O V

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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.

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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.

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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

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Signaling Systems

PSTN IP Telephone VOIP PSTN GATEWAY Analog Telephone

VOIP network Signaling

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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

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SIP format Ipv4 address

sip:bob@201.23.45.78

sip:bob@fhda.edu

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

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SIP components

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

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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

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SIP agent communication

UAC Signaling msg UAS UAS UAC

SIP user agent

SIP user agent

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SIP Servers

Proxy Redirect Registrar Location UAC UAC

UAS
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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

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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

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SIP request Messages

INVITE

ACK

BYE

OPTIONS

CANCEL

REGISTER

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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

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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
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SIP response Messages

1XX

2XX

3XX

4XX

5XX

6XX

Response msg header format

SIP Version Status Code

Reason phase

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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
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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
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OK

LS

Location query INVITE

200 ACK OK PS

Location response

UA1

INVITE
200 ACK OK UA2

SIP Call Setup Using Proxy Server

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SIP registration
UA RS

REGIS TER

200 OK

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SIP call setup using redirect server

LS

Lo ca tio n
UA1

INVITE
302 MOVED

RS

ACK

0O AC K

IN V 20 ITE

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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

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SIP Network Architecture

SIP User Agent (Called)

Hardware Interface
ATA

A-D converter Connects standard phone to the internet.

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Hardware Interface

IP PHONE

Gets connected directly to the router Has NIC card

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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

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Computer-to-computer

DSL Modem DSL Modem

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Phone to phone

AT A

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IP phone

COMPUTER TO PHONE

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IP telephony

(Voice Over Internet Protocol)

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

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IP Protocol Stack

Application Layer Transport Layer Network Layer Data Link Layer Physical Layer
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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

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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.

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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

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Interleaving

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Thanks

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