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    VOIP Honey+Harish

    Uploaded by

    Honey Adlakha
    The document discusses Voice over Internet Protocol (VoIP) technology. It describes how VoIP works by breaking voice signals into digital packets which are transmitted over the Internet using protocols like SIP, RTP and UDP. It also discusses VoIP hardware interfaces, protocols, call setup processes, and compares VoIP to traditional PSTN telephone networks. Key advantages of VoIP mentioned include lower connection costs and added features like caller ID.

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    Attribution Non-Commercial (BY-NC)
    Download as PPTX, PDF, TXT or read online from Scribd

    VOIP Honey+Harish

    Uploaded by

    Honey Adlakha
    76 views41 pages
    The document discusses Voice over Internet Protocol (VoIP) technology. It describes how VoIP works by breaking voice signals into digital packets which are transmitted over the Internet using protocols like SIP, RTP and UDP. It also discusses VoIP hardware interfaces, protocols, call setup processes, and compares VoIP to traditional PSTN telephone networks. Key advantages of VoIP mentioned include lower connection costs and added features like caller ID.

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    The document discusses Voice over Internet Protocol (VoIP) technology. It describes how VoIP works by breaking voice signals into digital packets which are transmitted over the Internet using protocols like SIP, RTP and UDP. It also discusses VoIP hardware interfaces, protocols, call setup processes, and compares VoIP to traditional PSTN telephone networks. Key advantages of VoIP mentioned include lower connection costs and added features like caller ID.

    Copyright:

    Attribution Non-Commercial (BY-NC)
    Download as PPTX, PDF, TXT or read online from Scribd
    Download as pptx, pdf, or txt
    76 views41 pages

    VOIP Honey+Harish

    Uploaded by

    Honey Adlakha
    The document discusses Voice over Internet Protocol (VoIP) technology. It describes how VoIP works by breaking voice signals into digital packets which are transmitted over the Internet using protocols like SIP, RTP and UDP. It also discusses VoIP hardware interfaces, protocols, call setup processes, and compares VoIP to traditional PSTN telephone networks. Key advantages of VoIP mentioned include lower connection costs and added features like caller ID.

    Copyright:

    Attribution Non-Commercial (BY-NC)
    Download as PPTX, PDF, TXT or read online from Scribd
    Download as pptx, pdf, or txt
    of 41

    O V

    2 /1 3 /1 4

    KH A le ty L s D tle A i bt u r s EY TA te P as N M O t di H GU e to - SH k lic : C Y B RI A H

    P I

    nd a

    VOIP

    Voice & video transmission over IP network. Convergence around Internet telephone encompassing fax, data, video, voice. Voice/video broken into chunks, transmitted using Packet Switching and reassembled at the Rx.

    2 /1 3 /1 4

    Why VOIP??

    Cheaper connection(economically) Call from anywhere Less maintenance Improved and expanded features eg. caller id, call waiting, call transfer, repeat dial, return call, call forwarding, three way calling.

    2 /1 3 /1 4

    VoIP connection

    Analog voice digital signals (bits) compression of bits (MP-MLQ,ACELP) Insert voice packets in data packets using a realtime transport protocol (RTP over UDP over IP) Use signaling protocols to call users

    At Rx disassemble packets, extract 2 /1 data, convert them to analog voice 3 /1 4 signals

    Call Signaling Protocols

    How does the communication path get established ? Signaling: defined as procedures undertaken to setup, manage, and terminate session btn two end points Protocols: SIP, H.323, MGCP, MEGACO/H.248

    2 /1 3 /1 4

    Signaling Systems

    PSTN IP Telephone VOIP PSTN GATEWAY Analog Telephone

    VOIP network Signaling


    2 /1 3 /1 4

    Inter exchange signaling(SS7)

    Subscriber loop Signaling

    Session Initiation Protocol

    Developed by IETF Appl layer control protocol SIP establishes, manages, terminate sessions (multimedia conference, internet telephony, media distribution) Sender/receiver identified by IP address, email, telephone no. 2

    1 3/ 1 4/

    SIP format Ipv4 address


    sip:bob@201.23.45.78

    sip:bob@fhda.edu

    E-mail address
    sip:bob@408-864-8900

    2 /1 3 /1 4

    SIP components

    Caller: initiates conference Callee: caller invites callee Redirect Server: accepts SIP rqst, maps address, & returns address to client

    2 /1 3 /1 4

    Location Server: provides services to SIP Redirect or Proxy server regarding callees possible location(s) Registrar Server: accept REGISTER rqst User Agent Client: client appl that initiates SIP rqst User Agent Server: server appl that

    2 /1 3 /1 4

    SIP agent communication


    UAC Signaling msg UAS UAS UAC

    SIP user agent

    SIP user agent

    2 /1 3 /1 4

    SIP Servers

    Proxy Redirect Registrar Location UAC UAC

    UAS
    2 /1 3 /1 4

    SIP Agent and Server Communication

    UAS

    SIP Cont..

    SIP is text based protocol uses ISO 10646 character set in UTF-8 encoding Rqst msg has header and body Msg can be tx over UDP or TCP Header describes msg structure, callers capability, media type

    2 /1 3 /1 4

    Protocol header structure

    The protocol is composed of a start line, message header, an empty line and an optional message body Request packet header format:
    Method Request URI

    SIP version

    2 /1 3 /1 4

    SIP request Messages

    INVITE

    ACK

    BYE

    OPTIONS

    CANCEL

    REGISTER

    2 /1 3 /1 4

    Methods Command INVITE ACK BYE call Function Initiate Call Confirm final response Terminate and transfer

    CANCEL Cancel searches, pending rqst and ringing OPTIONS Querry a server about its 2 /1 capability 3 1

    4/

    Request URL A SIP URL or a general Uniform Resource Identifier, is the user or service to which this request is being addressed SIP version The SIP version being used; this should be version 2.0
    2 /1 3 /1 4

    SIP response Messages

    1XX

    2XX

    3XX

    4XX

    5XX

    6XX

    Response msg header format


    SIP Version Status Code

    Reason phase

    2 /1 3 /1 4

    SIP version
    The SIP version being used

    Status Code
    A 3-digit integer result code of the attempt to understand and satisfy the request, indicates the current condition of rqst
    2 Response Phase /1 3

    1 4/

    Response codes

    Response Code Prefix 1xx 2xx 3xx

    Function

    Informational (rqst received and processing is continuing) Success(ACK, action received, understood, accepted) Redirection(further action required to process the rqst) Client error(bad syntax in

    2 4xx /1 3 /1 4 rqst)

    UA1

    SIP Call Signaling

    UA2

    INVITE

    200 OK ACK

    BYE
    2 /1 3 /1 4

    OK

    LS

    Location query INVITE


    200 ACK OK PS

    Location response

    UA1

    INVITE
    200 ACK OK UA2

    SIP Call Setup Using Proxy Server


    2 /1 3 /1 4

    SIP registration
    UA RS

    REGIS TER

    200 OK

    2 /1 3 /1 4

    SIP call setup using redirect server


    LS

    Lo ca tio n
    UA1

    INVITE
    302 MOVED

    RS

    ACK

    0O AC K

    IN V 20 ITE

    2 /1 3 /1 4

    UA2

    Lo re cat sp io on n se

    qu er y

    SIP Redirect Server

    Location Server RS S

    Non-SIP Protocol Ti IP NETWORK S SIP Proxy S SIP Proxy S SIP Proxy

    2 SIP User Agent /1 3 (Calling) /1 4

    SIP Network Architecture

    SIP User Agent (Called)

    Hardware Interface
    ATA

    A-D converter Connects standard phone to the internet.

    2 /1 3 /1 4

    Hardware Interface

    IP PHONE

    Gets connected directly to the router Has NIC card

    2 /1 3 /1 4

    Flavors of VOIP!

    Computer-to-computer

    Cheapest in VoIP connection (No calling cost) Through Software support U need to have high speed internet connection ,Microphone and sound card

    2 /1 3 /1 4

    Computer-to-computer

    DSL Modem DSL Modem

    2 /1 3 /1 4

    Phone to phone

    AT A

    2 /1 3 /1 4

    IP phone

    COMPUTER TO PHONE

    2 /1 3 /1 4

    IP telephony

    (Voice Over Internet Protocol)


    k lic C to ed a tM i s er st u le tit b le ty s

    2 /1 3 /1 4

    IP Protocol Stack

    Application Layer Transport Layer Network Layer Data Link Layer Physical Layer
    2 /1 3 /1 4

    User requests a Web Page (e.g., clicks on a hyperlink) Client


    GET www.bits.edu/home.index HTTP User-Agent: Mozilla/10.0 .. Application Layer Data + TCP header Segment (Source Port no, Destination Port no, sequence no., acknowledgement no.) Segment + IP header Datagram Source & Destination IP address Datagram + header + Trailer Frames for Error detection & correction 2 /1 3 Data converted in binary form and /1 4

    Server

    HTTP 200 OK Server: Apache/1.3.0 data.data..data. Segment Application Layer Data Datagram Segment Frames Datagram Data in binary form

    VOIP
    APPLICATION RTP UDP IP DATA LINK PHYSICAL

    2 /1 3 /1 4

    RTP Packet

    Voice Source is PCM encoded at 64 kbps. Application collects encoded data every 20ms. So 1280 bits or 160 bytes in each chunk.

    Application precedes each 2 /1 3 /1 chunk of data with RTP 4

    RTP Header Fields


    Payload Type

    Sequence Number

    Time Stamp

    Other fields

    Payload Type field is 7 bits long so 2^7 or 128 different payloads can be supported by RTP. For an audio stream, it is used to indicate type of audio encoding (ex PCM, ADM) that is being used. Seq. field is 16 bits long and is used by receiver to detect packet loss. Timestamp field is 32 bits long. It reflects sampling instant of the first byte in RTP data packet. Used by receiver to remove packet jitter introduced in n/w & to provide sync. play out at receiver.

    2 /1 3 /1 4

    Sending Voice Packets using RTP over UDP over IP Caller (after channel setup)
    Voice RTP Packet RTP Packet + UDP header Segment (Source & Destination Port no) Segment + IP header Datagram Source & Destination IP address Datagram + header + trailer Frame Error detection & correction, framing, etc Data converted in binary form and 2 transmitted over transmission line. /1 3 /1 4
    Callee

    RTP Packet Voice Segment RTP Packet Datagram Segment Frame Datagram Data in binary form

    UDP as Transport Layer Protocol

    Lost Packet: lost in n/w or delay>400 ms. So no need of retransmission. UDP header 8 bytes, TCP header 20 bytes. Question: What about Lost Packets ?? Forward Error Control Interleaving

    2 /1 3 /1 4

    Interleaving

    2 /1 3 /1 4

    Thanks

    2 /1 3 /1 4

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