Genesys Voice Platform is a software-only solution that runs on off-the-shelf hardware and software. Genesys Voice Platform is based on VoiceXML and interoperates with any voice application that is written to this standard, empowering companies to conduct business interactions 24×7, transforming the phone from a simple communications device into a powerful, anywhere-anytime information access tool.
Tightly integrated into the Genesys product offering, Genesys Voice Platform is uniquely positioned to deliver an integrated interaction management solution that seamlessly blends self-service resources with human agents and provides enterprises a common view of the customer. Self-service resources are deployed as virtual extensions of the contact centre staff, enabling around-the-clock service, freeing agents from answering basic, repetitive questions and allowing them to focus on more complex interactions that require human involvement.
Genesys has two VoiceXML Platforms the Genesys Voice Platform Enterprise Edition (GVP: EE) and the Genesys Voice Platform Network Edition (GVP: NE).
GVP: EE is a voice self-service platform offering telephony communication software for carrier and PBX networks connectivity, speech engine support for advanced caller interactions, an open interface for application integration and a set of tools for application development and reporting.
GVP: EE is integrated in the Genesys Suite.
GVP: NE is offered for service providers interested in leveraging the capabilities of Genesys Voice Platform as an in network solution, or large, multi-site enterprises with plans to deploy a large number of ports. GVP: NE provides all of the capabilities available in the enterprise solution with added functionality, enabling multitenancy of applications within a managed service network or an enterprise.
For service providers, GVP: NE provides an enhanced communications platform that enables internal, consumer and business services on a commonly managed voice infrastructure. Service providers can realize incremental revenue by providing enhanced services, such as hosted IVR, speech-enabled self service, and integrated contact center solutions.
The following features are supported by GVP: EE and GVP: NE:
Support for the World Wide Web Consortium (W3C) in Voice Extensible Markup Language (VoiceXML) Version 2.1, W3C Recommendation (June 13, 2005). However, the following tags are not supported in this release:
# VXML CTI extensions and TXML call control extensions continue to be supported
# VXML extensions to support passage of custom payload to and from application using the SIP INFO method
# For more information, refer to the Genesys Voice Platform 7.2 VoiceXML2.1 Reference Manual.
# Working draft of Call Control XML (CCXML) supported
# Restricted availability feature, available for development and limited production rollouts
# Limited implementation of CCXML tags:
# Call setup
# Call teardown
# Call transfer
VoIP Standards and Drafts
# Session Initiation Protocol (SIP, RFC 3261)
# RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals (RFC 2833)
# SIP INFO (RFC 2976)
# SIP REFER (RFC 3515)
# Session Timers in the Session Initiation Protocol (SIP)
# Basic Network Media Services with SIP
# Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping (RFC 3398)
# H.323, ITU-T H.323 Recommendation (07/03), Packet-based Multimedia Communications Systems.
# Session Description Protocol (SDP, RFC 2327)
# Real Time Transport Protocol (RTP, RFC 1889)
# A Media Resource Control Protocol Developed by Cisco, Nuance, and Speechworks
Genesys Voice Platform interfaces with other Genesys products through Genesys Framework
Genesys Voice Platform interfaces with other Genesys products through Genesys Framework. Calls, along with any data associated with them, are passed to the Genesys IVR Server from the IVR Interface (IVR Client), which resides on the IP Communication Server (IPCS) or Voice Communication Server (VCS). The IVR Server, in turn, transmits the information on to other elements of the Genesys suite for integrated call-handling purposes.
The IPCS/VCS includes an IVR Server Client that communicates with a Genesys IVR Server 7.x on a remote host. The IVR Server or IVR Client detects new calls arriving at the IPCS/VCS and the termination of existing calls.
The IVR Server is contacted at call setup. In a IPCS/VCS Behind-the-Switch configuration, the IVR Server furnishes the ANI and DNIS of the incoming call. During call setup, the IVR Server, along with the URS controls the VoiceXML script executed by the IPCS/VCS, which relays call-related data to IVR Server. When necessary, the IVR Server orchestrates the transfer of the call to the appropriate agent at the end of the call flow for the voice application.
Media Resource Control Protocol (MRCP)
The Media Resource Control Protocol (MRCP) IETF draft was created by Cisco, Nuance and SpeechWorks in 2001. It is a communication protocol which allows speech servers to provide various speech services (such as speech recognition and text to speech) to its clients. Typically, this means the server software will be running on one computer and clients can send MRCP messages to the server over a network. GVP supports Media Resource Control Protocol version 1.
The web server
The web server is a physical, customer-provided server where GVP applications reside. A standard Web Server such as:
# Microsoft Internet Information Server 5.0 or 6.0 depending on which version of Windows Server.
# Jakarta Tomcat on Windows
# Apache on Windows/Unix (if stand alone web server) must be added to the GVP deployment architecture .
Requests and information exchange on the VCP/IPCS are handled in a similar way as normal html requests, but the markup language is VoiceXML. The IPCS or VCS, which contains the VoiceXML browser, parses the VoiceXML and executes call handling (answering, bridging, disconnecting calls), media management (play greetings, prompts, and messages using cached voice files and text-to-speech), and user input (collect touchtone digits, contact speech recognition engines).
VoiceXML enables a web voice application to drive an interaction with a caller in the same way that it would interact with a desktop web browser to render a screen and react to keyboard or mouse input. As with the desktop browser, any changes to the application software at the web server become effective the next time a page is requested.
The Voice Application Reporter (VAR)
The Voice Application Reporter (VAR) server is a sub-component of Genesys Studio, and as such can be used within either GVP: EE or GVP: NE installations.
VAR takes logging events from Genesys Studio generated applications. The logged information is routed and stored on a separate server, where the Voice Application Report server can compute summary statistics. The reporting client offers pre-defined reports on the basic call information handled by the Voice Platform including details by application and call outcome. This application information is available on an hourly, daily, and/or weekly basis. From any browser, the supervisor or any other interested party can examine call summaries or individual call details over selected time intervals and by application type. This information can be exported to spreadsheets for further analysis.
Although VAR is designed as a single-tenant product it can still be used by GVP: NE users by installing one instance of VAR per tenant.
Genesys Studio is a optional GUI tool for building, testing, and debugging voice applications using the â€œDrag-and-Dropâ€ paradigm. This approach frees developers from having to learn and write VoiceXML tags and allows them to focus on the call flow of voice applications, which further accelerates the development process.
The IP Communication Server (IPCS) handles
The IP Communication Server (IPCS) handles calls through Voice Over IP (VoIP), using the Session Initiation Protocol (SIP). The IPCS parses, interprets, and executes the VoiceXML commands in the XML documents served by the web server. The IPCS also integrates with text-to-speech (TTS) and automatic speech recognition (ASR) software through the MRCP standard.
The VCS software has the following primary functions
# Communicates with the Dialogic circuit boards and instructs the boards what action to take, such as playing prompts or listening for DTMF
# Parses, interprets, and executes the VoiceXML commands in the XML documents served by the web server
# Integrates with text-to-speech (TTS) and automatic speech recognition (ASR) software.
In an In-Front-of-the-Switch configuration, the VCS uses a Dialogic circuit board that connects to T1 or E1 line(s) that carry the voice calls. For a Behind-the-Switch configuration, the VCS has a Dialogic circuit board that connects to the premise-based switch.
GVP is composed of the following software components:
Voice Portal Manager (VPM) or Element Management Provisioning System (EMPS) – Provides centralized configuration of VCS/IPCS and application configuration. It is a web based interface accessible via Internet Explorer. The VPM/EMPS configuration enables the VCS/IPCS to correctly identify the desired voice application using the DNIS of the incoming call. The primary voice application configuration features of the VPM/EMPS allow authorized administrators to:
# Add, modify, and delete voice applications
# Configure the voice application URL
# Assign specific external telephone numbers to a voice application
# Enable ASR capture (when ASR is enabled)
# Configure a fail over voice application URL upon failure of the primary URL and configure failover transfer numbers.
Genesys Studio – Genesys Studio is a Windows-based GUI tool that can be used to build and test voice applications using a â€œdrag and dropâ€ paradigm.
Voice Application Reporter (VAR) – A browser-based reporting program that runs on a separate server, and provides reporting on both JSP and ASP applications. Studio applications store call information on a web server, and VAR can report on call statistics by application summarizing calls by time of day, application, and outcome.
Voice Communication Server (VCS) – Handles calls through TDM using Dialogic telephony boards. The VCS parses, interprets and executes tags contained in VoiceXML documents, as well as terminates and processes voice calls. The VCS integrates with text-to-speech (TTS) and automatic speech recognition (ASR) through MRCP standard.
Internet Protocol Communication Server (IPCS) – Handles calls through Voice Over IP (VoIP). The IPCS parses, interprets and executes tags contained in VoiceXML documents, as well communicates with Media Gateways. Like the VCS, IPCS also integrates with text-to-speech (TTS) and automatic speech recognition (ASR) through MRCP standard.
IP Call Manager (IPCM) – Consists of a set of components that coordinates and assigns the resources needed to process a VoIP call using Session Initialization Protocol (SIP). Includes SIP/H.323 Session Manager and Resource Manager.
Resource Manager – Maintains resource states for IPCS and Media Gateway resources. The Resource Manager maintains dynamic status of each resource whether it is in use or available. The Resource Manager can perform load balancing of the IPCS resources using a round-robin algorithm. It can also select the IPCS resource based on specific feature requirements. Resource Manager checks the health of each resource periodically and marks the resource as healthy/not healthy.
SIP Session Manager – Acts as a SIP Proxy to relay SIP messages between a Media Gateway or Signaling Gateway and IPCS. When a new call arrives at the SIP Session Manager, it fetches the application profile from the VPM based on the dialed number or URI. With the help of the Resource Manager, the SIP Session Manager routes the call to an IPCS based on the application profile requirement. When the call is disconnected, it informs the Resource Manager to free up the IPCS resource.
H.323 Session Manager – Acts as an H.323 Proxy. It processes H.323 messages received from Media Gateway or Signaling Gateway, and forwards equivalent SIP messages to IP Communication Server. When a new call arrives at the H.323 Session Manager, the H.323 Session Manager, with the help of the Resource Manager routes the call to an IPCS, based on the application profile requirement. When the call is disconnected, it informs the Resource Manager to free up the IPCS resource.
Web Server A customer-provided server where the VoiceXML (2.0 or 2.1) application resides. Any HTTP 1.1 compliant web server can be used. When an inbound call arrives the DNIS is mapped to a URL, as provisioned in the VPM/EMPS and the VoiceXML application is retrieved from the web server.
Genesys Voice Platform (GVP) is a software-only product that brings Internet technologies to the world of voice.
GVP removes constraints of legacy IVR systems by offering standards-based development, flexible deployment options, simplified integration and improved time to rollout for speech-directed voice applications.
Genesys Voice Platform collects basic call information such as originating and dialled number, and customer interaction information, which is passed to Genesys Enterprise Routing for intelligent customer segmentation and call routing instructions. Using the call information, companies can manage customer interactions in a more personalized, consistent and efficient manner, maximizing the productivity of contact centre resources. This tightly integrated solution provides companies with the ability to segment and prioritize customer interactions according to business value, desired service level or specific needs.
One of Genesys Voice Platforms many functionalities is prompting the customer and accepting responses using voice (as well as digits). Thus, the customer can conduct business solely by voice, thereby freeing an agent for other customer interactions. For example, a customer calling a bank could transfer funds from checking to savings or a customer calling an airline could check the status of a flight by interacting with the systems directly.
VoiceXML is a language for creating voice-user interfaces, particularly for the telephone. It uses speech recognition and touchtone (DTMF keypad) for input, and pre-recorded audio and text-to-speech synthesis (TTS) for output. It is based on the Worldwide Web Consortium’s (W3C’s) Extensible Markup Language (XML), and leverages the web paradigm for application development and deployment. By having a common language, application developers, platform vendors, and tool providers all can benefit from code portability and reuse.
With VoiceXML, speech recognition application development is greatly simplified by using familiar web infrastructure, including tools and Web servers. Instead of using a PC with a Web browser, any telephone can access VoiceXML applications via a VoiceXML â€œinterpreterâ€ (also known as a â€œbrowserâ€) running on a telephony server. Whereas HTML is commonly used for creating graphical Web applications, VoiceXML can be used for voice-enabled Web applications.
A VoiceXML application consists of several components, as shown below:
# Application Server – Typically a Web server, which runs the application logic, and may contain a database or interfaces to an external database or transaction server.
# VoiceXML Telephony Server – A platform that runs a VoiceXML interpreter that acts as a client to the application server. The interpreter understands VoiceXML dialogs and controls speech and telephony resources. These resource include ASR, TTS, audio play and record functions, as well as a telephony interface (TDM or VoIP).
# Internet Style Network – A TCP/IP-based packet network that connects the application server and telephony server via HTTP.
# Telephone Network – Typically the Public Switched Telephone Network (PSTN), but could be a private telephone network (i.e. PBX), or VoIP packet network.
# Caller – Any telephone that can connect to the telephone network.
Communication between the web server and GVP is analogous to the desktop web browser model. In the case of the Internet, the Web Server produces HTML, which is understood by a browser Internet Explorer for example and rendered on the screen by the video card. In the case of GVP, the Web Server produces VoiceXML which is understood by the VoiceXML browser located on the VCS/IPCS, and rendered into voice interactions, which includes:
# Call handling – answering, bridging, disconnecting calls
# Media management – playing of prompts, using pre-recorded .vox or .wav files or text to speech
# Caller input – either DTMF (dual tone multi-frequency) or speech recognition.
Voice applications are generally developed as Active Server Pages (ASP) or Java Server Pages (JSP).
In either case the Web Server can query a database and retrieve information which is then used in the corresponding output.