Introduction to Voice Traffic Engineering
This section introduces voice traffic engineering. Voice traffic engineering is the science of selecting the correct number of lines and the proper types of service to accommodate users. From trunks and DSPs to WAN and campus components, detailed capacity planning of all network resources should be considered to minimize degraded voice service in integrated networks.
The bandwidth requirements for voice traffic depend on many factors, including the number of simultaneous voice calls, grade of service required, codec and compression techniques used, signaling protocol used, and network topology. The following sections introduce how to calculate the WAN bandwidth required to support a number of voice calls with a given probability that the call will go through.
Terminology
This section introduces the following terminology used in voice traffic engineering:
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Blocking probability
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Grade of Service (GoS)
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Erlang
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Centum Call Second (CCS)
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Busy hour
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Busy Hour Traffic (BHT)
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Call Detail Record (CDR)
Blocking Probability and GoS
The number of simultaneous conversations affects the voice traffic. Users vary widely in the number of calls they attempt per hour and the length of time they hold a circuit. Any user’s attempts and holding times are independent of the other users’ activities. A common method used to determine traffic capacity is to use a call logger to determine the number of simultaneous calls on the network and then determine the probability that exactly x simultaneous calls will occur. Voice systems can be provisioned to allow the maximum number of simultaneous conversations that are expected at the busiest time of the day.
Erlang
The Erlang is one of the most common measurements of voice traffic.
For example, if a trunk carries 12.35 Erlangs during an hour, an average of a little more than 12 lines (connections) are busy. One Erlang indicates that a single resource is in continuous use. The traffic measurement in Erlangs is used to determine whether a system has too many or too few resources provisioned.
CCS
Note | Centum means one hundred. |
A system port that can handle a continuous one-hour call has a traffic rating of 36 CCSs (or 1 Erlang). Station traffic varies greatly among users, but the typical range is approximately 6 to 12 CCSs per port. If no exact statistical data exists, assume that the average typical trunk traffic is 30 CCSs per port.
For example, one hour of conversation (one Erlang or 36 CCSs) might be ten 6-minute calls or 15 4-minute calls. Receiving 100 calls, with an average length of 6 minutes, in one hour is equivalent to ten Erlangs, or 360 CCSs.
Busy Hour and BHT
For example, if you know from your call logger that 350 calls are made on a trunk group in the busiest hour and that the average call duration is 180 seconds, you can calculate the BHT as follows:
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BHT = Average call duration (seconds) * calls per hour/3600
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BHT = 180 * 350/3600
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BHT = 17.5 Erlangs
CDR
A CDR is a record containing information about recent system usage, such as the identities of sources (points of origin), the identities of destinations (endpoints), the duration of each call, the amount billed for each call, the total usage time in the billing period, the total free time remaining in the billing period, and the running total charged during the billing period. The format of a CDR varies among telecom providers and call-logging software; some call-logging software allows the user to configure the CDR format.
Erlang Tables
Erlang tables show the amount of traffic potential (the BHT) for specified numbers of circuits for given probabilities of receiving a busy signal (the GoS). The BHT calculation results are stated in CCSs or Erlangs. Erlang tables combine offered traffic (the BHT), number of circuits, and GoS in the following traffic models:
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Erlang B: This is the most common traffic model, which is used to calculate how many lines are required if the traffic (in Erlangs) during the busiest hour is known. The model assumes that all blocked calls are cleared immediately.
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Extended Erlang B: This model is similar to Erlang B, but it takes into account the additional traffic load caused by blocked callers who immediately try to call again. The retry percentage can be specified.
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Erlang C: This model assumes that all blocked calls stay in the system until they can be handled. This model can be applied to the design of call center staffing arrangements in which calls that cannot be answered immediately enter a queue.
Note | Erlang tables and calculators can be found at many sites, including http://www.erlang.com/. |
Erlang B Table
Figure 8-36 shows part of an Erlang B table. The column headings show the GoS, the row headings show the number of circuits (the number of simultaneous connections), and the table cells indicate the BHT in Erlangs for the specified number of circuits with the specified GoS.
Erlang Examples
Having established the BHT and blocking probability, the required number of circuits can be estimated using the Erlang B traffic model. For example, given BHT = 3.128 Erlangs, blocking = 0.01, and looking at the Erlang table in Figure 8-36, the number of required circuits is eight.
As another example using Figure 8-36, 4.462 Erlangs of traffic is offered for ten circuits (simultaneous connections) with a GoS of P01 (1 percent block probability). 4.462 Erlangs equals approximately 160 CCSs (4.462 * 36). Assuming that there are 20 users in the company, the following steps illustrate how to calculate how long each user can talk:
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BHT = Average call duration (seconds) * calls per hour/3600
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Average call duration = 803 seconds = 13.3 minutes
In another example, six circuits at P05 GoS handle 2.961 Erlangs. 2.961 Erlangs equals approximately 107 CCSs (2.961 * 36). Assuming that the company has ten users, the following illustrates how to calculate how long every user can talk:
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BHT = Average call duration (seconds) * calls per hour/3600
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2.961 = Average call duration (seconds) * 10/3600
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Average call duration = 1066 seconds = 17.8 minutes
Trunk Capacity Calculation Example
The objective of this example is to determine the number of circuits, or the trunk capacity, required for voice and fax calls between each branch office and an enterprise’s headquarters office. The following assumptions apply to this sample network:
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The network design is based on a star topology that connects each branch office directly to the main office.
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There are approximately 15 people per branch office.
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The bidirectional voice and fax call volume totals about 2.5 hours per person per day (in each branch office).
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Approximately 20 percent of the total call volume is between the headquarters and each branch office.
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The busy-hour loading factor is 17 percent. In other words, the BHT is 17% of the total traffic.
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One 64-kbps circuit supports one call.
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The acceptable GoS is P05.
Following are the voice and fax traffic calculations for this example:
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2.5 hours call volume per user per day * 15 users = 37.5 hours daily call volume per office
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37.5 hours * 17 percent (busy-hour load) = 6.375 hours of traffic in the busy hour
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6.375 hours * 60 minutes per hour = 382.5 minutes of traffic per busy hour
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382.5 minutes per busy hour * 1 Erlang/60 minutes per busy hour = 6.375 Erlangs
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6.375 Erlangs * 20 percent of traffic to headquarters = 1.275 Erlangs volume proposed
To determine the appropriate number of trunks required to transport the traffic, the next step is to consult the Erlang table, given the desired GoS. This organization chose a P05 GoS. Using the 1.275 Erlangs and GoS = P05, as well as the Erlang B table (in Figure 8-36), four circuits are required for communication between each branch office and the headquarters office.
Off-Net Calls Cost Calculation Example
This example calculates the off-net cost of calls between two locations, New York and London, as shown in Figure 8-37. The PSTN path is used when the transatlantic tie line cannot accept additional on-net calls.
Assume that all calls between these two sites use 64 kbps of bandwidth, which corresponds to one circuit, and that a GoS of .03 is acceptable. How many minutes per month of calls use off-net calling because of the service block on the transatlantic tie line? The transatlantic tie line can simultaneously carry a maximum of ten calls. In the calculation, we assume that a 1-minute call between New York and London costs $.10.
Note | The $.10 per minute rate is used here for ease of calculation. |
The calculation is as follows:
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According to the Erlang B table in Figure 8-36, 5.53 Erlangs can be offered at P03 and ten circuits.
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At P03, 3 percent of the 5.53 Erlangs of calls are overflowed and sent off-net.
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Therefore, in the peak hour, .03 * 5.53 Erlangs * 60 minutes = 10 overflow minutes.
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Assume that there are two peak hours per day and 21 business days per month. Therefore, 21 days * 2 peak hours per day * 10 overflow minutes = 420 overflow minutes per month.
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420 overflow minutes per month * $.10 per overflow minute = $42.00.
The calculation shows that 420 minutes per month of off-net calling between New York and London is used, costing $42.00. This cost should be compared to that of adding circuits between New York and London to see whether it is worth adding bandwidth.
Calculating Trunk Capacity or Bandwidth
The first component of this formula, the number of simultaneous calls to be supported, is the number of circuits required for the known amount of traffic, as calculated from the Erlang tables.
Note | If 100 percent of calls must go through, Erlang tables are not required; instead, the maximum number of simultaneous calls required should be used. |
The second component of this formula, the bandwidth required for one call, depends on the codec used and whether cRTP and VAD are used. Earlier in this chapter, the section “Voice Bandwidth Requirements,” including Table 8-6, illustrated some bandwidth calculations.
Caution | Including VAD in bandwidth calculations can result in insufficient bandwidth being provisioned if the calls do not include as much silence as assumed and when features such as music on hold are used. |
As an example of calculating the trunk capacity, assume that G.729 compression is used over a PPP connection at 50 pps and cRTP is used. From Table 8-6, each call uses 11 kbps. If five simultaneous calls are to be supported, 5 * 11 = 55 kbps is required for the voice calls.
Note | The bandwidth for other traffic that will be on the link must also be accounted for. |
As another example, based on the Erlang tables, ten circuits are required between two locations to satisfy user demands. VoIP over PPP is used on the link. The G.729 codec, using 50 samples per second, is used. cRTP is not used.
The per-call bandwidth information in Table 8-6 indicates that one voice call without header compression requires 26 kbps of bandwidth. Therefore, 10 * 26 = 260 kbps of bandwidth is required between the two locations, in each direction, to carry ten simultaneous voice calls.
Cisco IP Communications Return on Investment Calculator
The Cisco IP Communications (IPC) Return on Investment (ROI) calculator can be useful for analyzing IP telephony requirements and estimating the cost savings a customer will experience when migrating to IP telephony. The IPC ROI calculator is available at http://www.cisco.com/web/partners/sell/technology/ipc/ipc_calculator.html.
Note | You must have a Cisco partner account to access this tool. |
Summary
In this chapter, you learned about voice design principles, with a focus on the following topics:
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Analog and digital signaling, including the process to convert between the two
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The features of and similarities and differences between PBXs and PSTN switches
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The connections and signaling between the various devices in a traditional telephony network
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PSTN numbering plans and various PSTN services
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The H.323 standard for packet-based audio, video, and data communications across IP-based networks, including H.323 components
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The concepts of VoIP and IP telephony, including the components and sample design scenarios
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Protocols used to transport all control (signaling) and voice conversation traffic
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Voice quality issues, including delay, jitter, packet loss, and echo
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Voice coding and compression standards
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Bandwidth considerations and requirements for integrated networks
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QoS mechanisms for voice
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Voice traffic engineering to select the correct number of lines and the proper types of service, including the use of Erlang tables to calculate trunk capacity for voice calls.
Note | This chapter introduced voice design principles; additional resources, such as voice-related Cisco Press books and documents on http://www.cisco.com/, are required to successfully integrate voice services into a network. |
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