sip ppt final 1

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    Rajarshi Shahu College, Pune

    Department of Information Technology

    Presentation on

    By

    Ms. Shriya Arvind Jadhav

    Ms. Nikheta Dattatray Godse

    Ms. Sarika Vidhyadhar Gawade

    Guide

    Prof. Mrs. D.H.Patil

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    Contents

    Introduction of system

    Literature Survey

    System Architecture

    System Description

    System Requirements

    Modules Descriptions

    Project Flow

    Conclusions

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    Introduction

    Providing mobileenvironment for VoIPcalls

    You can use SIPphones to call yourfriends and family atany geographicallocation

    Provide low rate callsor even free calls.

    Hi !

    Hello..

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    LiteratureSurvey

    For developing this project we have referred number of IEEE papersfor gaining better knowledge in this subject. Some of those papers

    are as follows:

    Call set-up time modeling for SIP-based stateless and

    stateful calls in Next Generation Networks

    Voice over Internet Protocol on Mobile Devices

    Efficient User Controlled Inter-Domain SIP MobilityAuthentication, Registration, and Call Routing

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    SYSTEM ARCHITECTURE

    .

    SIP PROXY

    Media

    Signaling

    Registration100.101.102.103

    SIP User Agent SIP User Agent

    Registration100.101.102.103

    Signaling

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    What is IP Telephony?

    Internet Telephony (Voice over IP) essentiallymeans a voice message transmitted using theInternet Protocol

    Basically this means sending voice information indigital form in discrete packets as opposed to thetraditional circuit-committed protocols of the publicswitched telephone network.

    PBX: Private Branch Exchangeis a telephonesystem within an enterprise that switches callsbetween enterprise users

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    IP Telephony v/s PSTNPSTN : Static Switching

    IP Telephony : Dynamic Routing

    SERVER

    Telephone central office

    Internet

    Call connectedCalling

    CallingCall connected

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    What is SIP and Why SIP?

    In the world of VoIP, SIP is a call setup protocol thatoperates at the application layer

    SIP is the core protocol for initiating, managing andterminating sessions in the Internet

    These sessions may be text, voice, video or a combination

    of these

    SIP sessions involve one or more participants and can use

    unicast or multicast communication

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    SIP Entities

    User Agent

    User Agent Client (UAC)a client application that

    initiatesSIP requests.

    User Agent Server (UAS)

    a server application thatcontactsthe user when a SIP request is received and

    that returns a response on behalf of the user.

    Proxy Server

    Redirect Server

    Registrar

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    SIPMessageThere are two types of SIP messages:

    1]Request methods

    REGISTER : Used by a client to register with a SIP server

    INVITE : Initiates a call

    ACK : Confirms a final response for INVITE

    CANCEL : Used to cancel a pending request BYE : Terminates a call

    OPTIONS : Used to query a server

    2]Response code

    1xx : searching, ringing 2xx : success

    3xx : forwarding

    4xx : client mistakes

    5xx : server failures

    6xx : busy, refuse, not available anywhere

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    Architecture Diagram

    .

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    SIP Flows - Basic

    Talking Talking

    User A User B

    180 : Ringing

    200 OK : Answers the call

    ACK

    INVITE

    200 OK

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    SIP Flows Via ProxyThe SIP proxy only participates in the SIP user authentication and

    messages, once the call is set up, the phones send their voice traffic

    directly to each other without involving the proxy

    INVITETrying

    180- - Ringing 180 - Ringing

    200 - OK

    ACK

    Hangs up

    200 - OK

    User A User BMIT.EDUProxy

    Talking Talking

    INVITE

    200 OK : Answers the

    call

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    SystemRequirements

    1) S/W Requirements

    Operating System

    Microsoft Windows XP.

    miniSipServer

    2) Programming Languages

    Dot Net

    3) Hardware Requirements

    Processor: P4/Dual core RAM: 64MB

    Hard Disk: 10 GB

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    MainModule

    Basic Two Modules

    Server Side - miniSIPserver for routingcalls

    Client Side - For making calls andreceiving calls

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    Server Settings

    The software works as a fully featured telephone switch connecting tophone lines and extensions using state-of-the-art VoIP technology.

    Offering all the normal features of a traditional PBX

    Works with Any Standard Soft Phone

    Support static as well as dynamic IP

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    Add new usersThe procedure of adding new user to the network is

    not a tedious job from this application.Thus provides,

    Easy adding user

    Easy to modify the current user

    Easy deletion of existing user

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    Project FlowSTEP 1]

    SIP account setting

    USERNAME

    PASSWORD

    DOMAIN NAME

    SERVER IP

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    Local User StatusAt the server side, it shows the status of the registered users.

    Here the BLUE color icon shows the active users i.e.

    users that are online at that instant.

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    .

    Step 2]

    For caller : Dial user number

    Make call

    Hang-up the call after

    connection done

    Line status :

    Line1- Dialing /Trying, forestablishing the call

    Line1-Ringing,indicating call got

    established

    Line1- Connected, indicates that

    the call got connected with end

    user

    Line1-Busy,line is busy with other

    end user on call

    Line2-Free,this line can be use to

    make or accept call.

    .

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    .

    Step 3]

    For callee:

    When the caller makes a call, on the callee side the

    callee gets this window having the following

    options:

    Accept the call

    Reject call

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    Advantages and Disadvantages

    1) Advantage

    Reduce long distance calling cost.

    Free calls for SIP to SIP user agents.

    Portable

    Clear voice quality

    2) Disadvantage This application is expensive to use

    No facility to make Emergency Calls

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    Applications

    Setting up a Private LAN in office

    Voice Mail Service

    Ringing Group Service

    In VPN help to make call within the network

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    Future Scope

    Compatibility with any mobile phones

    Developing more APIs using SIP like

    Virtual conference room service

    Voice message

    IP Billing

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    Conclusion.

    Thus this application will help people to

    communicate staying in any Geographical

    location at low cost.

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    ReferencesTechnical Papers.

    [1] Pirhadi, M. Hemami, S.M.S. Tabrizipoor, A.I. Sci. & Res. Branch, Islamic Azad Univ.,

    TehranAdvancedCommunication Technology, in Proc. 11th IEEE Workshop on Applicationsof

    IP Telephony, WashingtonDCpp,2008,Page(s): 34-34

    [2] Khoury, J.S. Jerez, H.N. Abdallah, C.T. Univ. of New Mexico, Albuquerque Mobile and

    Ubiquitous Systems: Networking & Services in Proc. 4th IEEE Workshop on Applicationsof

    Networking, Washington DC pp,2005,Pages 333-333

    Books.

    1. Session Initiation Protocol-Controlling Convergent Networks by Travis Russell

    2. Guide to Telecommunications Technology by Tamara Dean

    3. Internet Communications Using SIP: Delivering VoIP and Multimedia Services with Session

    Initiation Protocol (Networking Council) by Henry Sinnreich and Alan B. Johnston

    4. C# Programming Languages by Anders Hejlsberg Mads Torgersen, Scott Wiltamuth, and Peter

    Golde.

    Some online links.

    www.google.com

    www.sipcenter.com

    www.myvoipapp.com/minisipserver

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    Thank You