Our last three tutorials considered the design of the VoIP network, the resources and tools that would assist for a successful implementation, and some of the business case considerations. This installment will look at some of the application and user-related issues that are worthy of attention.
Proprietary system or not? Be aware that the standards supporting multimedia networks are still relatively new, and as a result, several generations of products are available: vendor-proprietary solutions; vendor-proprietary solutions that have been tested with other vendors for interoperability; products that claim to be compliant with standards, such as T.120 and H.323; and products that have been tested for interoperability. The International Multimedia Teleconferencing Consortium (IMTC), conducts periodic interoperability tests of vendors’ equipment. If you do chose a vendor that has a proprietary solution, keep in mind that you (and your applications) will have to live with that vendor for some time to come, and as the saying goes “for better, or for worse,” may also impact your financial bottom line.
Application priorities: Real-time traffic, such as voice and video, should be given priority over more routine transmissions such as file transfers and e-mail. Several methods are possible: setting a priority by IP address, by protocol, or by using a reservation protocol, such as the Resource Reservation Protocol (RSVP). However, not all routers are configurable to support one or all of the above schemes. Check your existing routing infrastructure to see if prioritization capabilities exist.
Dialing sequences: Remember the early days of alternative carriers, when you had to remember to dial an extra dozen access digits and accounts codes to complete a long distance call? Users have higher expectations now—voice gateways must be easy for the end users to operate and must work within existing dialing plans if they are going to be accepted. As you research various products, ask for a demonstration of any unique dialing sequences that are required to access the VoIP network, and verify that they are compatible both technically and procedurally with your existing methods of establishing, transferring, and otherwise managing voice calls.
Selecting a voice codec: Within a VoIP system, the analog voice signal must be converted to a digital pulse stream before it can be placed in packets and sent over an IP-based network. A codec (short for coder/decoder) is the device that performs these voice processing functions. A variety of standards are available, including the ITU-T G.711 (64 Kbps voice), G.729 (8 Kbps), and G.723.1 (5.3–6.4 Kbps). In addition, a number of vendors have developed their own proprietary codec schemes. Each one of these alternatives has unique characteristics, including the quality and delay associated with the coding algorithm, which vary with the amount of voice information that is being crammed into the packet. It is therefore important to understand the characteristics of the voice to be transmitted, as well as the expectations of the end users. Do the network requirements include the ability to pass music-on-hold, with its higher fidelity expectations, or is voice traffic the sole need? Does the gateway product allow for multiple codec options, or is it locked into only one standard and/or algorithm? Ask some questions about the codecs to better match your application requirements.
Interoperability with existing voice systems: The VoIP gateway may need to interoperate with a number of existing and future voice processing systems such as your legacy private branch exchange (PBX), automatic call director (ACD), interactive voice response (IVR) system, and others. Do the trunk circuit port types on your PBX match those that are available from your gateway vendors? Are the signaling protocols between switches compatible with the new VoIP gear? Are you planning any future expansion or applications, such as the installation of a Web-enabled call center? Make sure that the new VoIP hardware is compatible with all other existing voice applications.
Inform the “human applications”: As the network manager, you may be excited about the new VoIP network, however some of your end users may see this as just one more new system to learn and navigate through. Good communication is essential—keep your end users informed about the benefits and applications that they will be enjoy from the new system, plus the planned implementation schedule—and endeavor to turn potential system critics into ardent system supporters.
Copyright Acknowledgement: © 2005 DigiNet ® Corporation, All Rights Reserved
Mark A. Miller, P.E. is President of DigiNet ® Corporation, a Denver-based consulting engineering firm. He is the author of many books on networking technologies, including Voice over IP Technologies, and Internet Technologies Handbook, both published by John Wiley & Sons.