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6/11/2001 -- Cisco has a major push on to sell AVVID products, I'm sure you've seen ads in magazines and Web sites with AVVID in big letters, but what the heck is it? AVVID stands for Advanced Voice, Video and Integrated Data. While you may not be familiar with AVVID, you probably understand voice over IP (a.k.a. VoIP). That's what AVVID uses.
Voice over X refers to a number of different ways to move voice over a data network. This allows us to combine the management and cost of local and wide area networks (WANs) for both voice and data traffic. Why have a channelized T-1 for voice and a 1.5 mb Frame Relay PVC? By using voice over X, where X is Frame Relay or IP in this example, we can combine the two.
Sounds good so far though, right? Well, not so fast! Voice data isn't the same as regular TCP or UDP traffic, although voice does use UDP. You see, voice data falls into the :interactive” category. This means that it's rather difficult to resend. Another type of interactive traffic would be the video in video conferencing. Because of the interactive and non-repeatable nature of voice over X, we need to make sure that the data streams for our voice traffic are as reliable as possible.
We accomplish reliability in the voice work by prioritizing the traffic inside the router to make sure that voice traffic always gets to go first. No one will notice a quarter-second delay in regular data traffic, but delays of 250 ms in voice can be disruptive to the conversation.
Something else that can slow voice down is large packets on slow links. A full-sized (1,518 byte) Ethernet frame that starts moving across a 56k frame relay circuit will take 218ms to be moved from the interface buffer onto the link. During this time, our voice traffic will sit and wait. Queuing only works in deciding the order in the buffer. We can combat this problem by breaking these large frames into smaller chunks. I find 500 bytes to be small enough for most networks, but each network is different.
When using VoIP, we can tell the IP packet how important it is (on a scale of 0 through 7, you're a 5), and the packet can carry that information around. Routers will process higher numbered packets before getting to lower numbered packets -- this value is set in the IP Precedence fields, three bits that are a part of TOS, Type Of Service, in the IP header. This allows us to create prioritization once and our packets will usually receive increased priority on routers we don't control. This is different from queuing because that only happens on routers we control.
How we connect our telephones to a voice over X network varies. We can connect regular analog phones to a router equipped with FXS ports. An example of this would be a 2600 series router with the voice/fax network module. This comes in both one and two slow flavors. Each slot can hold a module with two voice ports on it; the FXS ports provide for power and dial tone to analog phones. These modules take a bit to power up though, so if you turn on the router and nothing happens, wait another minute or two. There are also FXO ports that allow the router to talk to the phone company and E&M for PBX applications.
A pure VoIP network won't even have a PBX! Instead, it will have a Windows 2000 server running Cisco's CallManager software. In order to install this, you need to use a specific Win2K installation key to prevent other software from running on the Server. No Exchange 2000 installation on this guy!
Cisco has a couple of classes on the whole voice thing. CVOICE covers the voice over IP, ATM and Frame Relay aspects, while CIPT (Cisco IP Telephony) covers how to install a CallManager and IP phones. There are other courses as well but those are the two basic ones. Learn more about them by clicking on the links above.
Copyright 2001 TCPMag.com. Reprinted with permission.
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