Final Year report on EPABX system
Short Description
Final year report for EPABX system...
Description
Project SPBX
Final Report
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Final Report
VoIP Soft PBX
Project Code: SPBX
Project Advisor: Aftab Alam
Project Team: Umair Ashraf 03-1853 Khadija Akram 04-0080 Imran Bashir 02-1658
(Team Lead)
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Acknowledgements
We would like to thank our respected internal faculty advisor Mr. Aftab Alam for his continuous support and guidance throughout the project. We also appreciate the confidence shown by him on our team.
We would also love to acknowledge the support, guidance and help by our external advisor Mr. Asad Gill. He helped and taught us the basic ideas of voice over IP. He also helped us in formulizing the ideas of project into reality.
We also acknowledge the efforts and contributions of our project manager Mr.Yasir Aleem towards this project. In spite of his short stay with us, he guided us very efficiently and his advices helped us to produce more fruitful and better results.
Team Project PBX 12-6-08
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Document Information Category
Information
Customer Project Document Document Version Identifier Status
FAST-NU Soft PBX Final Report 1.0 SPBX Draft Umair Ashraf (Team Lead)
Author(s)
Khadija Akram
Approver(s) Issue Date Distribution
Imran Bashir Aftab Alam June 10, 2008 Advisor & Library
Definition of Terms, Acronyms and Abbreviations Term
Description
RS
Requirements Specifications
PBX
Private Branch Exchange
SIP
Session Initiation Protocol
RTP
Real Time Protocol
VOIP
Voice over IP
IETF
Internet engineering Task Force
MCU
Multi control Unit
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Table of Contents 1. 2. 3. 4. 4.1 4.2 4.3 4.4 4.5 4.6 4.7 4.8 4.9 4.10 4.11 4.12 4.13 4.14 4.15 4.16 5. 5.1 5.2 5.3 5.4 5.5 5.6 6. 7. 8. 8.1 8.2 8.2.2 8.2.3 8.2.4 8.2.5 8.2.6 8.2.7 8.2.8 8.2.9 8.2.10 8.2.11
Introduction ........................................................................................................... 1 Goals and objectives ............................................................................................. 3 Scope of the project ............................................................................................... 3 Functional Requirements ..................................................................................... 4 Registering a user .................................................................................................... 4 Dialing and placing Call ......................................................................................... 4 Accepting (call pick up) /rejecting a call ................................................................ 4 Terminating the Session .......................................................................................... 4 Voice Conferencing ................................................................................................ 5 Missed Call Alert .................................................................................................... 5 Call Hold ................................................................................................................. 5 Call Detail Reporting (CDR) .................................................................................. 5 Caller identification ................................................................................................ 5 Dial Plan.................................................................................................................. 6 Speed dialing ........................................................................................................... 6 Administrative and management services .............................................................. 6 Phone book service ................................................................................................. 6 Call Recording ........................................................................................................ 6 Call Transfer ........................................................................................................... 6 Call forwarding ....................................................................................................... 6 Non Functional Requirements ............................................................................ 7 Performance and Reliability ................................................................................... 7 Robustness .............................................................................................................. 7 Security ................................................................................................................... 7 Standards compliance ............................................................................................. 8 Usability .................................................................................................................. 8 Portability................................................................................................................ 8 Hardware Requirements ...................................................................................... 9 Software Requirements ........................................................................................ 9 Detailed Design and Architecture...................................................................... 10 Use Case Diagram ............................................................................................... 10 Use Case Description .......................................................................................... 11 Use Case Receiving/rejecting a call .................................................................. 13 Use Case Missed calls ....................................................................................... 14 Use Case Register User ..................................................................................... 15 Use Case Generate Report ................................................................................ 16 Use Case Call Transfer ..................................................................................... 17 Use Case call conferencing ............................................................................... 18 Use Case Call Recording ................................................................................. 19 Use Case Phone book Service.......................................................................... 20 Use Case Call Hold .......................................................................................... 21 Use Case Call Unhold ...................................................................................... 22 iv FAST National University of Computer and Emerging Sciences Lahore
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8.2.12 8.2.13 8.2.14 8.2.15 8.2.16 8.2.17 8.3 8.3.1 8.3.2 8.4 8.4.1 8.4.2 8.4.3 8.4.4 8.4.5 8.4.6 8.4.7 8.4.8 8.4.9 8.4.11 8.4.12 8.4.13 8.5 8.6 8.6.1 8.6.2 8.6.3 9. 9.1 9.2 9.3 9.4 9.5 9.6 9.7 9.8 9.9 9.9.1 9.9.2 9.9.3 9.10 9.10.1 9.10.2 9.11
Use Case terminating the call ........................................................................... 23 Use Case Call forwarding ................................................................................. 24 Use Add caller id/dial plan .............................................................................. 25 Use Case Change caller id/dial plan ................................................................ 26 Use Case Delete Caller id /dial plan ................................................................. 27 Use Case Change hold on music play list ........................................................ 28 Class Diagrams .................................................................................................... 30 Soft PBX ........................................................................................................... 30 Soft Phone ......................................................................................................... 31 Sequence Diagrams ............................................................................................. 32 Bye Call ............................................................................................................ 32 Cancel Call ........................................................................................................ 33 Invite Call.......................................................................................................... 34 MOH Sequence Diagram .................................................................................. 35 Receive Call ...................................................................................................... 36 Register ............................................................................................................. 37 Reject Call ......................................................................................................... 38 Callee RTP Process ........................................................................................... 39 Caller RTP Process Diagram ............................................................................ 40 Receive Call Sequence Diagram ....................................................................... 42 Conference Call Sequence Diagram ................................................................. 43 Music on Hold Call Sequence Diagram............................................................ 43 Context Diagram ................................................................................................. 44 Data Flow Diagrams ........................................................................................... 45 Level 0 Data Flow Diagram .............................................................................. 45 Level 1 Data Flow Diagram .............................................................................. 46 Level 2 Data Flow Diagram .............................................................................. 47 Test Plan .............................................................................................................. 49 Purpose.................................................................................................................. 49 Scope ..................................................................................................................... 49 Source Documents ................................................................................................ 49 Derived Documents .............................................................................................. 49 Assumptions/Limitations ...................................................................................... 49 Modules to Be Tested ........................................................................................... 49 Features not to be tested ........................................................................................ 50 Strategy ................................................................................................................. 50 Functional Testing ................................................................................................ 50 Approach ........................................................................................................... 51 Entrance Criteria ............................................................................................... 51 Exit Criteria ....................................................................................................... 51 Integration Testing ................................................................................................ 51 Approach ........................................................................................................... 52 Entrance Criteria ............................................................................................... 52 User Interface Testing ........................................................................................... 52 v FAST National University of Computer and Emerging Sciences Lahore
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9.11.1 9.11.2 9.11.3 9.12 9.12.1 9.12.2 9.12.3 9.13 9.13.1 9.13.2 9.13.3 9.14 9.15 9.16 9.17 9.18 9.19 10. 10.1 10.2 10.3 10.4 10.5 10.6 10.7 10.8 10.9 10.10 10.11 10.12 10.13 10.14 10.15 11. 12.
Approach ........................................................................................................... 52 Entrance Criteria ............................................................................................... 52 Exit Criteria ....................................................................................................... 52 Sanity Testing ....................................................................................................... 53 Approach ........................................................................................................... 53 Entrance Criteria ............................................................................................... 53 Exit Criteria ....................................................................................................... 53 Data Verification ................................................................................................... 53 Approach ........................................................................................................... 53 Entrance Criteria ............................................................................................... 53 Exit Criteria ....................................................................................................... 54 Defect Tracking .................................................................................................... 55 Guidelines for Assigning severity to a defect ....................................................... 57 Test Suspension and Resumption Criteria ............................................................ 58 Schedule Overview ............................................................................................... 59 Testing Resources ................................................................................................. 59 Summary ............................................................................................................... 60 TEST CASES ...................................................................................................... 61 Dial and Place Call................................................................................................ 61 Transfer Call ......................................................................................................... 65 Call Conference .................................................................................................... 67 Phone Book ........................................................................................................... 69 Soft Phone Configuration ..................................................................................... 74 PBX Configuration ............................................................................................... 79 PBX Admin Login ................................................................................................ 81 Create Dial Plan .................................................................................................... 83 CDR (Call Detail Reporting) ................................................................................ 85 Register User ..................................................................................................... 87 Receiving/rejecting a call .................................................................................. 91 Speed Dial ......................................................................................................... 94 Missed Calls ...................................................................................................... 96 Terminating the call .......................................................................................... 98 Call Hold ......................................................................................................... 100 Appendix ............................................................................................................ 102 References .......................................................................................................... 104
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1.
Introduction
VoIP is simply the transport of voice traffic by using the Internet Protocol (IP). It is a technology that allows you to make voice calls over a broadband internet connection instead of a regular phone line. In VoIP your voice is converted into a digital signal that travels over the internet. If you are calling a regular phone number (i.e. on a PSTN), the signal is converted to a regular telephone signal. [Reference 1]
Why should we use VoIP? Traditional telephony carriers use circuit switching for carrying voice traffic. Circuit switching was designed for voice from the outset; hence it carries voice in an efficient manner. However it is an expensive solution. Nowadays people want to talk much more on phone, but they also want to communicate in a myriad of other ways – through e-mail, instant messaging, video, the World Wide Web, etc. Circuit switching is not suitable for this new world of multimedia communication. IP is an attractive choice for voice transport for many reasons, including the following:-
Lower equipment cost Integration of voice and data applications Lower bandwidth requirements Widespread availability of IP
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Lower Equipment Cost The IP world is different from the monolithic systems of mainframe computers and circuit-switching technology. IP systems tend to use distributed client-server architecture rather than large monolithic systems, which means that starting small and growing as demand dictates is easier. IP architectures and standards are more open and flexible plus they are competition friendly, than telephony standards, enabling the implementation of unique features so that a provider can offer new features more quickly. Hence, the range of choices is large; the equipment cost is drastically lower than that of circuit switching products; and the pace of development is incredibly fast.
Voice/Data Integration and Advanced Services IP is the standard for data transactions- everything from e-mail to web browsing to ecommerce. When we combine these capabilities with voice transport on a single network, we can easily imagine advanced features that can be based on the integration of the two. Lower Bandwidth Requirements Circuit-Switched telephone networks transport voice at a rate of 64 Kbps. Typical human speech has a bandwidth of 4000 Hz, when we would digitize this voice a telephone system would take 8000 samples per second. In VoIP we can use coding schemes which enable speech to be transported between 6 Kbps to 32 Kbps. Therefore, VoIP offers significant advantages over circuit-switching from a bandwidth point of view. However we would be using G.711 for this purpose which is also at a rate of 64 Kbps.
Widespread availability of IP IP is practically everywhere. Every personal computer produced today supports IP. IP is used in corporate LANs and WANs, dial-up internet access etc. As a result there is widespread availability of IP expertise and numerous application-development companies. This factor alone makes IP a suitable choice for transporting voice. [Reference 2]. 2 FAST National University of Computer and Emerging Sciences Lahore
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2.
Goals and objectives
Our primary goal is to gain a thorough understanding of the VoIP. We would be learning and implementing RTP (Real Time Protocol) and SIP (Session Initiation Protocol).We would be learning the different mechanisms and flows involved in the transference of voice over the internet. We would gain complete insight of the complete mechanism of communication between the Soft Phones and the working of the Soft PBX.
3.
Scope of the project
Vision and Scope The Virtual PBX (Private Branch Exchange) Numerous IP based PBX solutions are in place and more are being deployed daily. The idea of an IP-based PBX is useful. Firstly, this system integrates the corporate telephone system with the corporate computer network, removing the need for two separate networks. A new office is wired for voice communication, as well. The PBX itself becomes just another server or group of servers in the corporate LAN, which helps to facilitate voice/data integration. We would be implementing a soft PBX that is going to be a server that performs callrouting functions, replacing the traditional legacy PBX or key system. Our PBX would allow a number of attached soft phones to make calls to one another and to connect to other telephone services. The basic software would include many features available in proprietary PBX systems: voice mail, conference calling, interactive voice response, and automatic call distribution, just to name a few. Our system would also help the user in building the dial plan for the network. We would be using the Session Initiation Protocol [See Appendix for more details] as our VoIP protocol. Our PBX would be acting as both the registrar and as a gateway between the soft phones. 3 FAST National University of Computer and Emerging Sciences Lahore
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4.
Functional Requirements
4.1
Registering a user
The system shall allow a user (caller or callee) to register it to the PBX by its IP Address and caller ID through a soft phone. The user will set a specific IP address, caller id, and password and caller name in the configuration of the soft phone. After applying these configurations, the soft phone will send a request to the soft PBX and the PBX in response will register the user.
4.2
Dialing and placing Call
The system shall allow users to dial and place a call to each other using Soft phones through a PBX over the Network (internet or intranet) The caller dials the number and the number will be displayed on the screen and then the user presses the call button and the call will be placed to the specific callee and the session information will be displayed to the caller on the screen of his/her soft phone.
4.3
Accepting (call pick up) /rejecting a call
The system shall allow the user to receive or reject a call of the caller using Soft Phone. For the incoming call, the callee will receive a notification of the incoming call through the ringing tone and information will also be displayed on the screen. The callee can either reject or receive the call. Incase he receives the call a communication session will be established between the caller and the callee. Incase the callee rejects the call the information of the missed call will be updated in the missed call log.
4.4
Terminating the Session
The system shall allow the user to terminate the call at any time. If a call is already established between the caller and the callee, both of them shall be able to terminate the session at any time. In this case the session will be terminated and both the caller and callee will get nonfiction of call termination on the screen. 4 FAST National University of Computer and Emerging Sciences Lahore
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4.5
Voice Conferencing
The System shall allow users to have voice conferencing service amongst multiple users at a time. There will be a specific conferencing no that will be set in the multi conferencing unit. The caller shall be able to join the conferencing session by dialing that conferencing no. In this case a request will be sent to the multi control unit by the PBX to join the conferencing session. 4.6
Missed Call Alert
The System shall support missed call service and provide the information of the missed calls to the user. 4.7
Call Hold
The system shall provide the station user, to hold a call in progress, the ability to dial either an appropriate hold code, or to depress a feature button which would place the call in progress on system hold, and allow the station user to: (1) originate another call or, (2) use any other features provided by the system. 4.8
Call Detail Reporting (CDR)
The PBX shall be equipped to capture Call Detail information. The information to be captured shall as a minimum provide: Date of Call (Month and Day) Calling Station Number Called Number (all depressed digits) Time (Time Call Was Placed) Duration of Call (Minutes and Seconds) 4.9
Caller identification
The System shall allow the user to get the information of the caller person like his/her caller id and the name. 5 FAST National University of Computer and Emerging Sciences Lahore
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All the information related to caller person will be displayed on the screen for the callee. 4.10
Dial Plan
The System shall provide complete dial plan option to the users. The administrator can generate caller ids in different formats like for a particular department the caller id shall start from the number 3 and the caller id number could be in different lengths according to the plan generated by the administrator. 4.11
Speed dialing
The system shall support the services of the speed dialing through a PBX. 4.12 Administrative and management services The system shall provide the administrative and management services of the PBX like changing dial plans, managing users, managing passwords and creating call Detail Reporting services. 4.13
Phone book service
The system shall support and provide phone book service to the users. 4.14
Call Recording
The system shall allow the user to record the call of the conversation. 4.15
Call Transfer
The system shall allow the user to transfer his/her call to another user at a different location. 4.16
Call forwarding
The System shall allow the user to forward his/her call to another location.
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5.
Non Functional Requirements
5.1
Performance and Reliability
System must be scalable up to 50 devices per PBX. The system must ensure that sending and receiving packets are not discarded, and a mechanism must be adopted so that packet loss and retransmission should not occur due to algorithm used in the application otherwise, voice quality or service disruptions might occur. The jitter buffer (see appendix for details) configuration must be implemented in the software to avoid packet delay which should not be more than 100 millisecond between the two consecutive packet transmissions otherwise quality of voice may drop. The system must ensure to utilize as low bandwidth of the network as possible. In order to prevent high traffic over the internet by the application some compression codec‟s like G.729 (see appendix for more details) shall be supported by the system.
5.2
Robustness
The system should support proper exception handling like incase of unavailability of Network.
5.3
Security
The system shall provide complete security and privacy to the users and no other party shall be allowed to listen to the conversation between the two end users.
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5.4
Standards compliance
The system shall fulfill all the standards of the IEEE and IETF. The system shall support all standard protocols like SIP and RTP protocols. (See Appendix for details). 5.5
Usability
The system must be providing user-friendly interface to the end users. A dial pad and telephone like features must be provided with the interface so as to provide the end user a Complete Soft phone on his/her desktop.
The system shall provide an interface to the Administrator to maintain and configure the system. 5.6
Portability
The system shall work under Windows XP or windows 2000.
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6.
Hardware Requirements Pentium 4 system with at least 512 Mb of Ram Full duplex Sound Card A Conventional LAN based network or Intranet. 100 Mbytes of Disk space
7.
Software Requirements
The system shall be developed in rapidly growing and cutting edge technology of .Net and C sharp framework.
Microsoft Windows 2000 or Windows XP (Operating System)
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8.
Detailed Design and Architecture
8.1
Use Case Diagram
Soft PBX Phone Book
«uses» «extends»
Place phone Call
Place coference Call Hold Call Receive Call
Unhold Call
Terminate call
Record Call
Missed Call
Callee Call Deatail Report
Forward Call
End Conference Call
Accept Call
Caller
Reject Call
Change MCU no
Music Unhold
Change User
«uses»
Register user
«uses» Operator Remove User
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8.2
Use Case Description 8.2.1
Use case Dialing and placing a call
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8.2.2
Use Case Receiving/rejecting a call
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8.2.3
Use Case Missed calls
Use case Id 3: Missed Calls Actors: Feature: This will allow the user to see the missed calls on the soft phone. Use case Id: 3 Pre-condition: -
Scenarios Step#
Action
Software Reaction
1.
The use case starts when the callee wants to see the missed calls on his phone. User directs the soft phone to display missed calls by pressing misses call button.
The soft phone will display the missed calls on the display screen. The use case ends.
2.
Alternate Scenarios: None
Exceptions: None
Post Conditions Step#
Description
1-
Soft Phone will display the missed calls.
Use Case Cross referenced
None
Concurrency and Response Number of concurrent users The system will support maximum 50 users. Expected response time of the use case The response time will be 1 second.
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8.2.4
Use Case Register User
Use case Id 4: Register User Actors: Feature: This will allow the user to register itself to the PBX. Use case Id: 4 Pre-condition: The user must be authorized caller id to register to the system
Scenarios Step#
Action
The use case starts when the user wants to register itself to the system 2. The user presses the register a user button. 3The user enters the port no and the ip address of the user by pressing the continue button. Alternate-1 Alternate Scenarios: none
Software Reaction
1.
System displays a dialogue box to enter the ip address and the port no System will successfully register the user. Exception-1 The use case ends.
Exceptions: 1-The user did not enter the valid port no or ip address and repeats step 2.
Post Conditions Step#
Description
1-
System will successfully register the user.
Use Case Cross none referenced Concurrency and Response Number of concurrent users The system will support maximum 50 users. Expected response time of the use case The response time will be 1 second.
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8.2.5
Use Case Generate Report
Use case Id 5: Generate Report Actors: Feature: This will allow the operator to generate a report. Use case Id: 5 Pre-condition: The operator must be logged in.
Scenarios Step # 1.
2. 3-
Action The use case starts when the operator wants to generate the report of the calls. The operator presses the report generate button. The operator enters the duration.
Software Reaction
System displays a dialogue box to enter specific duration. System will successfully generate the CDR (call data Report). Exception-1 The use case ends.
Alternate Scenarios: none
Exceptions: 1-The operator did not enter the valid duration.
Post Conditions Step # 1-
Description System will successfully generate the report.
Use Case Cross none referenced Concurrency and Response Number of concurrent users The system will support maximum 1 user. Expected response time of the use case The response time will be 10 second.
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8.2.6
Use Case Call Transfer
Use case Id 6:
Call Transfer: Actors: < User> Feature: This will allow the user to transfer his/her call to another location. Use case Id: 6 Pre-condition: The user should be having the established communication session.
Scenarios Step # 1.
2.
3.
Action The use case starts when the user wants to forward his/her call to another location by pressing call transfer button. User presses the transfer button to transfer the call. User Enters the caller id of the person to whom the call is being transferred.
Software Reaction
The system will ask for the caller id where the call is being transferred. The system will transfer the call to another location and notifies the user that the call is being forwarded. (Exception-1) (Exception-2) The use case ends.
Alternate Scenarios: None Exceptions: 1- The network is busy or is not available and system will generate exception error. 2- The caller id does not exist and user repeats step3.
Post Conditions Step # 1-
Description System will transfer the call to another location and the caller will be idle.
Use Case Cross none referenced Concurrency and Response The system will support maximum 1 user. Expected response time of the use case The response time will be 1 second.
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8.2.7
Use Case call conferencing
Use case Id 7: Call Conferencing Actors: < User> Feature: This will allow the user to have call conferencing Use case Id: 7 Pre-condition: The user must be registered to the PBX
Scenarios Step # 1. 2.
Action 1-The use case starts when the user wants to have call conferencing 2-User dials the conferencing id ( MCU caller ID)
Software Reaction
System connects the user to the conferencing session. (Exceptions 1,2,3) The use case ends.
Alternate Scenarios: None
Exceptions: 1- The system generates error that the conferencing id is incorrect and the user repeats step 2 2- The system generates error that the network is unavailable. 3- The system generates error that the maximum no of users are already having the conferencing call over the same conferencing id.
Post Conditions : none Use Case Cross Includes referenced Concurrency and Response Number of concurrent users = 5 Expected response time of the user case 3 seconds
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8.2.8
Use Case Call Recording
Use case Id 8:
Call Recording Actors: < Caller,Callee> Feature: This will allow the user to record the call of conversation. Use case Id: 8 Pre-condition: The user should be having the established communication session.
Scenarios Step # 1. 2.
Action 1-The use case starts when the user wants to record a call of conversation. 2-User by presses the call record button.
Software Reaction
3-The soft phone will record the call of conversation, stores it in a file and notifies the user that the call of conversation is being recorded. (Exceptions-1) The use case ends.
Alternate Scenarios: none Exceptions: 1- System does not record the call of conversation.
Post Conditions : Step Description # 1Soft PBX will have the record of the call of conversation. Use Case Cross none referenced Concurrency and Response Number of concurrent users The system will support at max 1 recording at a time.. Expected response time of the use case The response time will be 5 seconds.
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8.2.9
Use Case Phone book Service
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8.2.10
Use Case Call Hold
Use case Id 10: Call Hold Actors: < User> Feature: This will allow the user to hold a call. Use case Id: 10 Pre-condition: The user should be having the established communication session.
Scenarios Step # 1. 2.
Action The use case starts when the user wants to hold a call. User presses the call hold button.
Software Reaction
System will hold the call for the user and play hold music. Execption-1 The use case ends.
Alternate Scenarios: None
Exceptions: none
Post Conditions : Step # 1-
Description System will hold the call for the user.
Use Case Cross None referenced Concurrency and Response Number of concurrent users 50 Expected response time of the use case The response time will be 2 seconds.
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8.2.11
Use Case Call Unhold
Use case Id 11: Call
Un Hold
Actors: < User> Feature: This will allow the user to Un hold the call. Use case Id: 11 Pre-condition: The call must be hold first to initiate this use case.
Scenarios Step # 1. 2.
Action The use case starts when the user wants to un hold the call and talk to another user User presses un hold button.
Software Reaction
System will again establish a communication session between the calle and the caller. Exception-1 The use case ends.
Alternate Scenarios: None
Exceptions: 1- The system could establish the session with the callee as the network is unavailable.
Post Conditions: The caller and callee will be having the communication session again after the un hold operation. Use Case Cross Includes < dialing and placing a call> referenced Concurrency and Response Number of concurrent users 50 Expected response time of the use case The response time will be 2 seconds.
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8.2.12
Use Case terminating the call
Use case Id 12: Terminating the call Actors: < User> Feature: This will allow the user to terminate the call. Use case Id: 12 Pre-condition: The user should be having the established communication session.
Scenarios Step # 1.
Action
Software Reaction
1-The use case starts when the user wants to terminate the call.
2. User presses the cancel button to terminate the call.
System will disconnect the call and display the user that the call is connected. The use case ends.
Alternate Scenarios: none
Exceptions: none
Post Conditions : Step # 1-
Description System will not maintain the communication channel anymore.
Use Case Cross none referenced Concurrency and Response Number of concurrent users 50 Expected response time of the use case The response time will be 1 seconds.
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8.2.13
Use Case Call forwarding
Use case Id 13:
Call forwarding Actors: < operator/administrator> Feature: This will allow the user to forward the call of a user to another location. Use case Id: 13 Pre-condition: The administrator must be logged in
Scenarios Step# 1.
2.
3.
Action The use case starts when the user wants to forward the calls to another caller id. User presses the call forward button to forward the call. User Enters the caller id and the id of the person to whom the call is being forwarded.
Software Reaction
The system will ask for the caller id and the id where the call is being forwarded of that caller id The system will forward and route the call to another location and notifies the user that the call is being forwarded. (Exception-1) (Exception-2) The use case ends.
Alternate Scenarios: None Exceptions: none
Post Conditions Step# 1-
Description System will forward the call to another location and the caller will be idle.
Use Case Cross none referenced Concurrency and Response Number of concurrent users The system will support maximum 1 user. Expected response time of the use case The response time will be 1 second.
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8.2.14
Use Add caller id/dial plan
Use case Id 14: Add caller id/dial plan Actors: Feature: This will allow the operator to add a caller id to the system. Use case Id: 14 Pre-condition: The operator must be logged in.
Scenarios Step # 1. 2. 3-
Action
Software Reaction
The use case starts when the operator wants to adds a caller id or dial plan The operator presses the add button The operator enters the caller id and the ip address of the user by pressing the continue button. Alternate-1
System displays a dialogue box to enter the caller id and the ip address System will successfully enters the new entry in the dial plan Exception-1 The use case ends.
Alternate Scenarios: none
Exceptions: 1-The user did not enter the valid caller id or ip address or the caller id and the ip address already exists.
Post Conditions Step # 1-
Description System will successfully register the user.
Use Case Cross none referenced Concurrency and Response Number of concurrent users The system will support maximum 1 user. Expected response time of the use case The response time will be 2 second.
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8.2.15
Use Case Change caller id/dial plan
Use case Id 15: change
caller id/dial plan Actors: Feature: This will allow the operator to change a caller id to the system. Use case Id: 15 Pre-condition: The operator must be logged in.
Scenarios Step#
Action
Software Reaction
1.
The use case starts when the operator wants to change a caller id or dial plan The operator presses the add button
System displays a dialogue box of all the registered caller ids and their ips
The operator selects caller id and the ip address of the user by pressing and enter the new values Alternate-1
System will successfully enters the new entry in the dial plan Exception-1 The use case ends.
2. 3-
Alternate Scenarios: none
Exceptions: 1-The user did not enter the valid caller id or ip address or the caller id or the ip address already exists.
Post Conditions Step#
Description
1-
System will successfully change the dial plan.
Use Case Cross none referenced Concurrency and Response Number of concurrent users The system will support maximum 1 user. Expected response time of the use case The response time will be 2 second.
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8.2.16
Use Case Delete Caller id /dial plan
Use case Id 16: delete
caller id/dial plan Actors: Feature: This will allow the operator to delete a caller id to the system. Use case Id: 16 Pre-condition: The operator must be logged in.
Scenarios Step # 1.
2. 3-
Action The use case starts when the operator wants to delete a caller id or dial plan The operator presses the add button The operator selects caller id and the ip address of the user and presses the delete button. The user confirms by clicking yes
4Alternate 1
Software Reaction
System displays a dialogue box of all the registered caller ids and their ips The system will display a confirmation dialog box System will successfully delete the entries Exception-1 The use case ends.
Alternate Scenarios: 1 The user presses the cancels button.
Exceptions: none
Post Conditions Step # 1-
Description System will successfully delete the dial plan.
Use Case Cross none referenced Concurrency and Response Number of concurrent users The system will support maximum 1 user. Expected response time of the use case The response time will be 2 second.
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8.2.17
Use Case Change hold on music play list
Use case Id 17: change
hold on music play list Actors: Feature: This will allow the operator to change the hold on music of the system. 17 Use case Id: The operator must be logged in. Pre-condition:
Scenarios Step#
Action
1.
The use case starts when the operator wants to change a music play list The operator presses the hold on music button The operator gives the path of new file Alternate-1
2. 3-
Software Reaction
System displays a dialogue box to the give the new file for hold on music System will successfully enters the new music file for hold on music Exception-1 The use case ends.
Alternate Scenarios: 1 The operator cancels the operation.
Exceptions: 1-The user did not enter valid path or file and system generates the error.
Post Conditions Step# 1-
Description System will successfully change the dial plan.
Use Case Cross referenced
none
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8.3
Class Diagrams
8.3.1 Soft PBX
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8.3.2 Soft Phone
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8.4
Sequence Diagrams
(Call Flow Diagrams) 8.4.1 Bye Call
Soft Phone (Caller)
PBX
Soft Phone (Callee)
SIP BYE BYE SIP 200 OK SIP 200 OK
Release
Release Complete
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8.4.2 Cancel Call
Soft Phone (Caller)
: Callee
PBX
Soft Phone (Callee)
Dial Callee Press Cancel Call Buttion
cancel Cancel SIP OK 200
SIP OK 200 Opration terminated
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8.4.3 Invite Call Soft Phone (Caller)
PBX
Soft Phone (Callee)
SIP Invite (callee #) SIP 100 Trying Invite Call Proceding Progress
SIP 180 Ringing
SIP 180 Ringing
Connect Connect ACK SIP 200 OK SIP ACK
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8.4.4 MOH Sequence Diagram
: Callee
Soft Phone (Caller)
Press HOLD Buttion
PBX
Request For MOH
MOH (Music on Hold)
Music_on()
SIP OK 200
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8.4.5 Receive Call
: Callee
Soft Phone (Callee)
PBX
Soft Phone (Caller)
Click on receive Buttion
Connection Esteblished On RX\TX Channel
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8.4.6 Register
: Callee
Soft Phone (Caller)
Set Caller ID and Password
PBX
Register SIP OK 200 Options SIP OK 200
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8.4.7 Reject Call
: Callee
Soft Phone (Callee)
Click On Reject Buttion
PBX
Soft Phone (Caller)
SIP BYE BYE SIP OK 200 SIP OK 200
Disconnect RX\TX Connection
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Basic Sequence Diagrams 8.4.8 Callee RTP Process
: Callee
RTP Processor
CSocket
JitterBuffernode
Audio Device
Audio codec
Start session
Creat Socket()
Start Streaming() GetPacket()
DecodePacket() Open Audio Device() Output Data()
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8.4.9 Caller RTP Process Diagram
: Caller
RTP Processor
Audio Device
Audio codec
CSocket
JitterBuffernode
StartSession InputAudiao() OpenDevice() close device
Encode Data() Creat Socket()
AddPacket()
CloseSocket()
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8.4.10
Dialed Call Sequence Diagram
: Callee
: softphoneapp
PBXengine
SIP Engin
: SIPMessageGenerator
: InviteStruct
: PacketHeader
CSocket
Dial Caller No
Place Call Dial Call()
Initiate()
Makeinvite()
generateMessage()
CreateInviteStruct()
ConstructPacket() CreateSocket() SendPacket()
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8.4.11
: Callee
Receive Call Sequence Diagram
Softphoneapp
Click on Receive Buttion
: SIP Engine
: PBXengine
CSocket
: SIP parser
: MAP
CallNode
: SIPcallState Machine
SIP Message Generator
Initiate
makeInvite() CreateSocket() Receive Packet() Parse Message() UpdateMap() UpdateNode()
GenerateState()
Generate Message() SendPacket()
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8.4.12
: Caller/callee
Conference Call Sequence Diagram
: softphoneapp
Click on Hold Option
PBXengine
Initiate
MCU
SIP Engin
MakeInvite() startConference Call()
8.4.13
: Caller/callee
Music on Hold Call Sequence Diagram
: softphoneapp
Click on Hold Option
PBXengine
Initiate
SIP Engin
MakeInvite()
MOH
PlayMusic()
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Context Diagram
UDP Connection
Soft PBX
Caller
Operator
SIP
Soft Phone
Receive Call, Reject, Call Hold Call, Transfer Call
SIP
Register user, Edit user, Generate CDR, Change Music
Soft Phone
Rx/Tx
Dial Call, Hold Call, Missed Call Terminate Call
8.5
Callee
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Data Flow Diagrams
8.6.1
Level 0 Data Flow Diagram
UDP Connection
Soft PBX
Register user, Edit user, Generate CDR, Change Music
SIP
Dial Call, Hold Call, Missed Call Terminate Call
Soft Phone
Rx/Tx
Caller
Operator
SIP
Soft Phone
Receive Call, Reject, Call Hold Call, Transfer Call
8.6
Callee
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Level 1 Data Flow Diagram
Multipoint Control Unit (MCU)
Rx/Tx
SIP
PBX Engine
Generate Add, Report Delete Change & Update Music & User Music No Request
Display Report Display Update Updated Music User Status Status
Soft Phone
Receive Call, Reject, Call Hold Call, Transfer Call
Place Conference Call
Display CDR (Call Detail Report)
Update Music No
Change Music No
Play Music
SIP
Request for CDR (Call Detail Report)
Dial Call, Hold Call, Missed Call, Terminate Call
Soft Phone
Music on Hold (MOH)
Log Generator
Request for Conference Call
Soft PBX
UDP Connection
Request for MOH
8.6.2
Callee
Caller
Operator
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8.6.3
Level 2 Data Flow Diagram
PBX SIP Engine Generate State Machine
MOH
Request for Music on Hold Music on Hold Change Music Track Update Music track
Make Request Update state
SIP Engin
Update state
Make Conference Call
MCU
Request to update node
Request for Conference Call Generate Message
SIP Message Generator
Call Node
Parse Message
Update Node Request to up date
Request For Map
SIP Parser
MAP Update Map
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RTP Processor
CSocket
Stream in Stream out
RTP processor
Get Packet Add Packet
JItterBuffer
Input AudioOutput Audio
Audio Device
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9. Test Plan 9.1 Purpose This document is developed to act as a QA plan for soft phone and soft PBX, being developed for Aftab Alam (Project Advisor). 9.2 Scope The document defines the scope, methodologies, resources and deliverables of testing activities to be undertaken for the application. This document encompasses the items being tested and the testing tasks to be performed to ensure the quality of the system. 9.3 Source Documents Requirement Specification Document Functional Specification Document 9.4 Derived Documents None 9.5 Assumptions/Limitations It is assumed that The system would work under all Microsoft operating systems like windows XP or windows 2000 Pentium 4 system with at least 512 Mb of Ram Full duplex Sound Card A conventional LAN based network or Intranet 100 Mbytes of disk space 9.6 Modules to Be Tested
The testing effort for of the system will test the following modules.
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Number
Module / Task Name Dial call/Place a call, Call conference, Call transfer, Call Record(Soft Phone ) Phone book service/Search person‟s number, add person‟s number(Soft Phone ) Soft phone configuration module/Soft phone Configuration(Soft Phone ) Register user/Registering user to the PBX(Soft Phone ) Call detail reporting/Call information(Soft PBX)
1 2 3 4 5
PBX configuration module/PBX configuration(Soft PBX)
678-
9.7
Admin Login(PBX) Create dial plan(PBX)
Features not to be tested Call hold/Call hold, Talking to another user during call hold(Soft Phone ) Missed calls/View missed calls(Soft Phone ) Receive a call/receive call, Terminate a call(Soft Phone ) MOH(Soft PBX) MCU(Soft PBX) Sip engine generator(Soft PBX) Call recording/Record call(Soft PBX)
9.8
Strategy
For each test type following strategy will be followed. 9.9
Functional Testing
The purpose of functional testing is to reveal defects related to the component‟s functionality. It aims to assist in ensuring that the item under test conforms to the source documents. We would be doing this testing to identify the errors in the beginning and errors in the beginning are easier to find rather than catching them in later stages. 50 FAST National University of Computer and Emerging Sciences Lahore
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9.9.1 Approach Test cases will be based on source documents to cover the functionality for each module. Two test cycles would be conducted for each of the new module/task. One test cycle would be conducted to verify the bugs, which remained open in system. Testing methodology that would be used is black box testing. We would not be doing white box testing since we are constrained by resources of time and persons that are required for testing. 9.9.2
Entrance Criteria The development team delivers a module for testing
9.9.3
Exit Criteria When all test cases for high priority functionality are passed
9.10
Integration Testing The purpose of integration testing is to ensure that all the modules provide their functionality to the user, when all of them are deployed jointly and there is no ripple effect of one on the other. We would be performing the integration testing by combining two applications i.e., soft phone and a soft PBX. We are performing this testing since it‟s very crucial for checking the functionalities of our system as they get fulfilled only when the system is integrated. By combining the two applications and running them jointly, we would be able to identify the workable and non workable functionalities of our system.
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9.10.1
Approach Users will access various components of the application one by one and simultaneously to ensure that all the modules can run concurrently without any problem. Testing methodology that would be used is black box testing. We would not be doing white box testing since we are constrained by resources of time and persons that are required for testing.
9.10.2
Entrance Criteria All modules are delivered for QA Review and functional testing is complete.
9.11
User Interface Testing User Interface testing verifies the user‟s interaction with the software. The goal of UI Testing is to ensure that the User Interface provides the appropriate access and navigation to the user through the functions of the application. We would be doing this testing since our system is very interactive.
9.11.1
Approach Navigate each window to verify the proper states of the objects on each screen as given in the Style Guideline for Web Application document.
9.11.2
Entrance Criteria When released to Quality Assurance, User Interface will be tested for each module.
9.11.3
Exit Criteria Each window adheres to source documents related to UI.
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9.12
Sanity Testing Sanity testing will be performed prior to Integration testing to ensure that the software application is stable for testing and is functioning according to the requirements. Sanity test is meant to run relatively quickly to get a quick assessment of the new software build. This sanity check will let the QA decide if the deployment is of sufficient quality to go forward with a full test effort. We wouldn‟t be dong this testing since we have limited resources available.
9.12.1
Approach Before integration testing, QA would conduct sanity test based on “product risk matrix”. Any problems found would be reported to the project manager and the head of QA department.
9.12.2
Entrance Criteria Sanity testing can start as soon as application is deployed for integration testing.
9.12.3
Exit Criteria If verified from „product risk matrix‟ and random testing of application that all the previously available functionality is delivered.
9.13
Data Verification The purpose of data verification is to ensure that Payroll Application can run smoothly on the Client‟s data. We would be doing data verification since our system is highly interactive.
9.13.1
Approach Data verification approach is given against each module in the following table.
9.13.2
Entrance Criteria Data is given to the QA Department. 53
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9.13.3 Exit Criteria Data is verified and any differences are noted and pointed to the Development
Module
Data File File maintained in the PBX Dial a call containing caller id File maintained in the soft phone Phone book containing service person’s information File maintained in Soft phone the containing configuration soft phone ip address and port number File maintained in PBX PBX configuration configuration containing PBX IP address and port number F File maintained in CDR PBX containing PBX ip address and port number Fi File containing Register User callerid and password Create dial plan
Verification
Strategy
-
Black box testing
-
Black box testing
-
Black box testing
-
Black box testing
-
Black box testing
-
Black box testing
-
Black box testing
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9.14
Defect Tracking
Defects found in software application will be assigned a severity grading and reported in Test Director. A development team member then assigns the defect to a developer, from where it is investigated, characterized and dealt with. It may be fixed, not fixed, reported as pending, found to be a feature or not a defect at all, etc. Feature requests and certain defects can be tailored by the project manager, with the consent of head of QA, to serve the "release criteria”. We would be doing defect tracking since our system have high probability of breakage resulting from catastrophic errors. This would enable us to identify and remove the catastrophic errors which would increase our confidence in the system. This would also help us to identify those errors which need to be immediately removed.
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Defect Tracking Process Flow Test
Add defect
Defect Found
Added in Test Repository
Development Investigate Assign Defect
Assigned to a Developer in Test Repository
Problem Identified
Developer fixes the defect Bug fixtue communicated to Dev Lead/PM
Pending Not a Bug Change Dev Status
Development Status updated to Fixed in the Test repository
Dev Lead/PM notify QA
Open Defect Verified
Change QA Status
QA Status Updated in Test Repository
Test Passed
Defect Closed
Close Defect
QA Status changed to Closed in Test Repository
Retest in the next cycle Reopen
Retested
Not Fixed QA Status Updated in Test Repository
Release
Released
Reported
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Severity Level Criteria for a Defect A five-level criterion will be used to assign Severity to a Defect in the Defect Repository. 9.15
Guidelines for Assigning severity to a defect
Urgent The defect results in the failure of the complete software system, or of a subsystem within the system in such a way that testing cannot continue. Complete inability to use the subsystem under test is the examples of this type. This is reserved for only the most catastrophic of problems Very High The defect results in the failure of a subsystem, or of a software unit (program or module) within the system. Data corruption errors lie in this area. High The defect does not result in a failure, but causes the system to produce incorrect, incomplete, or inconsistent results. However, there are acceptable processing alternatives, which will yield the desired result. Medium The defect does not cause a failure, and the desired processing results are easily obtained by working around the defect. These issues are generally not damaging to system operation, but instead, are more of an annoyance to the user Low
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The defect is the result of non-conformance to a standard, is related to the aesthetics of the system, or is a request for an enhancement. Defects at this level may be deferred or even ignored. Rule of Thumb Common sense should always have precedence over these rules. Any functional or aesthetic area emphasized by the customer would carry greater severity than the severity assigned by above guidelines. Quality Control All the test cases will be technically reviewed between the test cycles. Any deviations from the test requirements will be noted accordingly and the test cases will be updated to ensure that these test cases conform to the updated application requirements from the client. Team Meetings The test team will meet once every two weeks/after-each-test-cycle to evaluate the progress to date and to identify error trends and problems as early as possible. The test team leader will meet with development/project manager once every two weeks as well. These two meetings can be scheduled on different weeks. Additional meetings can be called as required for emergency situations. 9.16
Test Suspension and Resumption Criteria
Integration testing would only be conducted after application passes Sanity testing. If application fails in sanity testing then the test-effort will remain suspended until all the previously working functionality is available again and newly fixed defects are verified at random basis.
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9.17
Schedule Overview
QA Schedule will be provided separately in Microsoft Project 2000 Format Test Configuration Control
Test Server Update/ Install Request
QA Lead Notified
Analyze Test activity on the server
System Busy
Requester Notified of the delay
Yes
No
System Ready for Update QA Lead to Communicate Username/Password for system access Development/ Operations Update the Test Server Access rights on Test Server changed by QA Lead
Process
Password Changed and Testing Started
A requester can be any person from the Development or the Operations Department 9.18
Testing Resources
Name
Responsibilities
Time Software Hardware Man power Testing tools Risk management
To test the system and to identify the errors 59
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9.19
Summary
Following is the summary of test execution on the system. Majority of the bugs, which are not closed in release 1, have been given Low priority by the development for shipping them with future releases.
Bugs Total Bugs reported by QA =5 ________________________________________________ Bugs Closed till Extended Integration Testing Closed for ATM Closed by PM Remaining Bugs (QA Status)
=5 =3 =2 =1
Development Priority of Open bugs ------------------------------------------------------------------------Low Medium High Unknown
=2 =1 =1 =1
Suggestions Total Suggestion posted in test repository QA =10 __________________________________________________ Suggestions Closed Closed for ATM
=5 =5
Remaining Suggestions
= nil
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10. TEST CASES 10.1
Dial and Place Call
GUI No 1 (Soft phone Interface)
GUI NO 2
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GUI NO 3
GUI NO 4
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GUI NO 5
GUI NO 6
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Test Case ID:
1 (Dial call and place call)
QA Test Engineer
Khadija Akram
Test case Version:
1.0 7-05-2008
Test Execution Date: Use Case Reference(s)
GUI Reference(s)
Objective Product / Ver/ Module Environment: Assumptions: Pre-Requisite:
Test Case Description
Input Parameters
Reviewed By:
Testing Team lead (Umair Ashraf )
Use case id 1 (Dial call) GUI No 1 (soft phone interface) GUI No 2 GUI No 3 GUI No 4 GUI No 5 To Completely Verify and Validate the dial call functionality of the soft phone System. This test case will completely check the behavior of dial call operation for various inputs in the soft phone system. Soft phone System Ver 1.0 /dial call Function) windows xp or windows 2000 Environment Microsoft Internet explorer 6.0 The network connection must be fast enough and it should not be congested otherwise it might effect the Testing process. The caller and the calle should both be registered to the ip based PBX Testing of the dial call functionality of the soft phone system. It involves testing the behavior of the dial call functionality of the system by checking all possible outputs with respect to the possible inputs given to it. Expected Output
Actual Output
Test Conformance Status
Possible Reason(s) in case of failure
(Valid Data) The dial number should consist of four digits only. 4 digit dial number 1890
Invalid inputs Greater than 4 digit Dial number 90898 Less than 4 digit
For all valid data of equivalence classes
For all valid data of equivalence classes
Output of the system should be successful GUI no 2 will be displayed to the caller and 4 will be displayed to the callee .Dialing without registering will display GUI no 6
Successful Validation of the number and display of GUI no 2 to the caller and 4 to the callee will occur.
Passed for all valid input data cases No Test Case failed!!! No failure occurred .The only possible case was that the network was busy and system responded in terms of exceptional GUI (GUI no 5 exceptional GUI)
For all invalid data of equivalence classes.
For all invalid data
Dial error and system
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equivalence classes.
Number. 109 108 16
10.2
will display GUI no 3
Passed for all in valid input data cases
Dial error and system will display GUI no 3
Transfer Call
GUI NO 7
Test Case ID:
2 (Transfer call)
QA Test Engineer
Khadija Akram
Test case Version:
2.0 7-05-2008
Test Execution Date: Use Case Reference(s)
GUI Reference(s)
Objective Product / Ver/ Module Environment: Assumptions: Pre-Requisite: Test Case Description
Reviewed By:
Testing Team lead (Umair Ashraf )
Use case id 6(Transfer call ) GUI No 1 GUI No 2 GUI No 3 GUI No 5 GUI No 7 To Completely Verify and Validate the Transfer call functionality of the soft phone System. This test case will completely check the behavior of Transfer call operation for various inputs in the soft phone system. Soft phone System Ver 1.0 Transfer call Function) windows xp or windows 2000 Environment Microsoft Internet explorer 6.0 The network connection must be FAST enough and it should not be congested otherwise it might effect the Testing process. The users should be registered with the soft pbx. Testing of the Transfer call functionality of the soft phone system. It involves testing the
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behavior of the Transfer call functionality of the system by checking all possible outputs with respect to the possible inputs given to it. Input Parameters
Expected Output
Actual Output
Test Conformance Status
Possible Reason(s) in case of failure
For all valid data of equivalence classes For all valid data of equivalence classes
(Valid Data) The dial number should consist of four digits only 4 digit dial number 1789
Output of the system should be successful GUI no2 will be displayed on the caller side and 4 will be displayed on the callee side.
Successful Validation of the number and display of GUI no 2 to caller and 4 on the callee side.
For all invalid data of equivalence classes. For all invalid data equivalence classes. Invalid inputs Greater than 4 digit Dial number 90876 Less than 4 digit Number 987 908 90
Dial error and system will display GUI no 3.
Dial error and system will display GUI no 3
Passed for all valid input data cases
Passed for all in valid input data cases
No Test Case failed!!! No failure occurred .The only possible case was that the network was busy and system responded in terms of exceptional GUI (GUI no 5 exceptional GUI)
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10.3
Call Conference
GUI NO 8
Test Case ID:
3 (Call Conference)
QA Test Engineer
Khadija Akram
Test case Version:
3.0 7-05-2008
Test Execution Date: Use Case Reference(s)
GUI Reference(s)
Objective Product / Ver/ Module Environment: Assumptions: Pre-Requisite:
Test Case Description
Input Parameters
Testing Team lead (Umair Ashraf )
Use case id 7(Call Conference ) GUI No 1 GUI No 2 GUI No 3 GUI No 5 GUI No 8 To Completely Verify and Validate the Call Conference functionality of the soft phone System. This test case will completely check the behavior of Call Conference operation for various inputs in the soft phone system. Soft phone System Ver 1.0 Call Conference Function) windows xp or windows 2000 Environment Microsoft Internet explorer 6.0 The network connection must be FAST enough and it should not be congested otherwise it might effect the Testing process. The users should be registered with the soft PBX Testing of the transfer Call Conference of the soft phone system. It involves testing the behavior of the Call Conference functionality of the system by checking all possible outputs with respect to the possible inputs given to it. Expected Output
Actual Output
Test Conformance Status
For all valid data of equivalence classes
(Valid Data) The dial number should consist of four digits only. 4 digit dial number 1789 3002
Reviewed By:
For all valid data of equivalence classes
Output of the system should be successful GUI no 1 will be
Successful Validation of the number and display of GUI 1 will be displayed on the screen.
Passed for all valid input data cases
Possible Reason(s) in case of failure
No Test Case failed!!! No failure occurred .The only possible case was that the network was busy and system responded in terms of exceptional GUI (GUI no 5 exceptional GUI)
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displayed on the screen.
3004
Invalid inputs Greater than 4 digit Dial number 90876 Less than 4 digit Number 987 23
For all invalid data equivalence classes.
For all invalid data of equivalence classes.
Passed for all in valid input data cases
Dial error and system will display GUI no 3 Dial error and system will display GUI no3
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10.4
Phone Book
GUI NO 9
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GUI NO 10
GUI NO 11
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GUI NO 12
GUI NO 13
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Test Case ID:
4 (Phone Book)
QA Test Engineer
Khadija Akram
Test case Version:
4.0 7-05-2008
Test Execution Date: Use Case Reference(s)
GUI Reference(s)
Objective Product / Ver/ Module Environment: Assumptions: Pre-Requisite:
Test Case Description
Input Parameters
(Valid Data for caller id,caller name, caller address and mobile number )
Caller Id The caller id should consist of four digits only. 4 digit caller id 1789 3004
Caller Name The caller name (First name and last name) should consist of a maximum of 30 characters. Spaces are allowed between first name and the last name. The first name and the last name must consist of 3 characters. Digits
Reviewed By:
Testing Team lead (Umair Ashraf )
Use case id 9(Phone Book ) GUI No 1 GUI No 9 GUI No 10 GUI No 11 GUI No 12 GUI No 13 To Completely Verify and Validate the Phone Book functionality of the soft phone. This test case will completely check the behavior of Phone Book operation for various inputs in the system. Soft phone System Ver 1.0 Phone Book) windows xp or windows 2000 Environment Microsoft Internet explorer 6.0 The network connection must be FAST enough and it should not be congested otherwise it might effect the Testing process. None. Testing of the Phone Book of the functionality of the soft phone. It involves testing the behavior of the Register User functionality of the soft phone by checking all possible outputs with respect to the possible inputs given to it. Expected Output
Actual Output
Test Conformance Status
Possible Reason(s) in case of failure
For all valid data of equivalence classes For all valid data of equivalence classes
Output of the system should be successful GUI no1 will be displayed in case of add record,GUI no 11 in case of record list,GUI no 12 in case of search record and GUI no 13 in case of wrong caller id,caller name, caller address and mobile number.
Successful Validation of the number GUI no1 will be displayed in case of add record,GUI no 11 in case of record list,GUI no 12 in case of search record and GUI no 13 in case of wrong caller id,caller name, caller address and mobile number.
Passed for all valid input data cases
No Test Case failed!!! No failure occurred
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are not allowed. Aklo09 abc
Caller Address The caller address should consist of a minimum of 1 digit in the beginning and a maximum of 3 digits. A maximum of 30 characters can come after the specified digits. There can be nothing after the 3 digit address. There can be spaces between the address. 468 a block model town Lahore
Mobile No The mobile no must be of 11 digits. 03216543789 (Invalid Data for add record ) Invalid inputs for caller id Greater than 4 digit caller id 90876 Less than 4 digit caller id 987 146 Invalid inputs for caller name Ab Ab0 A2
Passed for all in valid input data cases For all invalid data of equivalence classes. For all invalid data equivalence classes Dial error and system will display GUI no 13.
Dial error and system will display GUI no 13
Invalid inputs for caller address Hjkk 1234 Invalid inputs for mobile number 126576
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10.5
Soft Phone Configuration
GUI NO 14
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GUI NO 15
Test Case ID:
5 (Soft Phone Configuration)
QA Test Engineer
Khadija Akram
Test case Version:
5.0 7-05-2008 GUI NO 1 GUI NO 11 GUI NO 12 To Completely Verify and Validate the Soft Phone Configuration. This test case will completely check the behavior of Soft Phone Configuration r operation for various inputs in the system. Soft phone System Ver 1.0 Soft Phone Configuration) windows xp or windows 2000 Environment Microsoft Internet explorer 6.0 The network connection must be FAST enough and it should not be congested otherwise it might effect the Testing process. Nothing
Test Execution Date: GUI Reference(s)
Objective Product / Ver/ Module Environment: Assumptions: Pre-Requisite:
Reviewed By:
Testing Team lead (Umair Ashraf )
Test Case Description
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Testing of the Soft Phone Configuration. It involves testing the behavior of the Soft Phone Configuration by checking all possible outputs with respect to the possible inputs given to it. Input Parameters
(Valid Data)
PBX IP Address The ip address must be of standard format. 128.0.01.
Expected Output
Actual Output
Test Conformance Status
Possible Reason(s) in case of failure
For all valid data of equivalence classes Passed for all valid input data cases
Output of the system should be successful GUI no 1 will be displayed on the screen.
Caller ID
For all valid data of equivalence classes
The caller id should consist of four digits only
Successful Validation of the number and display of GUI 1 will be displayed on the screen.
4 digit caller id 1789
Caller Name The caller name (First name and last name) should consist of a maximum of 25 characters. Spaces are allowed between first name and the last name. The first name and the last name must consist of 3 characters. Only first name consisting of 3 characters is applicable also. Digits are not allowed.
No Test Case failed!!! No failure occurred.
amn amnb abc
Caller IP The caller ip must be of standard format. 128.0.01.
Media IP The media IP must be of standard format. 128.0.01.
For all invalid data equivalence classes.
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The PBX port should consist of four digits only. 4 digit PBX Port 1789
Dial error and system will display GUI no13
Passed for all in valid input data cases
3002 Caller Port The caller port should consist of four digits only. No characters or special characters are allowed. 4 digit caller Port 1789
For all invalid data of equivalence classes. Dial error and system will display GUI no 13
Sending Port The sending port should consist of four digits only 4 digit sending Port 1789
Listening Port The listening port should consist of four digits only. 4 digit sending Port 1789
Password The password can be anything Only digits 6754 only characters abc
Invalid inputs for PBX IP address 149.09 657 Invalid inputs for caller id Greater than 4 digit caller id 90876 Less than 4 digit caller id 987
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Invalid inputs for caller name Ab Ab0 Abc vg Abc –vg Invalid inputs for caller IP 456 Invalid inputs for media IP 1345 Invalid inputs for PBX port Greater than 4 digit PBX port 90876 Less than 4 digit PBX port 987
Invalid inputs for caller port Greater than 4 digit caller port 90876 Less than 4 digit caller port 987 Invalid inputs for sending port Greater than 4 digit Sending port 90876 Less than 4 digit sending port 987
Invalid inputs for receiving port Greater than 4 digit receiving port 90876 Less than 4 digit receiving port 987 Invalid inputs for password none
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10.6
PBX Configuration
GUI NO 16
GUI NO 17
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Test Case ID:
6 (PBX Configuration)
QA Test Engineer
Khadija Akram
Test case Version:
6 7-05-2008
Test Execution Date: Use Case Reference(s) GUI Reference(s)
Objective Product / Ver/ Module Environment: Assumptions: Pre-Requisite:
Test Case Description
Input Parameters
Reviewed By:
Testing Team lead (Umair Ashraf )
Use case id GUI No 13 GUI No 16 GUI No 17 To Completely Verify and Validate the PBX Configuration functionality of the System. This test case will completely check the behavior of dial call operation for various inputs in the soft phone system. PBX System Ver 1.0 / PBX Configuration) windows xp or windows 2000 Environment Microsoft Internet explorer 6.0 The network connection must be fast enough and it should not be congested otherwise it might effect the Testing process. The admin should be logged in. Testing of the PBX Configuration functionality of the system. It involves testing the behavior of the PBX Configuration functionality of the system by checking all possible outputs with respect to the possible inputs given to it. Expected Output
Actual Output
Test Conformance Status
Possible Reason(s) in case of failure
(Valid Data)
PBX IP The PBX IP should be standard format. 128.0.01.
For all valid data of equivalence classes Output of the system should be successful GUI no 16 will be displayed.
For all valid data of equivalence classes
Passed for all valid input data cases
Successful Validation of the number and display of GUI 16.
Listening port
No Test Case failed!!! No failure occurred.
The listening port should consist of four digits only. No characters 4 digit listening Port 1789 For all invalid data equivalence classes. Invalid inputs for listening port Greater than 4 digit listening port 90876 Less than 4 digit listening port 987
Dial error and system will display GUI 13
For all invalid data of equivalence classes.
Dial error and system will display GUI no 13
Passed for all in valid input data cases
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10.7
PBX Admin Login
GUI NO 18
GUI NO 19
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Test Case ID:
7 (PBX Admin login)
QA Test Engineer
Khadija Akram
Test case Version:
7 8-05-2008
Test Execution Date: Use Case Reference(s) GUI Reference(s)
Objective Product / Ver/ Module Environment: Assumptions: Pre-Requisite:
Test Case Description
Input Parameters
Login Id The login id should be admin
Testing Team lead (Umair Ashraf )
Use case id GUI No 11 GUI No 13 GUI No 15 To Completely Verify and Validate the PBX Admin login functionality of the System. This test case will completely check the behavior of PBX Admin login for various inputs in the system. PBX System Ver 1.0 / PBX Admin login) windows xp or windows 2000 Environment Microsoft Internet explorer 6.0 The network connection must be fast enough and it should not be congested otherwise it might effect the Testing process. None Testing of the PBX Admin login functionality of the system. It involves testing the behavior of the PBX Admin login functionality of the system by checking all possible outputs with respect to the possible inputs given to it. Expected Output
For all valid data of equivalence classes (Valid Data)
Reviewed By:
Output of the system should be successful GUI no 18 will be displayed.
Actual Output
For all valid data of equivalence classes
Test Conformance Status
Passed for all valid input data cases
Successful Validation of the number and display of GUI 18.
No Test Case failed!!! No failure occurred.
Password Can be digits only 123 456
Invalid inputs for login id Admin
Possible Reason(s) in case of failure
For all invalid data equivalence classes. Dial error and system will display GUI 19
For all invalid data of equivalence classes.
Dial error and system will display GUI no 19
Passed for all in valid input data cases
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10.8
Create Dial Plan
GUI NO 20
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Test Case ID:
8(Create dial plan)
QA Test Engineer
Khadija Akram
Test case Version:
8 8-05-2008
Test Execution Date: Use Case Reference(s) GUI Reference(s)
Objective Product / Ver/ Module Environment: Assumptions: Pre-Requisite:
Test Case Description
Input Parameters
(Valid Data)
Caller id 4 digit caller id 1789
Testing Team lead (Umair Ashraf )
Use case id 14 GUI No 13 GUI No 16 GUI No 20 To Completely Verify and Validate the Create dial plan functionality of the System. This test case will completely check the behavior of Create dial plan for various inputs in the system. PBX System Ver 1.0 / Create dial plan ) windows xp or windows 2000 Environment Microsoft Internet explorer 6.0 The network connection must be fast enough and it should not be congested otherwise it might effect the Testing process. None Testing of the Create dial plan functionality of the system. It involves testing the behavior of the Create dial plan functionality of the system by checking all possible outputs with respect to the possible inputs given to it. Expected Output
Actual Output
For all valid data of equivalence classes Output of the system should be successful GUI no 16 will be displayed.
Password Consisting of digits only 1234
Invalid inputs for caller id Greater than 4 digit Caller id 90876 Less than 4 digit caller id 987
Reviewed By:
For all valid data of equivalence classes
Test Conformance Status
Possible Reason(s) in case of failure
Passed for all valid input data cases
Successful Validation of the number and display of GUI 16.
No Test Case failed!!! No failure occurred.
For all invalid data of equivalence classes. For all invalid data equivalence classes. Dial error and system will display GUI 13
Dial error and system will display GUI no 13 Passed for all in valid input data cases
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10.9
CDR (Call Detail Reporting)
GUI NO 21
Test Case ID:
9(Call Detail Reporting)
QA Test Engineer
Khadija Akram
Test case Version: Test Execution Date:
10 8-05-2008
Use Case Reference(s)
Use case id 5
GUI Reference(s)
GUI No 21
Objective Product / Ver/ Module Environment: Assumptions: Pre-Requisite:
Test Case Description
Reviewed By:
Testing Team lead (Umair Ashraf )
To Completely Verify and Validate the Call Detail Reporting functionality of the soft phone. This test case will completely check the behavior of Call Detail Reporting for various inputs in the system. Soft phone System Ver 1.0 / Call Detail Reporting) windows xp or windows 2000 Environment Microsoft Internet explorer 6.0 The network connection must be fast enough and it should not be congested otherwise it might effect the Testing process. None Testing of the Call Detail Reporting functionality of the system. It involves testing the behavior of the Call Detail Reporting functionality of the system by checking all possible outputs with respect to the possible inputs given to it.
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Input Parameters
Expected Output
Actual Output
Test Conformance Status
Possible Reason(s) in case of failure
(Valid Data)
Caller id 4 digit caller id 1789
Callee id 4 digit callee id 1789
Invalid inputs for caller id Greater than 4 digit Caller id 90876 Less than 4 digit caller id 987 Invalid inputs for callee id Greater than 4 digit Caller id 90876 Less than 4 digit caller id 987
For all valid data of equivalence classes Output of the system should be successful GUI no 21 will be displayed
For all valid data of equivalence classes
Passed for all valid input data cases
Successful Validation of the number and display of GUI no 21
No Test Case failed!!! No failure occurred.
For all invalid data equivalence classes. Dial error and system will display GUI 13
For all invalid data of equivalence classes.
Dial error and system will display GUI no 13
Passed for all in valid input data cases
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10.10 Register User GUI NO 22
GUI NO 23
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GUI NO 24
GUI NO 25
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GUI NO 26
Test Case ID:
10(Register User)
QA Test Engineer
Khadija Akram
Test case Version:
10 8-05-2008
Test Execution Date:
Reviewed By:
Testing Team lead (Umair Ashraf )
Product / Ver/ Module
Use case id 4 GUI No 1 GUI No 20 GUI No 21 GUI No 22 GUI No 23 GUI No 24 GUI No 25 GUI No 26 To Completely Verify and Validate the register user functionality of the system. This test case will completely check the behavior of register user for various inputs in the system. Soft phone System Ver 1.0 / Register User)
Environment:
windows xp or windows 2000 Environment
Use Case Reference(s)
GUI Reference(s)
Objective
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Microsoft Internet explorer 6.0 Assumptions: Pre-Requisite:
Test Case Description
Input Parameters
(Valid Data)
Caller id 4 digit caller id 1789
The network connection must be fast enough and it should not be congested otherwise it might effect the Testing process. None Testing of the register user functionality of the system. It involves testing the behavior of the register user functionality of the system by checking all possible outputs with respect to the possible inputs given to it. Expected Output
For all valid data of equivalence classes Output of the system should be successful GUI no 25 will be displayed
Actual Output
For all valid data of equivalence classes
Test Conformance Status
Possible Reason(s) in case of failure
Passed for all valid input data cases
Successful Validation of the number and display of GUI no 25
Password Consists of digits only 1234 Invalid inputs for caller id Greater than 4 digit Caller id 90876 Less than 4 digit caller id 987
No Test Case failed!!! No failure occurred.
For all invalid data equivalence classes. Dial error and system will display GUI 23
For all invalid data of equivalence classes.
Dial error and system will display GUI no 23
Passed for all in valid input data cases
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10.11 Receiving/rejecting a call GUI NO 27
GUI NO 28
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GUI NO 29
GUI NO 30
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Test Case ID:
11(Receive/Reject call)
QA Test Engineer
Khadija Akram
Test case Version:
10 8-05-2008
Test Execution Date: Use Case Reference(s)
GUI Reference(s)
Objective Product / Ver/ Module Environment: Assumptions: Pre-Requisite:
Test Case Description
Input Parameters
When call successfully establishes between a caller and the calle, following scenarios are possible. Scenario 1 In case of unattended call from the callee side,Gui no 27 will be displayed to the caller showing that there is no answer from the callee side. Missed call notification will occur on the callee side. Scenario 2 In case both caller and the callee accepts the call, then gui no 29 and 30 will be displayed to the caller and the callee.
Reviewed By:
Testing Team lead (Umair Ashraf )
Use case id 4 GUI No 1 GUI No 27 GUI No 28 GUI No 29 GUI No 30 To Completely Verify and Validate the Receive/Reject call functionality of the soft phone. This test case will completely check the behavior of Receive/Reject call for various inputs in the soft phone.. Soft phone System Ver 1.0 / Receive/Reject call) windows xp or windows 2000 Environment Microsoft Internet explorer 6.0 The network connection must be fast enough and it should not be congested otherwise it might effect the Testing process. The call between the caller and the callee should be established. Testing of the receive/reject functionality of the system. It involves testing the behavior of the receive/reject functionality of the system by checking all possible outputs with respect to the possible inputs given to it. Expected Output
Actual Output
In case of unattended call from the callee side, Gui no 27 will be displayed to the caller showing that there is no answer from the callee side. Missed call notification will occur on the callee side.
In case of unattended call from the callee side, Gui no 27 will be displayed to the caller showing that there is no answer from the callee side. Missed call notification will occur on the callee side.
In case both caller and the callee accepts the call, then gui no 29 and 30 will be displayed to the caller and the callee.
In case both caller and the callee accepts the call, then gui no 29 and 30 will be displayed to the caller and the callee
Scenario 3 In case the callee is busy,Gui no 31 will be displayed to the caller displaying that
Scenario 3 In case the callee is busy, Gui no 31 will be displayed to the caller displaying that the callee is busy.
Test Conformance Status
Possible Reason(s) in case of failure
Passed for all valid input data cases
No Test Case failed!!! No failure occurred.
Passed for all in valid input data cases
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the callee is busy. Scenario 3 In case the callee is busy, Gui no 31 will be displayed to the caller displaying that the callee is busy.
10.12 Speed Dial GUI NO 32
GUI NO 33
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Test Case ID:
12(Speed Dial)
QA Test Engineer
Khadija Akram
Test case Version:
8-05-2008
Test Execution Date: Use Case Reference(s) GUI Reference(s)
Objective Product / Ver/ Module Environment: Assumptions: Pre-Requisite:
Test Case Description
Input Parameters
1digit dial number
Testing of the speed dial functionality of the system. It involves testing the behavior of the speed dial functionality of the system by checking all possible outputs with respect to the possible inputs given to it. Expected Output
Output of the system should be successful GUI no 32 and 33 will be displayed.
1 digit only 8
Invalid inputs for dial number 56
Testing Team lead (Umair Ashraf )
Use case id 14 GUI No 13 GUI No 32 GUI No 33 To Completely Verify and Validate the Speed Dial functionality of the System. This test case will completely check the behavior of Speed Dial for various inputs in the system. PBX System Ver 1.0 / Speed Dial ) windows xp or windows 2000 Environment Microsoft Internet explorer 6.0 The network connection must be fast enough and it should not be congested otherwise it might effect the Testing process. None
For all valid data of equivalence classes (Valid Data)
Reviewed By:
Actual Output
For all valid data of equivalence classes
Test Conformance Status
Possible Reason(s) in case of failure
Passed for all valid input data cases
Successful Validation of the number and display of GUI 32 and 33 No Test Case failed!!! No failure occurred.
For all invalid data equivalence classes. Dial error and system will display GUI 13
For all invalid data of equivalence classes. Dial error and system will display GUI no 13
Passed for all in valid input data cases
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10.13 Missed Calls GUI NO 34
GUI NO 35
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Test Case ID:
13(Missed call)
QA Test Engineer
Khadija Akram
Test case Version:
10 8-05-2008
Test Execution Date: Use Case Reference(s)
GUI Reference(s)
Objective Product / Ver/ Module Environment: Assumptions: Pre-Requisite:
Test Case Description
Input Parameters
Reviewed By:
Testing Team lead (Umair Ashraf )
Use case id 4 GUI No 1 GUI No 34 GUI No 35 To Completely Verify and Validate the missed call functionality of the soft phone. This test case will completely check the behavior of missed call for various inputs in the soft phone.. Soft phone System Ver 1.0 / missed call) windows xp or windows 2000 Environment Microsoft Internet explorer 6.0 The network connection must be fast enough and it should not be congested otherwise it might effect the Testing process. The call between the caller and the callee should be established. Testing of the missed call functionality of the system. It involves testing the behavior of the missed call functionality of the soft phone by checking all possible outputs with respect to the possible inputs given to it. Expected Output
Actual Output
Test Conformance Status
Possible Reason(s) in case of failure
Passed for all valid input data cases
Whenever user wants to view the missed calls, he will press the missed call button and display of gui no 35 will be displayed to the user.
Whenever user wants to view the missed calls, he will press the missed call button and display of gui no 35 will be displayed to the user.
Whenever user wants to view the missed calls, he will press the missed call button and display of gui no 35 will be displayed to the user.
No Test Case failed!!! No failure occurred.
Passed for all in valid input data cases
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10.14 Terminating the call GUI NO 36
GUI NO 37
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Test Case ID:
14(Terminating the call )
QA Test Engineer
Khadija Akram
Test case Version:
10 8-06-2008
Test Execution Date: Use Case Reference(s)
GUI Reference(s)
Objective Product / Ver/ Module Environment: Assumptions: Pre-Requisite:
Test Case Description
Input Parameters
Reviewed By:
Testing Team lead (Umair Ashraf )
Use case id 4 GUI No 1 GUI No 36 GUI No 37 To Completely Verify and Validate the Terminating the call functionality of the soft phone. This test case will completely check the behavior of Terminating the call for various inputs in the soft phone. Soft phone System Ver 1.0 / Terminating the call) windows xp or windows 2000 Environment Microsoft Internet explorer 6.0 The network connection must be fast enough and it should not be congested otherwise it might effect the Testing process. The call between the caller and the callee should be established. Testing of the terminating call functionality of the system. It involves testing the behavior of the terminating call functionality of the soft phone by checking all possible outputs with respect to the possible inputs given to it. Expected Output
Actual Output
Test Conformance Status
Possible Reason(s) in case of failure
Passed for all valid input data cases
Whenever user wants to terminate the call, then display of gui no 36 will be displayed on the user who terminated the call and 37 on that user who was talking to the user who terminated the call.
Whenever user wants to terminate the call, then display of gui no 36 will be displayed on the user who terminated the call and 37 on that user who was talking to the user who terminated the call.
Whenever user wants to terminate the call, then display of gui no 36 will be displayed on the user who terminated the call and 37 on that user who was talking to the user who terminated the call.
No Test Case failed!!! No failure occurred.
Passed for all in valid input data cases
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10.15 Call Hold GUI NO 38
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Test Case ID:
15(Call Hold )
QA Test Engineer
Khadija Akram
Test case Version:
10 8-06-2008
Test Execution Date: Use Case Reference(s)
GUI Reference(s)
Objective Product / Ver/ Module Environment: Assumptions: Pre-Requisite:
Test Case Description
Input Parameters
Reviewed By:
Testing Team lead (Umair Ashraf )
Use case id 4 GUI No 1 GUI No 38 GUI No 39 To Completely Verify and Validate the Terminating the call functionality of the soft phone. This test case will completely check the behavior of Terminating the call for various inputs in the soft phone. Soft phone System Ver 1.0 / Call Hold) windows xp or windows 2000 Environment Microsoft Internet explorer 6.0 The network connection must be fast enough and it should not be congested otherwise it might effect the Testing process. The call between the caller and the callee should be established. Testing of the call hold functionality of the system. It involves testing the behavior of the call hold functionality of the soft phone by checking all possible outputs with respect to the possible inputs given to it. Expected Output
Actual Output
Test Conformance Status
Possible Reason(s) in case of failure
Passed for all valid input data cases
Whenever user wants to hold the call, then display of gui no 37 and 38 will be displayed on the caller and the callee.
Whenever user wants to hold the call, then display of gui no 37 and 38 will be displayed on the caller and the callee.
Whenever user wants to hold the call, then display of gui no 37 and 38 will be displayed on the caller and the callee.
No Test Case failed!!! No failure occurred.
Passed for all in valid input data cases
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11. Appendix Real-time Transport Protocol (RTP) Real-time Transport Protocol (RTP), which provides end-to-end delivery services for data with real-time characteristics, such as interactive Audio and video. Services include payload type identification, sequence numbering, time stamping and delivery monitoring. The media gateways that digitize voice use the RTP protocol to deliver the voice (bearer) traffic. The RTP protocol
provides features for real-time applications, with the ability to
reconstruct timing, loss detection, security, content delivery and identification of encoding schemes. For each participant, a particular pair of destination IP addresses defines the session between the two endpoints, which translate into a single RTP session for each phone call in progress. RTP is an application service built on UDP, so it is connectionless, with best-effort delivery. Although RTP is connectionless, it does have a sequencing system that allows for the detection of missing packets. As part of its specification, the RTP Payload Type field includes the encoding scheme that the media gateway uses to digitize the voice content. This field identifies the RTP payload format and determines its interpretation by the CODEC in the media gateway. A profile specifies a default static mapping of payload type codes to payload formats. With the different types of encoding schemes and packet creation rates, RTP packets can vary in size and interval. Administrators must take RTP parameters into account when planning voice services. All the combined parameters of the RTP sessions dictate how much bandwidth is consumed by the voice bearer traffic. RTP traffic that carries voice traffic is the single greatest contributor to the VoIP network load. [Reference 3].
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Session Initiation Protocol (SIP) The Session Initiation Protocol is part of IETF's multimedia data and control protocol framework. SIP is a powerful client-server signaling protocol used in VoIP networks. SIP handles the setup and tears down of multimedia sessions between speakers; these sessions can include multimedia conferences, telephone calls, and multimedia distribution. SIP is a text-based signaling protocol transported over either TCP or UDP, and is designed to be lightweight. It inherited some design philosophy and architecture from the Hypertext Transfer Protocol (HTTP) and Simple Mail Transfer Protocol (SMTP) to ensure its simplicity, efficiency and extensibility. SIP uses invitations to create Session Description Protocol (SDP) messages to carry out capability exchange and to setup call control channel use. These invitations allow participants to agree on a set of compatible media types. SIP supports user mobility by proxying and redirecting requests to the user's current location. Users can inform the server of their current location (IP address or URL) by sending a registration message to a registrar. This function is powerful and often needed for a highly mobile voice user base. The SIP client-server application has two modes of operation; SIP clients can ether signal through a proxy or redirect server. Using proxy mode, SIP clients send requests to the proxy and the proxy either handles requests or forwards them on to other SIP servers. Proxy servers can insulate and hide SIP users by proxying the signaling messages; to the other users on the VoIP network, the signaling invitations look as if they are coming from the proxy SIP server. [Reference 4].
Jitter Jitter is the measure of time between when a packet is expected to arrive to when it actually arrives. In other words, with a constant packet transmission rate of every 20 ms, every packet would be expected to arrive at the destination exactly every 20 ms. 103 FAST National University of Computer and Emerging Sciences Lahore
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The greatest culprit of jitter is queuing variations caused by dynamic changes in network traffic loads. Another cause is packets that might sometimes take a different equal-cost link that is not physically (or electrically) the same length as the other links. Media gateways have play-out buffers that buffer a packet stream, so that the reconstructed voice waveform is not affected by packet jitter. Play-out buffers can minimize the effects of jitter, but cannot eliminate severe jitter. Although some amount of jitter is to be expected, severe jitter can cause voice quality issues because the media gateway might discard packets arriving out of order. In this condition, the media gateway could starve its play-out buffer and cause gaps in the reconstructed waveform. [Reference 5].
G.729 Codec G.729 is an audio data compression algorithm for voice that compresses voice audio in chunks of 10 milliseconds. G.729 is mostly used in Voice over IP (VoIP) applications like SIP phones for its low bandwidth requirement. Standard G.729 operates at 8 kbit/s, but there are extensions, which provide also 6.4 kbit/s and 11.8 kbit/s rates for marginally worse and better speech quality respectively. It also requires less computation during encoding and decoding.
12. References 1. 2. 3. 4. 5.
Carrier Grade Voice over IP by Daniel Collins Understanding VOIP networks by Juniper Network Real-time protocol RFC SIP RFC Voice over IP Fundamentals by Jonathan Davidson
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