Transcript Chapter 1 - Introduction

Computer Networks and Internets, 5e
By Douglas E. Comer
Lecture PowerPoints
By Lami Kaya, [email protected]
© 2009 Pearson Education Inc., Upper Saddle River, NJ. All rights reserved.
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Chapter 29
Multimedia
and
IP Telephony (VoIP)
© 2009 Pearson Education Inc., Upper Saddle River, NJ. All rights reserved.
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Topics Covered
29.1 Introduction
29.2 Real-Time Data Transmission and Best-Effort Delivery
29.3 Delayed Playback and Jitter Buffers
29.4 Real-Time Transport Protocol (RTP)
29.5 RTP Encapsulation
29.6 IP Telephony
29.7 Signaling and VoIP Signaling Standards
29.8 Components of an IP Telephone System
29.9 Summary of Protocols and Layering
29.10 H.323 Characteristics
29.11 H.323 Layering
29.12 SIP Characteristics and Methods
29.13 An Example SIP Session
29.14 Telephone Number Mapping and Routing
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29.1 Introduction
• This chapter
– continues the discussion by examining the transfer of multimedia
over the Internet
– examines how multimedia can be sent over a best-effort
communication mechanism
– describes a general-purpose protocol for real-time traffic
– considers the transmission of voice telephone calls in detail
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29.2 Real-Time Data Transmission and
Best-Effort Delivery
•
•
Multimedia to refer to data that contains audio or video
Real-time multimedia refers to multimedia data that must be
reproduced at exactly the same rate that it was captured
• How can Internet be used for transfer of real-time multimedia?
– recall that the Internet offers best-effort delivery service
• packets can be lost, delayed, or delivered out of order
• If multimedia data is sent across the Internet without special
treatment, the resulting output may be unacceptable
– Early systems solved the problem by creating communication
networks specifically designed to handle audio or video
• The analog telephone network uses an isochronous network to provide highquality reproduction of audio
• Analog cable television systems are designed to deliver multiple channels of
broadcast video with no interruptions or loss
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29.2 Real-Time Data Transmission and
Best Effort Delivery
• The Internet uses additional protocol support
– Instead of requiring the underlying networks to handle real-time
transmission
• The most significant problem to be handled is jitter
• For example, consider a live webcast
• If a protocol uses timeout-and-retransmission to resend the
packet, the retransmitted packet will arrive too late to be useful
– the receiver will have played the video and audio from successive
packets
– it makes no sense to insert a snippet of the webcast that was missed
earlier
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29.3 Delayed Playback and Jitter Buffers
• To overcome jitter and achieve smooth playback of real-time
data, some techniques are employed, such as
• Timestamps
– A sender provides a timestamp for each piece of data
– A receiver uses the timestamps
• to handle out-of-order packets and
• to display the data in the correct time sequence
• Jitter Buffer
– To accommodate jitter (i.e., small variances in delay), a receiver
buffers data and delays playback
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29.3 Delayed Playback and Jitter Buffers
• In jitter buffer mechanism
– a receiver maintains a list of data items
– and uses timestamps to order the list
• Before playback
– a receiver delays for d time units
• means the data being played is d time units behind the data that is arriving
• If a given packet is delayed less than d,
– the contents of the packet will be placed in the buffer
• before it is needed for playback
– items are inserted into a jitter buffer with some variation in rate
– the playback process extracts data from a jitter buffer at a fixed rate
• Figure 29.1 illustrates the organization of a real-time
playback system
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29.3 Delayed Playback and Jitter Buffers
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29.4 Real-Time Transport Protocol (RTP)
• IP suite contains the RTP
– used to transmit real-time data across the Internet
• Term Transport is used because RTP sits above the
transport layer
• RTP does not ensure timely delivery of data
• Instead, it provides three items in each packet that permit a
receiver to implement a jitter buffer:
– A sequence number that allows a receiver to place incoming packets
in the correct order and to detect missing packets
– A timestamp that allows a receiver to play the data in the packet at
the correct time in the multimedia stream
– A series of source identifiers that allow a receiver to know the
source(s) of the data
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29.4 Real-Time Transport Protocol (RTP)
• Figure 29.2 (below) illustrates an RTP packet header:
– See how the sequence number, timestamp, and source identifier
fields appear in an RTP packet header
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29.5 RTP Encapsulation
• RTP uses UDP for message transport
– Thus, each RTP message is encapsulated in a UDP datagram for
transmission over the Internet
– The resulting messages can be sent via broadcast or multicast
• Figure 29.3 (below) illustrates the three levels of encapsulation that
are used when an RTP message is transferred over a single network
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29.6 IP Telephony
• Companies around the world are replacing traditional
telephone switches with IP routers
• The motivation is economic:
– routers cost much less than traditional telephone switches
• Sending both data and voice in IP datagrams lowers cost
– because the underlying network infrastructure is shared
• The basic idea behind IP telephony is straightforward:
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continuously sample audio
convert each sample to digital form
send the resulting digitized stream across an IP network in packets
and convert the stream back to analog for playback
• However, many details complicate the task
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29.6 IP Telephony (VoIP)
• However, many details complicate the task
– A sender cannot wait to fill a large packet
• because doing so delays transmission by several seconds
– The system must handle call setup
• when a caller dials, the system must translate the phone number to an IP
address, and locate the specified party
– When a call begins
• the called party must accept and answer the call
– When a call ends
• the two parties must agree on how to terminate communication
• The most significant complications:
– IP telephony strives to be backward compatible
• with Public Switched Telephone Network (PSTN) or some call Plain Old
Telephone System (POTS)
– Also, integration with Private Branch Exchange (PBX)
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29.7 Signaling and VoIP Signaling
Standards
• Two groups have created standards for VoIP:
– International Telecommunications Union (ITU)
– Internet Engineering Task Force (IETF)
• Both groups agree on the basics for encoding and transfer:
– Audio is encoded using Pulse Code Modulation (PCM)
– RTP is used to transfer the digitized audio
• The main complexity of VoIP
– lies in call setup and call management
• The process of establishing and terminating a call is known
as signaling and includes
– mapping a phone number to a location
– finding a route to the called party
– handling other details such as call forwarding
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29.7 Signaling and VoIP Signaling
Standards
• Mechanism used in the traditional telephone system to handle
call management is known as Signaling System 7 (SS7)
• To be compatible with existing telephones
– new protocols must be able to interact with SS7
– this is to place outgoing calls and to accept incoming calls
• Set of signaling protocols were proposed for use with VoIP
– IETF proposed
• Session Initiation Protocol (SIP)
• Media Gateway Control Protocol (MGCP)
– ITU proposed
• a large, comprehensive set of protocols under the umbrella of H.323
– The two groups jointly proposed
• Megaco (H.248)
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29.8 Components of an IP Telephone
System
• Figure 29.4 (below) lists main components of an IP
telephone system
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29.8 Components of an IP Telephone
System
• Figure 29.5 (below) illustrates how they are used to
interconnect networks
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29.8 Components of an IP Telephone
System
• An IP telephone
– connects to a network, uses IP for all communication
– offers a traditional telephone interface
• that allows a user to place or receive telephone calls
• A Media Gateway Controller (Gatekeeper or Soft Switch)
– provides overall control and coordination between a pair of IP
telephones
• allowing a caller to locate a callee or access services such as call forwarding
•
A Media Gateway
– provides translation of audio as a call passes across the boundary
between an IP network and the PSTN
• A Signaling Gateway
– also spans the boundary between a pair of disparate networks
– translation of signaling operations (either side to initiate a call)
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29.8 Components of an IP Telephone
System
• The concepts and terminology defined above present a
straightforward and somewhat simplified view of VoIP
– that was derives from work in the IETF and ITU on the Megaco and
Media Gateway Control Protocol (MGCP)
• Practical implementations of VoIP service are more complex
• The next sections give examples:
– 29.8.1 SIP Terminology and Concepts
– 29.8.2 H.323 Terminology and Concepts
– 29.8.3 ISC Terminology and Concepts
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29.8 Components of an IP Telephone System
29.8.1 SIP Terminology and Concepts
• The Session Initiation Protocol (SIP) defines set of
elements for the signaling system
• User Agent (device that makes or terminates phone calls)
• Location Server
– manages a database of information about each user (such as a set
of IP addresses, subscribed services, and the user's preferences)
• Support Servers (proxy, redirect, registrar)
– Proxy Server
• can forward requests from user agents to another location
– Redirect Server
• handle tasks such as call forwarding and 800-number connections
– Registrar Server
• to receive registration requests and update the database that location
servers consult
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29.8 Components of an IP Telephone System
29.8.2 H.323 Terminology and Concepts
• The H.323 defines alternative terminology and additional
concepts, focuses on PSTN interaction
• It is extremely broad and covers many details
• H.323 can be summarized as follows:
– Terminal
• provides the IP telephone function, which may also include facilities for video
and data transmission
– Gatekeeper
• H.323 gatekeeper provides location and signaling functions
• coordinates the operation of the gateway to provide connection to the PSTN
– Gateway
• H.323 uses a single gateway to interconnect the IP telephone system with
the PSTN
• the gateway handles both signaling and media translation
– Multipoint Control Unit (MCU)
• An MCU provides services such as multipoint conferencing
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29.8 Components of an IP Telephone System
29.8.3 ISC Terminology and Concepts
• Vendors formed International SoftSwitch Consortium (ISC)
– to create a uniform, comprehensive functional model
• that incorporates all models of IP telephony into a single framework
• ISC defined a list of functions that suffices for all situations:
– Media Gateway Controller Function (MGC-F)
• maintains state information in endpoints; it provides call logic and call control
– Call Agent Function (CA-F)
• The CA-F is a subset of the MGC-F that maintains call state
– InterWorking Function (IW-F)
• is a subset of the MGC-F that handles signaling between heterogeneous
networks such as SS7 and SIP
– Routing Function and Accounting Function (R-F/A-F)
• R-F handles routing of calls for the MGC-F
• A-F collects information used for accounting and billing
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29.8 Components of an IP Telephone System
29.8.3 ISC Terminology and Concepts
– Signaling Gateway Function (SG-F)
• handles signaling between an IP network and the PSTN
– Access Gateway Signaling Function (AGS-F)
• handles signaling between an IP network and a circuit-switched access
network such as the PSTN
– Application Server Function (AS-F)
• The AS-F handles a set of application services such as voicemail
– Service Control Function (SC-F)
• It is called when an AS-F must control (i.e., change) the logic of a service
– Media Gateway Function (MG-F)
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•
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handles translation of digitized audio between two forms
may also include detection of events such as whether a phone is off-hook
may include recognition of Dual Tone Multi-Frequency (DTMF) signals
the audio signaling standard that is known as Touch Tone encoding
– Media Server Function (MS-F)
• manipulates a media packet stream on behalf of an AS-F application
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29.9 Summary of Protocols and Layering
• Multiple groups have proposed protocols for VoIP
– competing protocols exist at most layers of the protocol stack
• Figure 29.6 (below) lists some of the proposed protocols
– along with their position in the Internet 5-layer reference model
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29.10 H.323 Characteristics
• H.323 standard consists of a set of protocols
– that work together to handle all aspects of telephone communication
• The highlights of H.323 are:
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Handles all aspects of a digital telephone call
Includes signaling to set up and manage the call
Allows the transmission of video and data while a call is in progress
Sends binary messages that are defined by ASN.1
Messages are encoded using Basic Encoding Rules (BER)
Incorporates protocols for security
Uses a special hardware unit known as a Multipoint Control Unit to
support conference calls
– Defines servers to handle tasks such as address resolution,
authentication, authorization, accounting, call forwarding, etc.
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29.11 H.323 Layering
• H.323 protocols use both TCP and UDP for transport
– audio can travel over UDP
– while a data transfer proceeds over TCP
• Figure 29.7 (below) illustrates the basic layering in the
H.323 standard
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29.12 SIP Characteristics and Methods
Highlights of IETF's Session Initiation Protocol (SIP) are:
• Operates at the application layer
• Encompasses all aspects of signaling
– including location of a called party, notification and setup,
determination of availability and termination
• Provides services such as call forwarding
• Relies on multicast for conference calls
• Allows the two sides to negotiate capabilities
– and choose the media and parameters to be used
• A SIP-URI contains a user's name and a domain name
– at which the user can be found
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29.12 SIP Characteristics and Methods
• SIP defines some message types and extensions
– Message types are known as SIP method
• Figure 29.8 (below) lists the basic SIP methods
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29.13 An Example SIP Session
• An example of the messages sent during a SIP session will
clarify some of the details
– and help explain the general idea behind most IP telephony
• Figure 29.9 lists a sequence of messages sent
– A user agent, A, contacts a DNS server
• then communicates with a proxy server, which invokes a location server
– Once the call is established, the two VoIP communicate directly
– Finally, SIP is used to terminate the call
• Typically, a user agent is configured with the IP address of
one or more DNS and one or more proxy servers
• Each proxy server is configured with the address of one or
more location servers
– If a given server is unavailable
• SIP can find an alternate quickly
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Figure 29.9
An example of
the messages
exchanged by
SIP to manage
a telephone
call
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29.14 Telephone Number Mapping and
Routing
• How should IP telephone users be named and located?
– The PSTN follows ITU standard E.164 for telephone numbers
– The SIP uses IP addresses
• The problem of locating users is complicated
– because multiple types of networks may be involved
• Designers define two sub-problems:
– locate a user in the integrated network
– and find an efficient route to the user
• The IETF has proposed protocols that correspond to the
mappings needed for the two sub-problems:
– ENUM converts a telephone number to a URI
– TRIP finds a user in an integrated network
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29.14 Telephone Number Mapping and
Routing
• ENUM (short for E.164 NUMbers )
– solves the problem of converting an E.164 tel number into a URI
– ENUM uses the DNS to store the mapping
• An ENUM mapping can be 1-to-1 or 1-to-many
• Telephone Routing over IP (TRIP)
– Solves the problem of finding a user in an integrated network
– A location server or other network element can use TRIP to advertise
routes
• Thus, two location servers use TRIP to inform each other about external
routes that they each know
– TRIP divides the world into a set of IP Telephone Administrative
Domains (ITADs)
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