IP voice solutions are a set of comprehensive systems—normally including IP PBX, voice gateway, media server, unified messages, IVR, CTI, messaging integration, contact center and media terminals, and so forth. Also, IP voice solutions can work on mixed TDM systems and IP systems.
Open Architecture and Standards
Figure 1 is a typical IP voice solution architecture; it should have two core fundamental architectural layers. One is the network layer that integrates TDM and IP’s key products and the other one is the application layer that runs any application such as IVR, CTI, CRM, ACD, and Contact Center over the network layer. Any applications under open architecture can seamlessly integrate with both the TDM and IP network.
To ensure the best usability and robustness for the IP voice solutions, the design of IP voice solutions should address the following service-level requirements:
- Open and standards-based architecture: Enable the IP voice solutions with the flexibility of a comprehensive solution portfolio that interoperates with existing TDM technologies and systems, while protecting existing investment.
- Extensibility: The IP voice solutions should have the ability to have additional functionality added or to modify existing functionality without impacting the existing functionality.
- Scalability: The IP voice solutions will provide the ability to quickly and easily extend the network infrastructure and application to handle increased contact loads or much more agents or new components and applications.
- Reusability: Addresses the ease with any functional components or concepts within the IP voice solutions can be reused by other applications.
- Maintainability: The IP voice solutions should have the ability to deal with technical issues in existing functionality without impacting other subsystem or components of the IP voice environment.
- Performance: The IP voice solutions should have good performance across all solutions. For example, good Quality of Service (QoS) mechanisms ensure high voice quality through tight control of delay, loss, and jitter.
- Manageability: The IP voice solutions must be easy to be used and managed by most the administrator and technical staff. Also, choose good network management products to provide network administration, operations, troubleshooting, configuring, fault monitoring, and element management.
IP PBX and Media Server
The IP PBX and media server is the core component of the IP voice solution. IP PBX and media server perform call processing capabilities and PBX features over the IP network infrastructure as well as extend and manage enterprise telephony features and capabilities to IP telephony network devices, media terminals, and applications such as media processing devices, messaging devices, IP phones, VoIP gateway, IVR, CTI applications, and so forth. The IP PBX and media server could be worked on single-site models and multi-site WAN models:
- Single-site call processing model: In the single-site model, each site or campus has its own IP PBX or media server to perform call processing functions; also, there are no voice calls communication over the WAN network. If you want to implement external calls or call remote sites, you can use PSTN.
- Multi-site WAN model with centralized call processing: In the multi-site WAN model, the IP PBX can either resides at a central campus or each site, and communication with remote branch offices or between sites normally takes place over the IP WAN or PSTN.
In the marketplace, the Cisco CallManager, Avaya 8xxx series Media Server, and Nortel Succession are the leading IP PBX and media server.
Voice Gateway and TDM/IP PBX integration
Voice Gateways provide a connection or a bridge between an IP telephony network and PSTN. Also, one can say that it is a connector between the TDM system and the IP network. Voice gateways play a core role in the integration between TDM PBX and IP PBX.
Voice Gateways can range from analog gateways to digital trunk gateways. Normally, the analog gateway consists of an analog station gateway and analog trunk gateways. The analog station gateways can connect IP PBX to analog telephones, IVR systems, fax machines, and voice mail systems with FXS ports. The analog trunk gateways can connect IP PBX to PSTN central office (CO) or PBX trunks with FXO ports. The digital trunk gateways connect IP PBX to the PSTN or to a PBX via digital trunks such as Primary Rate Interface (PRI), Basic Rate Interface (BRI), or T1 Channel Associated Signaling (CAS). Digital T1 PRI trunks may also be used to connect to certain legacy voice mail systems.
The Cisco, Avaya, and Nortel systems have shipped to the market much many voice gateways. For details see their Web sites.
Traditional TDM PBX can work with messaging systems such as voice mail systems. Similarly, the IP PBX also can integrate with different types of voice mail systems. Cisco did not have TDM PBX products itself, so right here is a typical example for using the Cisco IP PBX to show how IP PBX integrates with a third party’s messaging system.
- SMDI-Capable integration: Use the Simplified Message Desk Interface (SMDI) protocol when integrating voice mail systems. The Cisco CallManager IP PBX fully supports the SMDI protocol through either the Cisco Messaging Interface (CMI) service that normally runs on a server, or the Cisco VG248 Analog Phone Gateway. If you use the VG248, the VG248 itself directly provides the SMDI link.
- Non-SMDI Serial-Capable integration: In this case, you can place the VG248 between the PBX and the voice mail system, thereby enabling a dual integration with the Cisco CallManager IP PBX using a suitable Skinny Client Control Protocol (SCCP) or Media Gateway Control Protocol (MGCP) gateway that can provide analog FXS ports.
- Voice Mail Integration Using Cisco DPA: The Cisco Digital PBX Adapter (DPA) emulates both digital PBX ports as well as digital telephones, and it is available for Avaya G3 PBX and Nortel Meridian 1 PBX. Both products are designed specifically for the Avaya/Octel 200/300/250/350 voice mail systems and can be deployed in either single or dual integration mode.
- Integrating Cisco Unity: Cisco Unity can be deployed in either Unified Messaging or Voicemail-only mode.
Computer Telephony Integration (CTI) applications include both any applications that use CTI middleware (such as Genesys, Cisco ICM and Intel CT Connect) in TDM and any IP telephony applications that use the IP PBX CTI interface. In IP network, these applications are generally written using either C/C++/VB or are Java based on the IP PBX API. Also, you can write these applications over the Microsoft .NETframework using C#, ASP, VB, or .NET. You can implement either a simple screen pop application on the agent desktop or extend and customize more complex call control functionalities such as callback, chat, softphone, Web requests, ad so forth. Normally, the CTI interface can monitor and control the IP phones, CTI ports, and CTI route points. If necessary, you can integrate IVR, IP Contact Distribution (ICD), or other call center/CRM application into your CTI application.
Avaya provides an Avaya Communication Manager API that is an open, standard-based, Java and XML programming interface for developing CTI applications that work with the Avaya S8xxx series Media Server. Cisco provides both C/C++ API of Telephony Application Programming Interface (TAPI)-compatibility and Java API of Java Telephony Application Programming Interface(JTAPI)-compatibility for performing CTI applications.
You need to be aware that the IVR is going to open and standard so far. Both of the current standards are VoiceXML and SALT, which are open and standard-based programming languages. Basically, in a VoiceXML/SALT-based, speech-enabling IVR application, one server will run the speech engine for speech recognition in the user interface as well, and another one will run the voice browser for interpreting VoiceXML or SALT. If your IVR application is a pure IP-based system, you do not need the telephony hardware in your IVR platform; otherwise, you can consider allowing your IVR to run on both TDM and IP environments through adding telephony hardware in the IVR platform.
In the market, some vendors’ IVR platforms can be chosen, such as IBM WebSphere Voice Server, Microsoft Speech Server, Cisco IP-IVR, Nortel PeriPhonics, Genesys VoicePortal, and so on. So far. the MS Speech Server is SALT-based only; the other vendors’ equipment is VoiceXML-based.
Actually, many applications can work with IP voice solutions though proper integration. These applications are Contact Center, CRM, ACD, ICD, Unified Messaging, Web, Chat, Speech, and the like. What you select will depend on your real solution. You can use Cisco, Avaya, Genesys and Nortel applications; also, you can integrate a third party’s application. For instance, the Cisco, Avaya, Nortel, and Genesys systems have the IP Contact Center applications and they can easily integrate into IP voice solutions.
Notes on Design and Implementation
- Assess the current voice environment. Create an existing and a desired dial plan, inventory of features, current applications, and all hardware.
- Check your existing system and IP products vendor’s compatibility matrix. Determine whether the components of the IP voice system you want to deploy are compatible with your existing components and third party’s components.
- Build the network and data architecture, create a detailed project plan, and choose IP products’ vendors.
- Integrate the application over this network. You may need to create integrated components by coding.
- Test on different levels, such as unit test, compatibility test, network test, application test, integration test, and so forth.
About the Author
Xiaole Song is a professional in designing, integrating, and consulting CTI, Contact Center, IVR, IP Telephony, CRM, and Speech applications. He has performed various roles for Intel, Dialogic and Minacs, and so on. Feel free to e-mail any comments about this article to Xiaole Song.