Applying Traffic Analysis to VoIP Networks 19
Example: Using the Poisson Traffic Model
Problem:You are creating a new trunk group to be utilized only by your new office and you
need to figure out how many lines are needed. You expect them to make and receive
approximately 300 calls per day with an AHT of about 4 minutes or 240 seconds. The goal
is a P.01 Grade of Service or a 1 percent blocking rate. To be conservative, assume that
approximately 20 percent of the calls happen during the busy hour.
300 calls × 20% = 60 calls during the busy hour.
(60 calls × 240 AHT) / 3600 = 4 Erlangs during the busy hour.
Solution: With 4 Erlangs of traffic and a blocking rate of 0.81 percent (close enough to 1
percent), you need 10 trunks to handle this traffic load. You can check this number by
plugging the variables into the Poisson formula, as demonstrated in Equation 1-4.
Equation 1-4
Another easy way to find blocking is by using Microsoft Excel’s Poisson function with the
following format:
= 1 – POISSON(<circuits>–1,<traffic load>,TRUE)
EART/EARC and Neal-Wilkerson Traffic Model
These models are used for peaked traffic patterns. Most telephone companies use these
models for rollover trunk groups that have peaked arrival patterns. The EART/EARC model
treats blocked calls as cleared and the Neal-Wilkinson model treats them as held. Because
the EART/EARC and Neal-Wilkerson traffic models are covered in many books dedicated
to traffic analysis, they are not covered here.
Applying Traffic Analysis to VoIP Networks
Because Voice over IP (VoIP) traffic uses Real-Time Transport Protocol (RTP) to transport
voice traffic, you can use the same principles to define your bandwidth on your WAN links.
10 12
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20 Chapter 1: Understanding Traffic Analysis
Some challenges exist in defining the bandwidth. The following considerations will affect
the bandwidth of voice networks:
Voice codecs
Voice activity detection (VAD)
RTP header compression
Point-to-point versus point-to-multipoint
Voice Codecs
Many voice codecs are used in IP telephony today. These codecs all have different bit rates
and complexities. Some of the standard voice codecs are G.711, G.729, G.726, G.723.1,
and G.728. All Cisco voice-enabled routers and access servers support some or all of these
Codecs impact bandwidth because they determine the payload size of the packets
transferred over the IP leg of a call. In Cisco voice gateways, you can configure the payload
size to control bandwidth. By increasing payload size, you reduce the total number of
packets sent, thus decreasing the bandwidth needed by reducing the number of headers
required for the call.
The number of samples per packet is another factor in determining the bandwidth of a voice
call. The codec defines the size of the sample, but the total number of samples placed in a
packet affects how many packets are sent per second. Therefore, the number of samples
included in a packet affects the overall bandwidth of a call.
For example, a G.711 10-ms sample is 80 bytes per sample. A call with only one sample
per packet would yield the following:
80 bytes + 20 bytes IP + 12 UDP + 8 RTP = 120 bytes/packet
120 bytes/packet × 100 pps = 12,000 × 8 bits / 1000 = 96 kbps per call
The same call using two 10-ms samples per packet would yield the following:
(80 bytes × 2 samples) + 20 bytes IP + 12 UDP + 8 RTP = 200 bytes/packet
200 bytes/packet × 50 pps = 10,000 × 8 bits / 1000 = 80 kbps per call
Layer 2 headers are not included in the preceding calculations.

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