Pacific IT Solutions Case Study - SIP Trunking - VoIP Technologies - PBX Solution - IT Assessment Answer

Assessment Task:

Pacific IT Solutions Case Study IT Assessment Answer

About this assessment:

These assessment tasks provide an opportunity to demonstrate the competencies covered in the VoIP - Unified Communications subject.

•    You are allowed to refer to your textbooks, notes and the Internet during the Assessment.
•    The documentation and research work must be entirely your own.
•    By commencing this assessment you confirm that you have read and agree to abide by the ACIT Academic Honesty Policy

Successful completion of this assessment contributes towards attaining competency in the following:
ICTTEN512    Design and implement an enterprise voice over internet protocol and a unified communications network
ICTNWK610    Design and build integrated VoIP networks
ICTPMG611    Prepare a detailed design brief
ICTTEN611    Produce an ICT network architecture design
ICTNWK529    Install and manage complex ICT networks

 

There are 3 parts to this assessment

1.    Design and planning
2.    Deployment
3.    Question and answer

These assessment tasks provide an opportunity for you to demonstrate the competencies required to design, plan and deploy unified communications solutions.

 

Part 1 - Design and planning

This part requires you to prepare an architectural design of the unified communications solution and a deployment plan.

Scenario

You are employed by Pacific IT Solutions, an IT services company providing systems integration and consultancy services.  You have been appointed as an IT consultant for Online Travel Services (OTS) which is a client of Pacific IT Solutions. 

Background

OTS is a national organisation within Australia with 7 retail and sales offices in Australian capital cities and a central office and Data Centre on the Gold Coast. All the sites are connected to the central office via VPN.  In total OTS employs 120 people across the organisation with the majority of the staff provided with remote access through either a dial-up or a virtual private network (VPN) connection.

Business Profile

Global Travel Pty Ltd (The Company) is a leading provider of Travel services in the Asia Pacific region with headquarters in Singapore. The Company is the holding group for five smaller companies: OTS, Asia Hotel Services, APAC Flight Brokers, Asia Adventure Holidays and Happy Insurance. All of the companies owned by The Company focus on their individual target markets and employ several hundred staff. The Company is currently trying to reduce its costs and decrease the time it takes to break into new markets. OTS is a wholly-owned subsidiary of The Company.

The business was set up in 2001 by the General Manager (Jack Slasher) and run from a small office on the Gold Coast until 2004 when the business was bought by Global Travel Pty Ltd and expanded to have offices nationally. This offered the ability for OTS to expand from the online travel services to offer personalized service to clients through the regional offices.  OTS targets the provision of specialist travel packages to high-value clients, advertising the packages it offers through the website. The business is currently highly successful, with a turnover in the last financial year of over $100 million and a profit of $6 million.

Current technology

OTS does not currently have any unified communications technologies. The company uses a legacy PBX and all calls are made using the PSTN.  
All servers are deployed in a VMWare virtualized environment.  They have enough spare capacity in the VMWare virtualized environment to deploy any additional virtual servers they may need. All servers are attached to a Gigabit network.

Requirements

OTS has obtained a feasibility report in relation to migrating their telecommunications systems to a solution that uses a relatively low-cost SIP trunking service provided by an ITSP.  They have decided to proceed with the migration as it will provide substantial savings on their telephone costs. They will start with a staged implementation at their Gold Coast office.  This means that they will replace their current PSTN lines with a SIP trunk but continue to use their existing PBX system for most of their telephone handsets as a cost-saving measure. Only 5 executives will get new IP phones.

You have been given the task of designing the solution. Your contact at OTS is the office manager, Jane Moss, who will also sign off on the project completion. Your design and implementation plan must provide for the following:

An IP telephony solution that will:

1.    Use SIP trunking
2.    Integrate with their existing PBX
3.    Provide flexibility to expand and link to other sites
4.    Provide call parking, call queuing, voice mail, ring groups, and an IVR service
5.    Achieve these outcomes with minimal cost

Task 1

Research the following VoIP technologies.
•    Asterisk
•    Shoretel
•    Cisco

Compare the technologies on the following criteria.  Create a table/matrix showing the comparison.
•    Cost
•    Scalability
•    Size of online community
•    Quality of support documentation
•    Ability to integrate with legacy PBX
•    Features - voice mail, IVR, etc.

Task 2

Write a description of the solution you envisage.  Assume that you have chosen Asterisk as the solution.  You should include:
1.    A description of the architecture - include VoIP, IP telephony, network, and firewall elements
2.    A thorough explanation of how your solution will work - the exact process of how calls are made, received, interfaced with the PSTN
3.    An explanation of how your solution meets the business requirements
4.    A description of each component
5.    The rationale for the design decisions you made

Task 3
Compile an inventory of hardware and software components.  You must include the recommended hardware requirements for an Asterisk/FreePBX installation for an organization of this size.

Task 4
Prepare a work breakdown of your deployment plan.

Task 5
Write a post-deployment test plan.

Task 6
Create a drawing of the solution using Visio.  Include all physical and virtual components

Part 2 - Deployment

This part requires you to deploy an IP-private branch exchange (PBX) solution.  You shall work in pairs for the purpose of testing your solution.

You are provided with:
1.    An Asterisk/FreePBX distro ISO file
2.    A PC with Internet access
3.    Access to a VMWare VSphere infrastructure host in which you can create new VM's.
4.    Asterisk and FreePBX installation and configuration documentation
5.    IP addressing scheme, extension numbering scheme, and DID
6.    Head office trunk name and password

 

Scenario

OTS has implemented an Asterisk PBX at their head office.  A SIP trunk has been provisioned to this site and is now fully operational.   OTS now wishes to deploy IP telephony to their regional offices.  It is much more economical to have one large SIP trunk than many small SIP trunks and therefore the entire organization will use the SIP trunk provisioned to the head office.  The regional offices are connected to the head office via site to site VPN's. Some regional offices are connected to each other via site to site VPN's. OTS has decided that each regional office will have its own PBX. Calls to external numbers and other OTS offices that are not directly connected will be trunked via the head office.

You have been given the task of designing and deploying the solution for a regional office. Your design and implementation plan must provide for the following:

An IP-private branch exchange (PBX) solution that will:
1.    Provide extensions for the local office that can call each other
2.    Provide an efficient low bandwidth trunk to the head office and one other office
3.    Receive incoming calls from extensions in other offices
4.    Receive incoming calls on the assigned DID
5.    Make outgoing calls to other office extensions and external PSTN numbers
6.    Provide call parking, IVR, call queuing, voice mail and a ring group

Task 1
Install Asterisk/FreePBX IP-private branch exchange (PBX) server and perform initial configuration. Use the IP addressing scheme provided.  Record the IP address, username and password of the server.

Task 2
Create 2 extensions in accordance with your extension numbering scheme and configure soft phones or handsets for each of them. You may install and configure one softphone on the PC you are using and another on a mobile device that is connected to the campus network.  If you are doing this assessment remotely you will need to create 2 client VM's and configure a softphone on each of them. Test the extensions by to make sure they are working. Record the extension numbers.

Task 3
Create VoIP trunks to the head office PBX and one other PBX (this is your other team member's PBX).  The trunks must consume as little bandwidth as possible. The head office PBX IP address is 10.10.100.8. 

Task 4
Configure outbound call routing so that calls can be made to extensions in another PBX. (This is the PBX of the other student you are working with. If you do not have another student to work with you can configure outbound routing to the head office PBX for extensions beginning with 77). Verify that you can make and receive calls to and from the extensions in the other PBX.
Record the dial pattern that you are matching for this route and the name of the trunk through which calls will be forwarded.

Task 5
Configure an inbound route so that calls can be received on your DID. Verify that you can receive calls with 2 way audio made from the PSTN.  (you can do this by calling your DID from your mobile or any other phone on the PSTN). Record the test procedure and results.

Task 6
Configure an outbound route so that you can make calls to any external number.  Record the dial pattern that you are matching and the name of the trunk through which calls will be forwarded. How did you test that you could make calls to the PSTN?

Task 7
Configure call parking, IVR, call queuing, voice mail and a ring group.
How did you test that these functions are working?

Task 8
Complete the post-installation checklist in appendix 'A'.  Use today's date for acceptance and cutover dates.

Part 3 - Question and Answer

Start Wireshark Network Protocol Analyzer and open the capture file named 'Wireshark Capture VoIP Assessment'. This file is located in the student materials folder.
Examine line number 255.

1.    Describe what is happening here - what is this packet doing? 
2.    What is the application layer protocol?
3.    What are the IP addresses and names/numbers of the endpoints involved?
4.    What are the source and destination port numbers?

Question 2

Using the same capture file from the question above.
Examine line number 262.
1.    Describe what is happening here - what is this packet doing? 
2.    What is the application layer protocol?
3.    What are the IP addresses and names/numbers of the endpoints involved?
4.    What are the source and destination port numbers?

Question 3
Using the same capture file from the question above.
Examine line number 282.
1.    Describe what is happening here - what is this packet doing? 
2.    What is the application layer protocol?
3.    What are the IP addresses and names/numbers of the endpoints involved?
4.    What are the source and destination port numbers?

Question 4
Using the same capture file from the question above.
Examine line number 284.
1.    Describe what is happening here - what is this packet doing? 
2.    What is the application layer protocol?
3.    What are the IP addresses and names/numbers of the endpoints involved?
4.    What are the source and destination port numbers?

Question 5
Using the same capture file from the question above.
Examine line number 286.
1.    Describe what is happening here - what is this packet doing? 
2.    What is the application layer protocol?
3.    What are the IP addresses and names/numbers of the endpoints involved?
4.    What are the source and destination port numbers?

Question 6
Using the same capture file from the question above.
Examine line number 479.
1.    Describe what is happening here - what is this packet doing? 
2.    What is the application layer protocol?
3.    What are the IP addresses and names/numbers of the endpoints involved?
4.    What are the source and destination port numbers?

Question 7
Using the table below, fill in the missing information for Bandwidth (Incl. Overhead), Bandwidth for 15 Concurrent Calls, and Quality.

Question 8
Describe the impact that the choice of the codec will have on network performance.

Question 9
Compare H.264 and H.265.  Describe what these codecs do. Select which one is better and explain why.

Question 10
What is the purpose of the ITU-T E.164-numbering scheme?

Question 11
What is the purpose of ENUM?

Question 12
Provide an example configuration for an ENUM lookup in an Asterisk dialplan.

Question 13
Describe 3 potential benefits that could be provided via the IMS IP-Multimedia Subsystem standards

Question 14
Describe possible network congestion solutions that would enable quality of service (QoS) requirements to be met during periods of network congestion.  

Question 15
Make a copy of the text file named 'ooh323.conf'. This file is located in the student materials folder.  Edit this file to configure an Asterisk PBX to act as a H.323 gateway and connect with a H.323 gatekeeper at 10.10.100.100.

Question 16
Write an Asterisk dial plan to facilitate emergency calls from VoiP handsets in Australia. 

Question 17
You need to allow the SIP protocol through your firewall. What port number would you open?

Question 18
Read the following post about a PBX security breach. http://forums.whirlpool.net.au/archive/1740551
What measures could be taken in Asterisk to prevent this from happening? Describe 3 measures that could be taken.

You are required to submit the following as evidence for this assessment:
1.    Written tasks should be completed on a word processor
2.    The edited 'ooh323.conf' file must be submitted
3.    You must click the submit button

 

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