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:
[email protected]sip:[email protected]
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
