Ordering VoIP Service

When you order VoIP service for a business, you need to meet certain hardware requirements, and the testing requirements for your VoIP system are similar to a TDM voice installation. The residential VoIP offerings, on the other hand, are designed to be as simple as possible, and they include plug-and-play hardware and software installations. The basic hardware and service requirements for all VoIP applications (whether for business or home use) are the same:

• You must have a dedicated Internet connection.

• You must have a VoIP phone system, or an adapter that can translate the

VoIP protocol to a protocol that your existing phone system can use.

Larger applications for big businesses with hundreds of employees may include data integration, videoconferencing, and Web conferencing — all on the same Internet line. Of course, having these additional perks requires more Internet bandwidth and more complex hardware to manage all the individual pieces of the network.

 Tip  If you want to use VoIP technology on a small scale (for home use), you may need to purchase an inexpensive VoIP adapter (it is actually a small VoIP-to-TDM gateway) and a DSL or digital cable Internet connection. You can spend as much, or as little, money as you want on your VoIP hardware.

 Remember  The key element to keep in mind is that generally the more complex your hardware, the more features available to you. But also remember that with great power comes great responsibility, and more advanced hardware also places a greater burden on you for troubleshooting any issue you may encounter.

Choosing your VoIP hardware

The hardware you need to buy for VoIP service depends on the complexity of the VoIP network you create. If you have one phone, you need a single piece of hardware. More likely, however, you want a complex network of voicemail servers, additional end users, and a host of advanced features. If so, you need a small squad of hardware to take care of all your requirements.

Capacity concerns are based on concurrent calls

 Remember  VoIP is a very fluid and unstructured protocol when it comes to bandwidth consumption. When speaking about everyday TDM voice circuits, your biggest concern has to do with how many minutes you plan to talk. With VoIP, the question is more frequently about the number of concurrent calls your system can accept.

The TDM world is very structured; no matter what you do, a normal voice T-1 line can only handle a maximum of 24 phone calls at any one time. VoIP is a completely different animal, and depending on your codec, you may be limited to a few as 15, or as many as 76 calls on the same T-1 of bandwidth. Your carrier will assign ports based on your needs. Telling your VoIP carrier that you want to talk 100,000 minutes a month (as you would with a TDM order) means nothing for provisioning, but it may influence the per-minute rate the carrier offers you. With VoIP it’s conceivable (if unlikely in most situations) that you could make 100,000 minutes of calls within a time frame of 2 hours a day, over two T-1s using G.729.

 Tip  Speak to your hardware vendor for assistance on setting up these more complex systems, or for a referral to a business that specializes in VoIP technology and hardware if your vendor isn’t familiar with the technology.

Here’s a list of hardware you can expect to purchase:

• An Analog Telephone Adapter (ATA): An ATA is the most basic level of hardware required for home applications and is installed between the wall jack for your dedicated Internet connection and your regular telephone. ATAs convert the VoIP signal coming from your Internet connection into a signal your phone can use. After an ATA is in place, you dial out and receive calls just as you did before you had VoIP. An ATA is a common piece of hardware for home use; you will probably need something more complex for a business application with more features.

• A softphone: A softphone is a software program for your computer that enables it to function like a phone. Softphones usually look like the keypad for a cellphone and use the speakers and a microphone on your computer to transmit and receive your conversation. I have used a few of the softphones available, and I don’t think the quality is wonderful. Usually, the biggest limitation to quality is the fact that if the speakers and microphone for your computer are crummy, so is your VoIP conversation. If you have a great computer and are comfortable with using your computer to make calls, a softphone may be your phone of choice.

• A gateway: A gateway is a device that converts VoIP to TDM and TDM to VoIP. If you are deploying VoIP service and have any non-VoIP phones that need to use the service, you need a gateway.

• An Edge Proxy Server (EPS): An EPS is the first piece of hardware a VoIP call comes into contact with on your network; it’s used for larger VoIP applications. This server receives and routes the signaling and may also route the RTP. Because the RTP and signaling are separate, the RTP could be directed to another piece of hardware for voicemail, music on hold, or handling by an automatic call distribution (ACD) program that prompts you to “Press 1 for Sales, press 2 for Customer Service . . .”, and so on.

• A media server: Media servers are computer servers that receive and process the RTP stream in a VoIP call. The media server may be a part of the EPS, or it may be a completely separate device.

Conserving bandwidth by controlling the RTP stream

The largest consumption of bandwidth on a VoIP call is the RTP stream, because it has to house and transport 64 kbps of voice communication. The signaling is very small in comparison and is the only real part of a VoIP call that must be handled by the EPS. Figure 15-4 shows the bandwidth requirements for a VoIP call where the RTP is transmitted through the EPS.

Figure 15-4: Bandwidth consumed by the EPS during a VoIP call.

As you can see, both the 64 kbps RTP stream and the smaller overhead stream are handled by your EPS. If the IP phone you are transferring the call to is not part of your local area network (LAN), the call is sent back to your IP connection, and the RTP alone ends up using at least 128 kbps — for just one call.

Getting more information on VoIP protocols

 Tip  I understand that VoIP isn’t for the faint of heart. There’s a lot to know before you make a commitment to tying your phone service to your Internet connection. Perhaps because VoIP is so tied to the Internet, I thought it might be good to give you some Internet-based resources to answer your questions. In addition to picking up a copy of VoIP For Dummies, by Timothy V. Kelly (Wiley), you can also find more information online:

• A good Web site for information about VoIP protocols, H.323, and SIP is www2.rad. com/networks/2001/voip/prtcls.htm.

• If you’re looking for a good Web site with information to answer your RTP-related questions, visit www.cs.columbia.edu/~hgs/rtp/faq.html.

Passing the Interoperability test

After you order an Internet connection and purchase the required hardware, you may still have to pass an interoperability (InterOp) test. The only way you should be allowed to skip the InterOp test is if you are ordering very basic, preconfigured plug-and-play hardware from your carrier. If your carrier sends you the hardware, you should receive instructions on setting it up. In such a case, InterOp testing isn’t as crucial.

However, the InterOp test becomes very important when you have ordered your own hardware and you are beginning a new VoIP relationship with a carrier. VoIP carriers have their own default transmission configurations, and you will either be able to modify that default, or be forced to live with it, depending on the construction of your carrier’s network. Your carrier’s switch may have a global setting for features like SIP-T; SIP-T sends ANI Infodigits, which identify whether a call was made from a pay phone, hospital, or a prison (to name a few locations).

Before you can enter InterOp testing, you have to fill out an InterOp questionnaire. This document, supplied by your carrier, not only tells your carrier important information about your network configuration and projected call volume, but it also shows your level of knowledge of VoIP. A standard InterOp questionnaire asks you for some of the following information:

• Your IP protocol (H.323 or SIP): Your carrier needs to know this info to establish the most basic level of connectivity between your router and your carrier.

• The H.323 or SIP hardware, location, and IP address: The physical location of your hardware may determine which Edge Proxy Server your carrier uses. Most carriers don’t have thousands of servers to receive customers across North America, so the location of your hardware determines which regional server is assigned.

• RTP media gateway: If you’re using a separate media gateway to transmit your RTP stream, your carrier needs to know where it is located, its make and model, as well as its IP address.

• Other SDP and/or RTP hardware: Many customers have a separate server with Automatic Call Distribution features so you can “Press 1 for Sales,” or “Press 2 for Customer Service.” If you’re planning to use different pieces of hardware to provide additional features, your carrier needs to know the function of each piece, as well as info about the make, model, and IP address. The information you provide about your network design helps the carrier to troubleshoot complex issues you may encounter later.

• Codecs you want to use: Your carrier may not offer every codec you want to use, so be sure to ask your carrier which codecs are available. For example, the T.38 FoIP codec exists, but hasn’t been fully deployed by all carriers.

The InterOp questionnaire has even more questions on it, but its main purpose is to communicate to your carrier your business model and the hardware you are using to provide VoIP service. Many carriers also ask for a diagram of your network and the hardware involved.

InterOp testing for compatibility

VoIP has millions of options, variations, and nuances that can prevent you from completing a successful call. The InterOp questionnaire allows your carrier to determine all the types of calls you will be making. The point is to test every variety of call your hardware is capable of, preventing protracted troubleshooting in the long run. Your carrier determines the criteria for your InterOp testing and what decides when the test is completed.

This process can take as little as 30 minutes, or as long as months. The more knowledgeable you are about your hardware, the less time consuming InterOp testing is.

 Remember  If you aren’t required to complete an InterOp test and are using hardware that wasn’t provided and configured by your carrier, you should ask for an InterOp test. Just because your carrier thinks it isn’t necessary doesn’t mean that you can’t require it. If the carrier doesn’t require an InterOp test, call the carrier’s customer service department and open a technician assistance (tech assist) ticket to schedule some testing time with a VoIP tech.

 Tip  If you are new to VoIP, hire a qualified technician. You may be having a wonderful time discovering the nuances of SIP, but eventually you will reach the limits of your knowledge and begin grasping at straws. Simply pay someone else with the experience to finish the job so that you can eliminate your frustration with VoIP.