Transcript VOIP BSIT 2007-2011 bzupages

11/18/2009
Department of IT, Institute of Computing, BZU, Multan
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Voice Over IP
Group members
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BSIT07-15
BSIT07-22
BSIT07-01
Department of IT, Institute of Computing, BZU, Multan
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Muhammad Aatif Aneeq
Shah Rukh
Muhammad Wasif Laeeq
MUHAMMAD AATIF ANEEQ
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Department of IT, Institute of Computing, BZU, Multan
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BSIT07-15
Circuit Switched Network:
 In circuit switched networks, a circuit is
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established when data is needed to be
transferred & all the communication is done
through that circuit.
Packet Switched Network
 It is a switching network, in which data is broken
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down in small chunks (Packets) and is
transferred in form of packets. This data may
reach to the destination from different paths.
 Each packet finds its way using the information
it carries, such as the source and destination IP
addresses.
PSTN
 The public switched telephone network is the
 Example: PTCL Landline
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network of the world's public circuit-switched
telephone networks.
 Originally a network of fixed-line analog
telephone systems
PSTN
 PSTN lines come in two common standards
 Single Dedicated line
 Digital
 Multiple lines in one line
• e.g. T1 , E1
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 Analog
VoIP
 Voice over Internet Protocol (VoIP) is a general
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term for a family of transmission technologies for
delivery of voice communications over IP networks
such as the Internet or other packet-switched
networks.
IP telephony
Internet telephony
voice over broadband (VoBB)
broadband telephony
broadband phone
VOIP
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codec
 Voip will not be possible without
compression/decompression.
 Voice first encoded from Analog to digital
IP Packets and then decoded back to
analog at receiver end.
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 compressor-decompressor
 coder-decoder
Choice of codec
G.711 (PCM) : requires 64Kbps
G.729A : requires 8Kbps (16kbps including overheads)
Using G.729A.
16kbps * 30 = 480kbps
512kbits/second link is enough to carry 30 simultaneous voice
channels on
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Depends on requirements & equipment available…
Origination
Two Types:
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» PC based origination
» Phone based origination
Phone Based Origination
 DID (Direct Inward Dialing)
 Can setup your own DID’s or Purchase from
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other organizations…
 SIP Origination: Call is transferred to your SIP
address…
 didx.net provides cheap wholesale DID’s
Termination
 a gateway is used that takes calls off the
Internet and delivers to PSTN lines.
termination service providers…
 almvoip.com provides cheapest white label
termination for Pakistan…
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 Can also use termination service by other
What actually those service
Providers use?
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Digium Wildcard TE412P
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BSIT07-22
SHAH RUKH
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The equipments (for client)
ATA
Soft phone
IP Phone
Wi-Fi/WLAN phone
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ATA
Internet connection for use with VoIP.
 ATAs are sometimes referred to as VoIP
gateways.
Ordinary Phone  ATA  Ethernet  Router  Internet  Service Provider
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 Analog Telephone Adaptor
 converts analog signals to digital data
 allows to connect a standard phone to your
Linksys ATA
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Soft phone
 A soft phone is actually a software application
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that you install on your computer to create a
VoIP user interface. In order to use a soft
phone, you’ll need a headset and/or
microphone.
X-lite : SIP based free softphone
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IP phone
 An IP phone, or hard phone, is a self-contained
 IP Phone  Ethernet  Router  Internet  VOIP Service
Provider
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piece of equipment (that looks like a regular
phone) that can communicate directly via your
Internet connection.
Linksys SPA941 SIP VOIP
Phone
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Wi-Fi/WLAN phone
 Like IP phones, Wi-Fi/WLAN phones don’t
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require a computer or ATA to use VoIP. They
link directly to your IP Internet connection.
Unlike IP phones, they’re wireless and connect
to the Internet via a wireless base station.
Linksys WIP300 Wi-Fi IP Phone
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VOIP connecting directly
 It is also possible to bypass a VOIP Service
IP Phone Ethernet  Router  Internet 
Router  Ethernet  IP Phone
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Provider and directly connect to another VOIP
user. However, if the VOIP devices are behind
NAT routers, there may be problems with this
approach.
Benefits of VoIP
 Operational cost
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Routing phone calls over existing data networks to
avoid the need for separate voice and data
networks.
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Conference calling, IVR, call forwarding, automatic redial, and
caller ID features that traditional telecommunication companies
normally charge extra for are available free of charge from
open source VoIP implementations such as Asterisk or
FreeSWITCH
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VoIP can be a benefit for reducing communication and
infrastructure costs. Examples include:
Benefits of VoIP (cont.)
 Costs are lower, mainly because of the way
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Internet access is billed compared to regular
telephone calls.
 regular telephone calls are billed by the minute
or second,
 VoIP calls are billed per megabyte (MB).
Benefits of VoIP (cont.)
Increased Functionality
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Incoming phone calls are automatically routed to your VOIP phone
where ever you plug it into the network. Take your VOIP phone with
you on a trip, and anywhere you connect it to the Internet, you can
receive your incoming calls.
Protocols - the language of VOIP
 Many protocls…
 Most commonly used
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 H.323
 SIP
 IAX2 (Inter-asterisk exchange)
Session Initiation Protocol
 IETF-based
 Developed from work on multi-party conferences
and IMS)
 Huge amount of work extending the protocol
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 The protocol chosen for next generation mobile and fixed networks (3GPP
SIP Architecture
 SIP is used for
 SDP (attached to SIP messages) is used to negotiate the media for
the call
 RTP/RTCP carries the media directly between the endpoints
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 Registration and Call Routing
 Call Admission Control (performed by proxy)
 Call Establishment
SIP Terminology
 Endpoints are SIP User Agents (UA)
 Proxies forward requests and responses
 They cannot generate new requests
 Registrars are UAS that record the location of clients
 A Registrar is normally colocated with a proxy
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 User Agent Clients (UAC) send requests
 User Agent Servers (UAS) process requests and send responses
 Most endpoints are both UAC and UAS
SIP URI
 sip:user:password@host:port;uriparameters?headers
be passed in URI for security.
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 Password can be passed in URI but should not
Structure of a SIP message
 Request
sip:user@host
To: …, From: …, etc.
SDP offer
 Response
 Status Line
 Headers
 Body
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180 Ringing
To:…, From: …, etc.
SDP answer
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 Request URI
 Headers
 Body
SIP Request Commands
 REGISTER
 INVITE
 Used to invite another User agent to
communicate, and then establish a SIP session
between them.
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 Used when a user agent first goes online and
registers their SIP address and IP address with a
Registrar server.
MUHAMMAD WASIF LAEEQ
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Department of IT, Institute of Computing, BZU, Multan
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BSIT07-01
SIP Request Commands (cont.)
 ACK
 CANCEL
 Used to cancel a pending request without
terminating the session.
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 Used to accept a session and confirm reliable
message exchanges.
SIP Request Commands (cont.)
 BYE
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 Used to terminate the session.
 Either the user agent who initiated the session, or
the one being called can use the BYE command
at any time to terminate the session.
Registration of UAC with
Registrar
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Request and Response Made through
Proxy Server
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SIP Responses
 Informational (1xx)
 Success (2xx)
 The request was acknowledged and accepted.
 Redirection (3xx)
 The request can’t be completed and additional steps
are required (such as redirecting the user agent to
another IP address).
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 The request has been received and is being
processed.
SIP Responses
 Client error (4xx)
 Server error (5xx)
 Global failure (6xx)
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 The request contained errors, so the server can’t
process the request
Private Branch eXchange
 A telephone system within an enterprise that
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switches calls between enterprise users on local
lines
 Allowing all users to share a certain number of
external phone lines.
 The main purpose of a PBX is to save the cost
of requiring a line for each user to the telephone
company's central office
PBX
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PBX Features
Welcome Message
Voice Mail
IVR
Call Transfer
Conference Call
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Department of IT, Institute of Computing, BZU, Multan
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What is Asterisk™?
 Asterisk™ is a complete PBX in software. It runs on
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Linux, BSD and MacOSX and provides all of the
features you would expect from a PBX and more.
Asterisk does voice over IP in many protocols, and can
interoperate with almost all standards-based telephony
equipment using relatively inexpensive hardware.
 Development of Asterisk™ is governed by Digium.
Asterisk™ Architecture
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SIP Proxy
#1 INVITE
#3 INVITE
#2 100 Attempt
#4 180 Ringing
#5 180 Ringing
#6 200 OK
#7 200 OK
#8 SIP ACK
#9 Bi-directional RTP channel
#10 SIP BYE
#11 SIP 200 OK
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SIP
Proxy
Another Implementation of VOIP
 Using Jingle
Presence Protocol)
 XMPP Sponsors:
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 An extension of XMPP (eXtensible Messaging &
XMPP
 XML based, making it very easy to use and
extend
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<message to=‘[email protected]'
from=‘[email protected]' type='chat'>
<thread>thread1</thread>
<body>How's that presentation going?</body>
</message>
Jingle
 Google launched their XMPP network with voice
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File transfer
Screen sharing
Video
Whiteboard
Anything else that uses a lot of bandwidth or
that does streaming
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support, then joined the standards effort to
define Jingle.
Voovi using XMPP and Jingle
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Client can be downloaded from
http://voovi.org/client.rar
Voovi using XMPP and Jingle
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 Thanks

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