Transcript IP Telephony (VoIP)

IP Telephony (VoIP)
CSI4118
Fall 2005
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Introduction (1)
• A recent application of Internet technology – Voice over
IP (VoIP): Transmission of voice over Internet
• How VoIP works
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Continuously sample audio
Convert each sample to digital form
Send digitized stream across Internet in packets
Convert the stream back to analog for playback
• Why VoIP
– IP telephony is economic; High costs for traditional telephone
switching equipments.
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Introduction (2)
• Challenge
– Voice transmission delay
– Call setup: call establishment, call termination, etc.
– Backward compatibility with existing PSTN (Public
Switched Telephone Network)
• IP Telephony Standards:
– ITU (International Telecommunication Union) controls
telephony standards.
– IETF (Internet Engineering Task Force) controls
TCP/IP standards.
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Encoding, Transmission, &
Playback (1)
• Both groups agree on the basics for encoding
and transmission of audio:
– Audio is encoded using a well-known standard such
as Pulse Code Modulation (PCM).
– Audio is transferred using the Real-time Transport
Protocol (RTP).
– RTP message is encapsulated in a UDP datagram
that is further encapsulated in an IP datagram for
transmission.
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Encoding, Transmission, &
Playback (2)
• UDP is used for transport because
– lower overhead: audio must be played as it arrives.
– Playback cannot be stopped to wait for a
retransmitted packet.
• Two independent RTP sessions exist, because
an IP phone call involves transfer in two
directions
– IP phone acts as sender for outgoing data, and
– IP phone acts as receiver for incoming data.
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Signaling Systems & Protocols
• Main complexity of VoIP: Call setup and call
management.
• The process of establishing and terminating a call is
called Signaling.
– In traditional telephone system, signaling protocol is SS7
(signaling System 7).
– In VoIP, signaling protocols are:
• SIP (Session Initiation Protocol), by IETF
• H.323, by ITU
• Megaco & MGCP, jointly by IETF and IUT.
– VoIP signaling protocols should be able to interact with SS7.
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A Basic IP Telephone System
• The simplest IP telephone system uses two basic components:
- IP telephone: end device allowing humans to place and receive
calls.
- Media Gateway Controller: providing overall control and coordination
between IP phones; allowing a caller to locate a callee (e.g. call
forwarding)
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Interconnection with Others (1)
• IP telephone system needs to interoperate
with PSTN or another IP telephone
system.
• Two additional components needed for
such interconnection:
– Media Gateway
– Signaling Gateway
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Interconnection with Others (2)
• Media gateway: translates audio between IP
network and PSTN.
• Signaling Gateway: translates signaling
operations.
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Signaling Protocols
• Two major protocols: H.323, SIP
• H.323, invented by ITU, defines four elements
that comprising a signaling system:
– Terminal: IP phone
– Gatekeeper: provides location and signaling
functions; coordinates operation of Gateway.
– Gateway: used to interconnect IP telephone system
with PSTN, handling both signaling and media
translation.
– Multipoint Control Unit: provides services such as
multipoint conferencing.
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Signaling Protocols
• SIP: Session Initiation Protocol. Invented by
IETF.
• SIP defines three main elements that comprise a
signaling system:
– User Agent: IP phone or applications
– Location servers: stores information about user’s
location or IP address
– Support servers:
• Proxy Server: forwards requests from user agents to another
location.
• Redirect Server: provides an alternate called party’s location
for the user agent to contact.
• Registrar Server: receives user’s registration requests and
updates the database that location server consults.
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H.323 Characteristics
• H.323 consists of a set of protocols that work
together to handle all aspects of communication,
including:
– Transmission of a digital audio phone call
– Signaling to set up and manage phone call
– Allows transmission of video and data while a phone
call is in progress
– Sends binary message
– Incorporates protocols for security
– Uses a special hardware Multipoint Control Unit for
conferencing calls
– Defines servers for address resolution, authentication,
accounting, features, etc.
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H.323 Layering
• H.323 uses both UDP and TCP over IP.
– Audio travels over UDP
– Data travels over TCP
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SIP Characteristics
• Operates at the application layer.
• Encompasses all aspects of signaling, e.g. location of
called party, ringing a phone, accepting a call, and
terminating a call.
• Provides services such as call forwarding.
• Relies on multicast for conference calls.
• Allows two sides to negotiate capabilities and choose the
media and parameters to be used.
• SIP URI is similar to email address. (with prefix “sip:”)
E.g.
sip:[email protected]
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SIP Methods
• Six basic message types, known as
methods:
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An Example SIP Session
• User agent A contacts DNS
server to map domain name in
SIP request to IP address.
• User agent A sends a INVITE
message to proxy server that
uses location server to find the
location of user agent B.
• Call is established between A
and B. Then media session
begins.
• Finally, B terminates the call by
sending a BYE request.
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Telephone Number Mapping &
Routing (1)
• How should users be named?
– PSTN follows ITU standard E.164 for phone numbers. E.g. 1613-123-4567
– SIP uses IP addresses. E.g. sip:[email protected]
• In an integrated network (PSTN + IP), two problems
defined:
– Locate a user
– Find a efficient route to the user
• IETF proposed two protocols:
– ENUM: E.164 NUMbers
– TRIP: Telephone Routing over IP
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Telephone Number Mapping &
Routing (2)
• ENUM
– Converting E.164 phone number into a Uniform
Resource Identifier (URI)
– Using Domain Name System to store mapping
– A phone number is converted into a special domain
name: e164.arpa
• E.g. 1-800-555-1234  4.3.2.1.5.5.5.0.0.8.e164.arpa
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Telephone Number Mapping &
Routing (3)
• TRIP
– Finding a user in an integrated network
– Used by location server or other NEs to advertise
routes
– Independent of signaling protocols
– Dividing the world into a set of IP Telephone
Administrative Domains (ITADs)
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IP Telephones and Electrical
Power
• Analog telephone system continues to work when
electrical power are unavailable
– The wires that connect a telephone to the central office supply
the power
• Currently, IP telephones have to depend on an external
source of power
– IP phones must have both network connection and power
connection.
– Several mechanism proposed to integrate power with network
connections.
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Summary (1)
• IP telephony or VoIP refers to the transmission of voice telephone
calls over IP networks.
• Hot area both in research and market because of low cost
• Challenge in backward compatibility with PSTN
• The complexity of IP telephony is on signaling. Both ITU and IETF
propose signaling standards.
– H.323, by IUT
– SIP, by IETF, offering similar functions to H.323, but simpler than H.323.
– Both are competing to be recognized as #1 signaling protocol
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Summary (2)
• H.323 uses a set of protocols for call setup and
management
• SIP uses a set of servers to handle various aspects of
signaling
• ENUM maps an E.164 telephone number into a URI
(usually SIP URI)
• TRIP provides routing among IP telephone
administrative domains
• IP telephones depends on external power, while analog
phones don’t.
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