Author: T. Y. Fam
Research Assignment

1. Introduction

1.1 General Information
Bloggs Information System wants to expand their product range and its presence in the market; they need to build a new call centre to handle all the sales-related calls. However due to the limited floor space in the company’s main office building, they will have to build the new call centre in a different building. The new building will be located on a leased warehouse which is 3 kilometres away from the Company’s main office building. By accident, the warehouse is located within line of sight to the Company’s main building. Geographically, the main office building has a direct sight on the roof of the warehouse if a two-metre mast is installed on either one of the building’s roof.

1.2 Specifications of Call Centre
Call centre is a growing business that large business organisations are looking on them as a main communication method with their customers. Call centre may help company to improve efficiency on their services. However it is a challenging task to build a call centre, there are many different areas of specification that need to be achieved in order to provide a good and efficient customer service call centre. The greatest challenge would be to provide accurate information to customer in a given time slot. Different tool may need to be implemented in order to gather and record the right solutions for further uses. Customer usually expects an instant answer, call centre operators need to be well-trained to handle any kind of situations [3].

Apart from the accuracy of information that needs to be provided, the system communications of the call centre need to be well designed as well. There are many design problems that need to be solved in order to develop a flawless call centre.

The calls centre is located line of sight from a 3 kilometres distance from the Company’s main building. Therefore it is possible to use wireless communication between the two buildings. Call centre will be equipped with personal computers, all computers will be using VoIP communications and capable for user database inquiries. Communication between database and personal computer will be telnet based, thus it will not create significant data load on the networks. User database traffic will be no more than 1 Megabit per second at most in a high peak situation. The complete design will be focused on the call centre performance, to create an efficient call centre. User workloads will be calculated in order to find out how many operators will be required in peak and off-peak time.

The advantages of VoIP are definitely cost saving. In some cases VoIP is consider as free phone calls. VoIP service providers usually offer call charges which is significantly lower than normal PSTN phone calls. It is easier for a company to manage the networks as only one type of network traffic need to be handling instead of separated by data and voice networks. In terms of scalability, VoIP is very easy to add or remove individual node within the networks.

2. Design Solutions

2.1 Erlang Formulas
To calculate the traffic measurement in the call centre system, Erlang formulas will be used in the calculations. The calculation steps will try to sort out the number of agents required during peak or off-peak daily working hours. The calculations will be based on each working hour. In order to calculate the estimated agents required per hour, the following information have to be gathered:

• Number of calls received per hour, this will be calculated by Erlang formulas.

• Number of operators, 100 of operators will be operated in this scenario.

• Average call durations

• Average allowed wait time of each calls, also known as the average wrap up time where the unavailable time for an operator after finish a call and before answering the next incoming call.

Based on the Erlang formulas, an average of 515 calls can be handle by 100 operators and with an average waiting time per call of 2 seconds, 90% of calls to require no more than 5 seconds of waiting time, an average holding time of 10 minutes. Therefore, in a peak hour condition, an average of 500 to 600 of calls can be handled by 100 operators.

2.2 VoIP Traffic
The amount of bandwidth carried over a network includes the calculations of Codec, IP header, transmission medium and silence suppression. Codec usually declare the total amount of voice data per millisecond (ms). IP header usually has fixed packets which is 40 octets per packet. Transmission medium such as TCP/IP will add its own communication headers which need to be considered as well. Some codec has the silence suppression technology which can reduce the total bandwidth required by 50%. It reduces the data when it detects silence situations [5].

The use of codec is mainly to convert conversation waveform into digital form. 20 ms is a common time to generate a frame of data. Different codec may have different conversion time such as codec G.723.1 required 30 ms while codec G.729a only need 10 ms to do the job. Size of a frame is calculated by the number of bits produced per second and the transmission period. For example, Codec G.711 generating at rate of 20 ms, which generates 50 frames per second will transmits 64,000 bps. Each frame is equal to 64,000 divide by 50 frames which is 1,280 bits per frame transmitted.

IP header is the combinations of IP, UDP and RTP packets. The packets generated by codec are wrapped by IP header. Figure 1.0 below shows an example of IP header.

Figure 1.0 IP Header and Voice Data [5].

RTP contain 12 octets which control the timing of orders of the frame. UDP is around 8 octets which routes the data to the correct port of destination. IP function is to deliver the data to the destination host, sums up of 20 octets by it selves. There will be 40 octets need to be added into the codec per 20 ms which is 160 + 40 = 200 octets equivalent of 1600 bits being sent 50 times per second, which is 80 kbps.

TCP/IP Ethernet packet starts with 8 octets followed by a header of 14 octets which uses to define the source and destination of MAC address followed by a 4 octets CRC. Packets gap will be separated by 12 octets which results an additional total of 8 + 14 + 4 + 12 = 38 octets to be included in the total codec bandwidth.

Consider the new centre will be operates by 100 operators. Therefore each VoIP Codec G.711 of (200 octets + 38 octets) * 50 = 95.2 kbps multiply by 100 operators is equal to 9.52 Mbps. The call centre will be using a total of 9.52 Mbps bandwidth during peak hours. 9.52 Mbps will only cover the VoIP bandwidth and will not cover any others bandwidth such as database access or internet surfing.

Other types of codec also have been calculated at 100 operators during peak time and shows in figure 2.0 below:

Figure 2.0 Table of Codec comparisons [2]
2.3 Router Capacity
To get the best sound available, the VoIP G.711 codec will be choosing for the call centre. 9.52 Mbps of VoIP usage will be the peak time where 100 lines operated at the same time. The given bandwidth for database access during is 1 Mbps. So the total bandwidth usage for the call centre is the summation of 9.52 Mbps plus 1 Mbps, which sum up a total of 10.52 Mbps. The required router capacity will be 10.52 Mbps at least.

2.4 Suitable Routers
NetVanta 7100 from ADTRAN

Most of the enterprise level routers support from 100 to 1000 of operators, which is overwhelming for a 100 operators only call centre. NetVanta 7100 routers support up to 50 operators. Therefore, by using two NetVanta 7100 routers, 100 operators can be accommodated without any problems. This router coverage both IP telephony and data networking which is suitable for call centre to handle calls and database queries. QoS is also included in the system in order to maintain voice quality [6].

Figure 3.0 Picture of NetVanta 7100

This single chassis will be able to do all the required work such as IP PBX, gateway, voicemail, full-featured IP router, firewall, VPN, 24-port Ethernet switch with two Gigabit uplinks and two expansion slots for future upgrades. The installations for this router is simple because its reduced the amount of extra hardware required to operate [6].

Cisco 3825 Integrated Services Router

This router support up to 168 IP phones, which is suitable for the call centre to operates 100 calls simultaneous. The Cisco 3825 provides concurrent services such as security and voice. Its also support different modules to best suite operation needs.

In term of security features, Cisco 3825 provides on-board encryption and support up to 2000 VPN tunnels. This router supports both analogue and digital voice call support and with optional of voice mail support as well.

Figure 4.0 Picture of Cisco 3825

2.5 Microwave Link
Wireless link is often the solution to eliminate the required of wired, due to long distance required long cable work which is troublesome. However, many circumstances need to be applied in order to create a successful microwave link. Distance is the main are to determine the required power budget for the link based on the frequency strength gains from antenna, transmitter and receiver.

The connection between the office main building and the call centre will be using a microwave link connection. Microwave receiver and transmitter will be installed on both roofs of the two buildings. Even the frequency is enough for the distance of 3 Kilometers, there are many other propagation losses while transmitting the date. Latency will be an issue when transmitting wirelessly for a long distance. For example, in a worse case condition, a rain rate of 150 mm/hr and fog density of 200 g/m3 may be occurring. These signal lose need to be calculated and minus the total output frequency. If the result is greater than the receiver power, then the wireless network will be working properly in a bad condition.

In many situations, packet may be lost due to interferences, two widely used techniques are “discard and resend” and “forward error control” model. In “discard and resend” model, corrupted packet will be resend with a new packet. While “forward error control” model transmitting redundant packets so each data can be reconstitute corrupted bits themselves [8].

Wireless link basically is a link of antenna, receiver and transmitter. They are based on the required frequency and drop rates of the communication link. While transmitting, the transmitter will modulate radio-frequency power through an antenna; the power will then pass through an open environment and reach the receiver antenna. The receiver will then demodulate the frequency data. Frequency / wavelength link is the best choice for the scenario. 3 Kilometers is still considering close and may only caused cause very low data packet loss [8].

Suitable hardware can be found in the tabular information page. Choice of antenna, receiver and transmitter are based on the frequency around 5 to 10 GHz. This is suitable for the call centre traffic usage.

References and Bibliography

[1] T. Howard (2006), “Call Centre Design”, CRM Today, http://www.crm2day.com/library/EpFEkuAlpFBgOQuVcz.php

[2] Erlang (2006), “Call Centre Staffing and Trunk Design”, Westbay Engineers Limited, http://www.erlang.com/calldesign.html

[3] Step Two (2002) “Knowledge Management for Call Centres”, Step Two Designs Pty Ltd, pp.1-7, http://www.steptwo.com.au/ papers/kmc_callcentre/pdf/KMC_CallCentre.pdf

[4] (2004), “VoIP Terms and Definitions”, Patton Electronics Company, http://www.patton.com/manuals/VoIP_Glossary.pdf

[5] (2005) “VoIP bandwidth Calculations”, Newports, http://www.newport-networks.com/ whitepapers/voip-bandwidth1.html

[6] Adtran (2006) “The NetVanta 7000 Series”, https://www.adtran.com/adtranpx/Doc/0/P8DRSVBSTIUKH6 BDM6LVV91473/EN698c%20NetVanta%207100%20ICP.pdf

[7] Cisco (2006), “Deliver Integrated Services to Branch Offices and SMB“, http://www.cisco.com/en/US/products/ps5855/index.html

[8] Monash (2006) “Error rate and impact; RF Propagation in Wireless and Cellular Networks”, http://www.csse.monash.edu.au/courseware/ cse4884/SLIDES/CSE-4884-13-14.pdf

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