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Optimizing your VoIP network

Get the most out of your VoIP network by balancing the amount of delay against the bandwidth.

Optimizing your VoIP network
Tom Lancaster

Tweaking your VoIP network can be very challenging because there are so many delicately balanced constraints and every network is different. In some networks, minimizing delay might be the guiding constraint, while others might need to conserve as much bandwidth as they can. Fortunately, there are several things that you can do to improve one at the expense of the other.

Many vendors will allow you to specify the size of the voice payload in each packet either explicitly, or by specifying the number of samples per packet. So if you have delay to spare, but want to minimize bandwidth, you could increase the size of your payload and conversely, if you have bandwidth to spare but want to minimize delay, you could reduce the size of your payload.

This works because you have a fixed protocol overhead. If you sent one sample in each IP packet, you would have to repeat that overhead for every sample. By sending two samples in each packet, you cut your overhead in half, but you increase your delay, because the first sample can't be sent until the second sample is ready to be sent.

Using G.723.1 as an example, if you send 20 bytes in each packet, you will use 16.3 Kbps per call. By doubling your data per packet to 40 bytes, you will cut your bandwidth requirements down to 8.2 Kbps per call, but your delay will be increased equally dramatically. Whether or not it's noticed depends on what the delay currently is.

The math works like this:

G.723.1 has a bit rate of 5.3 Kbps. Configured so that each packet carries 20 bytes,

20 bytes * 8 = 160 bits.

5300 bps / 160 bits = 34 packets required per second.

Each VoIP packet requires an IP header (20 bytes), UDP header (8 bytes) and RTP header (12 bytes) for a total of 40 bytes.

So, (40 bytes of headers + 20 bytes of data) * 8 bits per byte * 34 packets per second = 16.3 Kbps per call.

If you're going to apply this to a specific circuit, don't forget to add your layer 2 headers to the equation.

Thomas Alexander Lancaster IV is a consultant and author with over ten years experience in the networking industry, focused on Internet infrastructure.

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Related Book

CIM Voice Internetworking, Basic Voice over IP (Network Simulator CD-ROM)
Author : Cisco Systems Inc
Publisher : Cisco Press
Published : Oct 2000
Summary :
With CIM Voice Internetworking: Basic Voice over IP, you can master the telephony and voice internetworking knowledge you need to enhance the versatility and value of your communications infrastructure. Offering self-paced instruction and practice, this robust learning tool gives you a quick and cost-effective way to acquire Cisco knowledge and expertise.

This was last published in April 2001

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