SQL database for open and complete access for analysis and reporting purposes
- Customizable web skins for localization and corporate/reseller branding for the end user web interface
International Support
- Supports international date formats
- Supports universal time for globally deployed networks
- Supports an unlimited number of additional languages with language overrides to compensate for language differences when speaking numerical values
- 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 clustering for data redundancy in mission critical environments
- Centralized management over groups of servers
Extensibility
- Talking SIP™'s architecture separates the application from the call processing engine so that service applications can be readily created and/or modified in response to market demand and then remotely deployed to a live server without any caller interruption
Performance and Scalability
- Supports 480 sessions in a single server or blade (approx. 6,000,000 minutes/month (based on a 5 minute average call duration))
- Easily and seamlessly scale out the network by integrating multiple server chassis into a unified network
- Mix multiple services on a single server or dedicate servers to specific services based on your 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 required
- Supports the leading voice codecs (all compression and decompression is handled at the gateway/IAD/IP Phone level)
Billing
- Integrated pre/postpaid real-time billing engine with call cut off
- Three (3) additional billing models and credit caps for corporations, groups, and multi-level marketing (end user, reseller, wholesaler, etc.)
- Multiple low water mark warnings
- Blocked, flat rate, single rate and multi-tier rating
- Rating by ANI, NPA, NPA+NXX, DNIS, Account, Device, time of day, day of week, date/time range and/or destination
- Surcharging by DNIS, ANI, Info Digits, Account, Device, time of day, day of week, date/time range and/or destination
- On-line account sign up and recharge option
Custom Call Scripting
- Ability to deploy multiple instances of the same enhanced services module with unique settings that can be managed individually or across all instances
- Multiple customizable branding points
- Customizable call flows and user menus on a group or individual basis
- Customizable language selection menus
System Security
- Up to 40 digit PIN length
- Additional account reference methods such as an alias, a sequence and a reference number to reduce the need for account number disclosure
- Ability to prevent simultaneous account use
- System does not permit duplicate account generation
- CLI logging of invalid authentication attempts for risk management
- Authentication via variable length account, account+pin, ANI, DNIS, Carrier Code or IP address
Talking SIP™ in a Redundant Network Configuration

Talking SIP™ in a Standalone Network

Talking SIP™ in a Voice Over Broadband Class 5 Network
ScreenShots
Management Console
Account Management
The Telephony Management Console provides two distinct user interfaces that clearly and logically
reflect the two distinct types of users of the system. Account Management 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
The Telephony Management Console provides two distinct user interfaces that clearly and logically
reflect the two distinct types of users of the system. System Management is modeled 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 be able to propagate 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.
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.
Monitoring Real Time Activity
The Telephony Management Console provides users with the ability to view real-time, low level events
that are occurring on a local or remote communications node, including SIP message headers, incoming
call details (e.g. ANI, DNIS, Info-Digits), DTMF digits being entered and collected, voice prompts that are
being played, call progress events as well as system messages.The Telephony Management Console’s real-time Event Viewer allows system administrators and technicians the flexibility and power to be able to view real-time events on a local or remote communications node for such tasks as confirming PSTN access, confirming network or device configurations and/or troubleshooting call scenarios.
System Monitoring
Call Monitor
The Telephony Management Console provides users with the ability to view real-time, high level events
that are occurring on a local communications node including the incoming device, the origin, the
virtual port, the application servicing the call, current caller activity, current system events, the
state of the DTMF buffer and the incoming ANI and DNIS. From the Call Monitor you can view real-time
call activity, the number of callers within the IVR, the number of callers within a conversation, the
running duration of the calls, the call duration within conversation and the number of minutes
remaining on the calls.End User
End User Web Interface
The End User Web Interface provides HTML-based account management where empowerment is the
key. By empowering end-users with the flexibility to view account balances, review call histories and
recharge their accounts, there are fewer resources needed to service the customer. Customer
satisfaction is also improved as end-users appreciate the convenience of being able to review their
accounts from any web browser, on any platform, 24 hours a day, 7 days a week.




