spbx test plan

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Project Code: SPBX

Project Advisor : Aftab Alam

Project Team:

Umair Ashraf 03-1853 (Team Lead)

Imran Bashir 02-1658 Khadija Akram 04-0080

Submission Date:23-11-2007

SPBX Test Cases Specifications Document Version 1.0

November 23, 2007 of 41

Document Information

Category Information

Customer FAST-NU

Project Soft PBX

Document Test Cases Specifications

Document Version 1.0

Identifier SPBX

Status Draft

Author(s) Umair Ashraf , Khadija Akram, Imran Bashir

Approver(s) Aftab Alam

Issue Date November 23, 2007

Document Location

Distribution Advisors (Aftab Alam , Asad Gill)

Definition of Terms, Acronyms and

Abbreviations

Term Description

TCS Test Cases 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

SPBX Test Cases Specifications Document Version 1.0

November 23, 2007 of 41

Table of Contents

INTRODUCTION......................................................................................................... 4

Purpose of Document .................................................................................................................................................................................. 5

Scope of Testing ............................................................................................................................................................................................ 5

TEST CASE DESIGN AND DESCRIPTION ..................................................... 6

Dial and place call Test Case .................................................................................................................................................................... 6

Transfer Call Test Case ............................................................................................................................................................................. 10

Receive Call Test Case .............................................................................................................................................................................. 14

Missed Call Test Case ................................................................................................................................................................................ 15

Hold calls Test Case ................................................................................................................................................................................... 16

Call Unhold Test Case: ...................................................................................................................................................................................... 18

GUI NO 11 Successful scenario: ....................................................................................................................................................................... 18

Phonebook Test Case ................................................................................................................................................................................ 19

Call Conference Test Case ....................................................................................................................................................................... 21

Generate Report Test Case ..................................................................................................................................................................... 23

Call record Test Case ................................................................................................................................................................................. 24

Terminating the call Test Case ............................................................................................................................................................... 26

Admin login Test Case ............................................................................................................................................................................... 28

Add caller id Test Case ...................................................................................................................................................................................... 31

Change caller id Test Case ...................................................................................................................................................................... 32

Delete caller id Test Case ........................................................................................................................................................................ 33

Change hold on music play Test Case ............................................................................................................................................................... 34

Register User (Caller) Test Case ........................................................................................................................................................... 35

EXIT CRITERIA ........................................................................................................ 38

SIGNOFF ...................................................................................................................... 38

REFERENCES .............................................................................................................. 38

APPENDICES ................................................................................................................. 39

Real-time Transport Protocol (RTP) ..................................................................................................................................................... 39

Session Initiation Protocol (SIP) ............................................................................................................................................................ 40

Jitter 41

G.729 Codec ................................................................................................................................................................................................. 41

SPBX Test Cases Document Version 1.0

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Section 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.

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 call-routing 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 as our VoIP protocol. Our

PBX would be acting as both the registrar and as a gateway between the soft

phones.


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Purpose of Document

The purpose is to completely document all the testing activates carried out by the testing and the development team on the SPBX system. These

documents would help the maintenance team to completely track all the testing and Quality assurance activities carried out on the project during the

project development life cycle. The document is meant for all the end users including the development team, maintenance team as well as the testing

team. The document ensures the client of the software that all the standard quality assurance and testing activates have been done and carried out on

the project before handing it over to the client.

Scope of Testing

The testing process involves the following main parts:

i) Examining the system against the functional and non functional requirements provided in the functional specification document.

ii) Completely checking the conformance of the system to its

requirements and user expectations according to the use cases given in the functional specification document.

iii) Executing an implementation of different modules of SPBX with test

data and examining the output of the System and its operational behavior that it is performing as required or expected.

iv) Different testing techniques have been carried out like Black box testing, white box testing and the acceptance test.


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Test Case Design and Description

Dial and place call Test Case

GUI No 1 (Soft phone Interface)


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(Successful scenario-1)

In this scenario, the calle is idle and can view the incoming call. The GUI‘s related to this

scenario are given below.

GUI NO 2 FOR CALLER

GUI NO 3 FOR CALLE


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(Successful scenario-2)

In this scenario, the calle is not idle and can’t view the incoming call.The GUI‘s related to this scenario are given

below.

GUI NO 4 FOR CALLER In this case, the display screen on the soft phone will display the busy status.

GUI NO 5 (Exception Interface)

The GUI displaying the exception message in this format will be shown to the caller.


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GUI NO 6 (Error)

Test Case ID: 1 (Dial call and place call)

QA Test Engineer Khadija Akram Reviewed By: Testing Team lead (Umair Ashraf )

Test case Version: 1.0

Test Execution Date: 21-11-2007

Use Case Reference(s) Use case id 1 (Dial call)

GUI Reference(s)

GUI No 1 (soft phone interface) GUI No 2 (caller) GUI no 3 (calle) GUI No 4 (caller) GUI no 5 (Exception interface) GUI 6(error)

Objective

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.

Product / Ver/ Module Soft phone System Ver 1.0 /dial call Function)

Environment: windows xp or windows 2000 Environment Microsoft Internet explorer 6.0


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Assumptions: The network connection must be fast enough and it should not be congested otherwise it might effect the Testing process.

Pre-Requisite: The caller and the calle should both be registered to the ip based pbx .

Test Case Description

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.

Input Parameters Expected Output Actual Output Test Conformance Status

Possible Reason(s) in case of failure

(Valid Data)

The dial number will

consist of four digits

and should be valid.

4 digit dial number

1890

Invalid inputs

Greater than 4 digit

Dial number

90898

Less than 4 digit

Number.

109

Negative no

-190

Special characters

@098

Strings and digits

mixed

146l

For all valid data of

equivalence classes

Output of the system

should be successful

GUI no 2,3 and 4 will

be displayed to the

caller and the calle.

For all invalid data

equivalence classes.

Dial error and system

will display GUI no 6


equivalence classes

Successful

Validation of the

number and display

of GUI no 2 and 3 or

4 to the caller and the

calle will occur.

For all invalid data of




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)

Transfer Call Test Case GUI NO 7 (transfer dialog Interface)


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GUI NO 8 (successful)

GUI NO 9 (Error)

The number provided was not valid


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Test Case ID: 2 (Transfer call)




Use Case Reference(s) Use case id 6(Transfer call )

GUI Reference(s)

GUI 5 GUI 7 GUI 8 GUI 9

Objective

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.

Product / Ver/ Module Soft phone System Ver 1.0 transfer call Function)


Assumptions: The network connection must be FAST enough and it should not be congested otherwise it might effect the Testing process.

Pre-Requisite: The user should be registered with the pbx.


Testing of the transfer call functionality of the soft phone system. It involves testing the behavior of the transfer call functionality of the system by checking all possible outputs with respect to the possible inputs given to it.


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(Valid Data)




4 digit dial number

1789

Invalid inputs


Dial number

90876

Less than 4 digit

Number

987

Negative no

-190

Special characters

@098

Strings and digits

mixed

146l


equivalence classes



GUI no 8 will be

displayed on the

screen.






equivalence classes

Successful

Validation of the

number and display

of GUI 8 will be

displayed on the

screen.






input data cases


input data cases


No failure occurred

.The only possible

case was that the


system responded in


GUI (GUI no 5

exceptional GUI)


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Receive Call Test Case GUI NO 2 (Interface of receive call)

GUI NO 3 (In case the calle accepts the call) GUI NO 10 (In case the calle ignores the call, the updation in the table given below table will occur)

Test Case ID: 3 (Receive call)




Use Case Reference(s) Use case id 2 (Receive call )

GUI Reference(s)

GUI 2 GUI 3 GUI 10

Objective

To Completely Verify and Validate the Receive call functionality of the soft phone System.This test case will completely check the behavior of Receive call operation for various inputs in the soft phone system.

Product / Ver/ Module Soft phone System Ver 1.0 t Receive Call Function)



Pre-Requisite: The call between the caller or calle is established.


Testing of the receive call functionality of the soft phone system. It involves testing the behavior of the receive call functionality of the system by checking all possible

outputs with respect to the possible inputs given to it.



The calle selects the

accept button.


ignore button.



GUI no 2 will be

displayed on the

screen.



GUI no 10 will be

displayed on the

screen.

GUI 2 will be

displayed on the

screen.



GUI no 10 will be

displayed on the

screen.

Passed for all inputs

No Test Case failed

!!!


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Missed Call Test Case GUI NO 1 (Interface)

GUI NO 10(successfull)

In case the calle ignores the call and clicks on the missed call button, he will be able to view

the missed calls log in this table.

GUI NO 10

In case the calle does not ignore the call and clicks on the missed call button, he will view the original table.

Test Case ID: 4 (Missed call)




Use Case Reference(s) Use case id 3 (Missed call )

GUI Reference(s) GUI No 1 (soft phone interface)) GUI 10

Objective

To Completely Verify and Validate the missed call functionality of the soft phone System.This test case will completely check the behavior of missed call operation for various inputs in the soft phone system.

Product / Ver/ Module Soft phone System Ver 1.0 missed call Function)




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Pre-Requisite: The user should be a registered with the pbx.


Testing of the missed call functionality of the soft phone system. It involves testing the behavior of the missed call functionality of the system by checking all possible outputs with respect to the possible inputs given to it.




accept button.


ignore button.



GUI no 10 will be

displayed on the

screen.

Successful

Validation of the

number and display

of GUI 10 will be

displayed on the

screen.

Passes for all inputs


Hold calls Test Case GUI NO 11 (Successful)

The user clicks on the call hold button.


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Test Case ID: 5 (Hold call)




Use Case Reference(s) Use case id 10 (Hold call )

GUI Reference(s) GUI 11

Objective

To Completely Verify and Validate the hold call functionality of the soft phone System.This test case will completely check the behavior of hold call operation for various inputs in the soft phone system.

Product / Ver/ Module Soft phone System Ver 1.0 hold call Function)



Pre-Requisite: The call between the caller and calle is established.


Testing of the hold call functionality of the soft phone system. It involves testing the behavior of the hold call functionality of the system by checking all possible outputs

with respect to the possible inputs given to it.




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hold call button.



GUI no 11 will be

displayed on the

screen.

Successful

Validation of the

number and display

of GUI 11 will be

displayed on the

screen.



Call Unhold Test Case: GUI NO 11 Successful scenario:

Test Case ID: 6 (Unhold call)




Use Case Reference(s) Use case id 11 (Hold call )

GUI Reference(s) GUI 2

Objective

To Completely Verify and Validate the unhold call functionality of the soft phone System.This test case will completely check the behavior of unhold call operation for various inputs in the soft phone system.

Product / Ver/ Module Soft phone System Ver 1.0 unhold call Function)



Pre-Requisite: The call between the caller or calle is established.


Testing of the unhold call functionality of the soft phone system. It involves testing the behavior of the unhold call functionality of the system by checking all possible





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un hold call button.



GUI no 2 will be

displayed on the

screen.

Successful

Validation of the

number and display

of GUI 2 will be

displayed on the

screen.



Phonebook Test Case GUI NO 12(Successful)


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Test Case ID: 7 (Phone Book)




Use Case Reference(s) Use case id 9 (Phone Book Service )

GUI Reference(s) GUI No 1 (soft phone interface)) GUI 12

Objective

To Completely Verify and Validate the phone book functionality of the soft phone System.This test case will completely check the behavior of phone book operation for various inputs in the soft phone system.

Product / Ver/ Module Soft phone System Ver 1.0 hold call Function)



Pre-Requisite: The user should enter be registered to the ip pbx.


Testing of the phone book functionality of the soft phone system. It involves testing the behavior of the phone book functionality of the system by checking all possible





phone book button.



GUI no 12 will be

displayed on the

screen.

Successful

Validation of the

number and display

of GUI 12 will be

displayed on the

screen.




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Call Conference Test Case GUI No 1 (Soft Phone Interface) GUI No 13 (Successful) The number provided by the user was valid

GUI No 14 (Error)

The number provided was not valid.


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Test Case ID: 8 (Call Conference)




Use Case Reference(s) Use case id 7 (Call Conference)

GUI Reference(s)

GUI No 1 (soft phone interface) GUI no 5 (Exception interface) GUI 13 GUI 14

Objective

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.

Product / Ver/ Module Soft phone System Ver 1.0 /call conference Function)



Pre-Requisite: The user should be registered to the PBX


Testing of the call conference functionality 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.



(Valid Data)




4 digit dial number

1567

Invalid inputs


Dial number

89076

Less than 4 digit

Number.

198

Negative no

-190

Special characters


equivalence classes



GUI no 13 will be

displayed to the user.



GUI no 14 will be



equivalence classes

Successful

Validation of the

number and display

of GUI no 13 will be




input data cases


input data cases

No Test Case failed

!!!

No failure occurred

.The only possible

case was that the


system responded in


GUI (GUI no 5

exceptional GUI)


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@098

Strings and digits

mixed

146l



will display GUI no

14

Generate Report Test Case GUI NO 15 (successful scenario)


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Test Case ID: 9 (Generate Report)




Use Case Reference(s) Use case id 5 (Generate Report Service )

GUI Reference(s) GUI No 1 (soft phone interface) GUI No 1

Objective

To Completely Verify and Validate the generate report functionality of the soft phone System.This test case will completely check the behavior of generate report operation for various inputs in the soft phone system.

Product / Ver/ Module Soft phone System Ver 1.0 generate report Function)





Testing of the generate report functionality of the soft phone system. It involves testing the behavior of the generate report functionality of the system by checking all possible outputs with respect to the possible inputs given to it.




generate report

button.



GUI no15 will be

displayed on the

screen.

Successful

Validation of the

number and display

of GUI 15 will be

displayed on the

screen.


No Test Case failed

!!!

Call record Test Case


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GUI NO 16 (successful scenario)

Test Case ID: 10 (Call Record)




Use Case Reference(s) Use case id 8 (Call Record Service )

GUI Reference(s) GUI No 1 (soft phone interface))

Objective

To Completely Verify and Validate the call record functionality of the soft phone System.This test case will completely check the behavior of call record operation for various inputs in the soft phone system.

Product / Ver/ Module Soft phone System Ver 1.0 call record Function)





Testing of the call record functionality of the system. It involves testing the behavior of the call record functionality of the system by checking all possible





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call record button.



GUI no16 will be

displayed on the

screen.

Successful

Validation of the

number and display

of GUI 16 will be

displayed on the

screen.


No Test Case failed!!

Terminating the call Test Case GUI NO 17 (successful scenario)


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Test Case ID: 11 (Terminate Call)




Use Case Reference(s) Use case id 12(Terminate Call Service )

GUI Reference(s) GUI No 17

Objective

To Completely Verify and Validate the terminate call functionality of the soft phone System.This test case will completely check the behavior of terminate call operation for various inputs in the soft phone system.

Product / Ver/ Module Soft phone System Ver 1.0 terminate call Function)



Pre-Requisite: The call between the caller and the calle should be established.


Testing of the terminate call functionality of the system. It involves testing the behavior of the terminate call functionality of the system by checking all possible outputs with respect to the possible inputs given to it.




terminate call button.



GUI no17 will be

displayed on the

screen.

Successful

Validation of the

number and display

of GUI 17 will be

displayed on the

screen.




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PBX: GUI NO 1(PBX interface)

Admin login Test Case GUI NO 2(admin password interface)


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GUI NO 3 (Successful)

GUI NO 1(error)


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Test Case ID: 12(admin login)




Use Case Reference(s) Use case id 18 (Admin login)

GUI Reference(s)

GUI No 1 (PBXinterface) GUI No 2 GUI No 3

Objective

To Completely Verify and Validate the admin login functionality of the soft phone System.This test case will completely check the behavior of admin login operation for various inputs in the PBX system.

Product / Ver/ Module Soft phone System Ver 1.0 / admin password Function)



Pre-Requisite: Admin should both be registered to the ip based pbx..


Testing of the admin login functionality of the PBX system. It involves testing the behavior of the admin login functionality of the system by checking all possible outputs with respect to the possible inputs given to it.



(Valid Data)

The password will

consist of six digits


457890

Invalid inputs


password

9865098

Less than 6 digit

Number.

13456

Special character

145@90

Characters and digits

mixed

1456ah


equivalence classes



GUI no 3 will be

displayed to the

admin.






equivalence classes

Successful

Validation of the

number and display

of GUI no 3 will be

displayed to the

admin.



GUI 1

Will be displayed to

the admin.


input data cases


input data cases


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Add caller id Test Case

Test Case ID: 13 (add caller id)




Use Case Reference(s) Use case id 14(add caller id )

GUI Reference(s) GUINO 1 GUINO 3

Objective

To Completely Verify and Validate the add caller id functionality of the soft phone System.This test case will completely check the behavior of add caller id operation for various inputs in the soft phone system.

Product / Ver/ Module Soft phone System Ver 1.0 add caller id Function)



Pre-Requisite: The admin must be logged in.


Testing of the add caller id functionality of the system. It involves testing the behavior of the add caller id functionality of the system by checking all possible outputs with respect to the possible inputs given to it.



The admin selects the

add button.



GUI no3 will be

displayed on the

screen.

Successful

Validation of the

number and display

of GUI 3 will be

displayed on the

screen.




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Change caller id Test Case

Test Case ID: 14 (change caller id)




Use Case Reference(s) Use case id 15(change caller id )


Objective

To Completely Verify and Validate the change caller id functionality of the soft phone System.This test case will completely check the behavior of change caller id operation for various inputs in the soft phone system.

Product / Ver/ Module Soft phone System Ver 1.0 add caller id Function)





Testing of the change caller id functionality of the system. It involves testing the behavior of the change caller id functionality of the system by checking all possible outputs with respect to the possible inputs given to it.




reset button.



GUI 3 no will be

displayed on the

screen.

Successful

Validation of the

number and display

of GUI 3 will be

displayed on the

screen.




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Delete caller id Test Case

Test Case ID: 15 (delete caller id)




Use Case Reference(s) Use case id 16(delete caller id )


Objective

To Completely Verify and Validate the delete caller id functionality of the soft phone System.This test case will completely check the behavior of delete caller id operation for various inputs in the soft phone system.

Product / Ver/ Module Soft phone System Ver 1.0 delete caller id Function)





Testing of the delete caller id functionality of the system. It involves testing the behavior of the delete caller id functionality of the system by checking all possible outputs with respect to the possible inputs given to it.




delete button.



GUI 3 no will be

displayed on the

screen.

Successful

Validation of the

number and display

of GUI 3 will be

displayed on the

screen.




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Change hold on music play Test Case

Test Case ID: 16 (change hold on music)




Use Case Reference(s) Use case id 17(change hold on music )


Objective

To Completely Verify and Validate the change hold on music functionality of the soft phone System.This test case will completely check the behavior of change hold on music operation for various inputs in the soft phone system.

Product / Ver/ Module Soft phone System Ver 1.0 change hold on music Function)





Testing of the change hold on music functionality of the system. It involves testing the behavior of the change hold on music functionality of the system by checking all possible outputs with respect to the possible inputs given to it.




change hold on

music button.



GUI no 3 will be

displayed on the

screen.

Successful

Validation of the

number and display

of GUI 3 will be

displayed on the

screen.


No Test Case failed

!!!


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Register User (Caller) Test Case

GUI NO 4(successful scenario)

When the user provides the correct ip address and caller id.

GUI NO 5

When the user provides the wrong ip address or the caller id


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Test Case ID: 17




Use Case Reference(s) Use case id 12

GUI Reference(s) GUINO4 GUINO5

Objective

To Completely Verify and Validate the register user functionality of the soft phone System. This test case will completely check the behavior of register user operation for various inputs in the soft phone system.

Product / Ver/ Module Soft phone System Ver 1.0 register user



Pre-Requisite: The caller id for valid input must be registered in the pbx.


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.


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For valid inputs

Caller id should be 4

digit number.

4567

For invalid inputs

Caller id shouldn’t

be negative.

-091

123456

@189

13j



GUI no 4 will be

displayed on the

screen.



GUI no 5 will be

displayed on the

screen.

Successful

Validation of the

number and display

of GUI 4 will be

displayed on the

screen.




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Exit Criteria

:

No high priority problems are open

All functions identified in the requirement document are present and

working

No input/output error found in all the modules during the testing

After testing, all the modules of the system are verified and validated with

respect to the requirement specification document.

All unit test applied on the modules are passed.

The system and all of its modules have clearly passed the acceptance test

and it should be handed over to the client.

Signoff

Project Advisor:

Team Lead

(Umair Ashraf)……………………………………

Development Team:

Imran Bashir…………………………………….

Khadija Akram …………………………………

References Ref. No. Document Title Date of Release/ Publication

Phase 1 Requirement Specifications ver 1.0

14th September 2007

Phase 2 Functional Specifications ver 1.1

19th October 2007

Phase 3 Design Document (class Diagrams ) ver 1.0

12,November 2007

Phase 4

Design Document ver 1.3 (Class Diagram + Sequence Diagrams Call flow diagrams )

19th November 2007


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Other References

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

How stuff works platform Wekipedia.org

Asterisk the Open Source Telephony Platform www.asterisk.org

Appendices

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.

http://www.asterisk.org/


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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.

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.


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Using proxy mode, SIP clients send requests to the proxy and the proxy

either handles requests or forwards them on to other SIP servers.

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.

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.

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.

http://en.wikipedia.org/wiki/Bit_rate

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