10. TEST CASES
10.14 Terminating the call
GUI NO 36
Test Case ID: 14(Terminating the call )
QA Test Engineer Khadija Akram Reviewed By: Testing Team lead (Umair
Ashraf )
Test case Version: 10
Test Execution Date: 8-06-2008
Use Case Reference(s) Use case id 4
GUI Reference(s)
GUI No 1 GUI No 36 GUI No 37
Objective
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.
Product / Ver/ Module Soft phone System Ver 1.0 / Terminating the call)
Environment: windows xp or windows 2000 Environment
Microsoft Internet explorer 6.0
Assumptions: The network connection must be fast enough and it should not be congested otherwise it
might effect the Testing process.
Pre-Requisite: The call between the caller and the callee should be established.
Test Case Description
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.
Input Parameters Expected Output Actual Output Test Conformance Status
Possible Reason(s) in case of failure
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.
Passed for all valid input data cases
Passed for all in valid input data cases
No Test Case failed!!! No failure occurred.
10.15 Call Hold
Test Case ID: 15(Call Hold )
QA Test Engineer Khadija Akram Reviewed By: Testing Team lead (Umair
Ashraf )
Test case Version: 10
Test Execution Date: 8-06-2008
Use Case Reference(s) Use case id 4
GUI Reference(s)
GUI No 1 GUI No 38 GUI No 39
Objective
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.
Product / Ver/ Module Soft phone System Ver 1.0 / Call Hold)
Environment: windows xp or windows 2000 Environment
Microsoft Internet explorer 6.0
Assumptions: The network connection must be fast enough and it should not be congested otherwise it
might effect the Testing process.
Pre-Requisite: The call between the caller and the callee should be established.
Test Case Description
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.
Input Parameters Expected Output Actual Output Test Conformance Status
Possible Reason(s) in case of failure
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.
Passed for all valid input data cases
Passed for all in valid input data cases
No Test Case failed!!! No failure occurred.
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].