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Wide Area Networking Technologies

Unit 45

Lecture 2

Objectives
Quality of service management eg DSCP
(Differentiated Service Code Point), IP precedence, queues,
base rules, congestion management; quality of service need
eg Voice over IP, video streaming, audio streaming;

Using Network Simulator, Cisco Packet Tracer


Interconnecting LANs with the Router

Monday Class
1. Three bank holydays
2. We have three sessions we need to relocate them to different day of the
week?
3. Three session means three weeks

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1
Quality of Service (QoS)

Quality of Service prioritise traffic throughout a network, QoS enables the end-to-
end traffic priority policy to improve control and throughput of important data.
Managing the available bandwidth so that the most important traffic goes first
Upgrade or downgrade traffic from various servers.
Control the priority of traffic from dedicated VLANs or applications.
Change the priorities of traffic from various segments of your network as your business
needs change.
Set priority policies in edge switches in your network to enable traffic handling rules
across the network.

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QoS on the switches covered by this guide supports these types of traffic marking:
802.1p prioritisation: Controls the outbound port queue priority for traffic leaving the
switch, and (if traffic exits through a VLAN-tagged port) sends the priority setting with
the individual packets to the downstream devices.
IP Type-of-Service (ToS): Enables the switch to set, change, and honour prioritisation
policies by using the Differentiated Services (diffserv) bits in the ToS byte of IPv4 packet
headers.
Steps for Configuring QoS on the Switch
a. UDP/TCP applications
b. Device Prioritydestination or source IP address (Note that destination has
precedence over source
c. IP Type-of-Service Precedence Bits (Leftmost three bits in the ToS field of IP
packets)
d. IP Type-of-Service Differentiated Service bits (Leftmost six bits in the ToS field
of IP packets)
e. VLAN Priority (requires at least one tagged VLAN on the network)
f. Source-Port
g. Incoming 802.1p Priority (requires at least one tagged VLAN on the network)

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Differentiated Services Code Point (DSCP) overview

Differentiated Services Code Point (DSCP) is a field in an IP packet that enables


different levels of service to be assigned to network traffic. This is achieved by
marking each packet on the network with a DSCP code and appropriating to it the
corresponding level of service.

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Differentiated Services Code Point (DSCP) overview

DSCP is the combination of IP Precedence and Type of Service fields. In


order to work with legacy routers that only support IP Precedence, DSCP
values are used because they are compatible with IP Precedence fields.

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3
How DSCP works

Quality of Service (QoS)-enabled programs request a specific service type for a


traffic flow through the generic QoS (GQoS) application programming interface
(API). The available service types are:
Guaranteed service
Guaranteed service provides high quality, quantifiable guarantees with bounded
(guaranteed minimum) latency.
Controlled load service
Controlled load service provides high quality, quantifiable guarantees without
bounded latency.
DSCP is a six-bit field carrying the default values that are displayed in the following
table. The equivalent IP Precedence and IEEE 802.1p values are shown in separate
columns.

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What is the difference between IP precedence and DSCP

ip precedence is the old way for type of service(ToS)


DSCP is the new way for type of service(ToS)
DSCP is backward compatible with ip precedence

Explicit Congestion Notification (ECN)

8 bit TOS
IP precedence use the first 3 bits
DSP use the first 6 bits
(in fact the 6th bit is never used)

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4
IP Precedence (Major class)

It use a scale of 0 to 7 and will be typically used as below


0 : Best Effort (000) | 1 : Medium priority(001)
2 : Hight priority(010) | 3 : Call signals(011)
4 : Video conference(100) | 5 : Voice(101)
6 : Reserved (110) | 7 : Reserved(111)
again this is the old way to be used with ToS
Configuration Example: <0-7> Precedence value
critical Set packets with critical precedence (5)
flash Set packets with flash precedence (3)
R1(config)#class-map match-all IP-PRECEDENCE flash-override Set packets with flash override precedence (4)
R1(config-cmap)#match protocol rtp immediate Set packets with immediate precedence (2)
R1(config-cmap)#exist internet Set packets with internetwork control precedence (6)
R1(config)#policy-map IP-PRECEDENCE-POLICY network Set packets with network control precedence (7)
R1(config-pmap)#class IP-PRECEDENCE priority Set packets with priority precedence (1)
R1(config-pmap-c)#set ip precedence ? routine Set packets with routine precedence (0)
R1(config-pmap-c)#set ip precedence critical
R1(config-pmap-c)#exit
R1(config)#interface gigabitEthernet 0/1
R1(config-if)#service-policy input IP-PRECEDENCE-POLICY

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Difference between IP precedence and DSCP

The advantages of DSCP is that you may have more levels to define than IP precedence. For each level
in IP precedence , there are 3 sub levels.at the same time it is backward compatible
default Match packets with default dscp (000000) (no sub level for level 0 or best effort)
af11 Match packets with AF11 dscp (001010)
in this level af11 = 001|010 , as the 6th bit is not used this will become in reality 001|01 which is equal
to 1|1 in decimal and the final result is af11
af12 Match packets with AF12 dscp (001100)
af13 Match packets with AF13 dscp (001110)
The second part or the second 3 bits is called Drop Preference ,the higher it is , the more chance it has to
be dropped . To resume af11 is better than af13 because af13 has more chance to be dropped than af11
af21 Match packets with AF21 dscp (010010)
af22 Match packets with AF22 dscp (010100)
af23 Match packets with AF23 dscp (010110)
af31 Match packets with AF31 dscp (011010)
af32 Match packets with AF32 dscp (011100)
af33 Match packets with AF33 dscp (011110)
af41 Match packets with AF41 dscp (100010)
af42 Match packets with AF42 dscp (100100)
af43 Match packets with AF43 dscp (100110)
ef Match packets with EF dscp (101110) (no sub levels for level 5 or critical level)
Configuring DSCP on a cisco router is the same as IP precedence the only difference will be in the
following line
Router(config-pmap-c)#set ip dscp ef

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QoS for VoIP Overview

VoIP is a replacement for standard public switched telephone network (PSTN) telephony services.
customers need to receive the same quality of voice transmission they receive with basic telephone services
meaning consistently high-quality voice transmissions. Like other real-time applications,
VoIP is extremely bandwidth- and delay-sensitive
voice packets should not be dropped, excessively delayed, or suffer varying delay (otherwise known as jitter).
The default G.729 codec requires packet loss far less than 1 percent to avoid audible errors
The ITU G.114 specification recommends less than 150 millisecond (ms) one-way end-to-end delay for
high-quality real-time traffic such as voice.
For international calls, one-way delay up to 300 ms is acceptable, especially for satellite transmission. This one-way
delay takes propagation delay into considerationthe time required for the signal to travel the distance
Jitter buffers (used to compensate for varying delay) further add to the end-to-end delay, and are usually
only effective on delay variations less than 100 ms
VoIP can guarantee high-quality voice transmission only if the voice packets, for both the signaling and
audio channel, are given priority over other kinds of network traffic
VoIP traffic must be guaranteed certain compensating bandwidth, latency, and jitter requirements. QoS
ensures that VoIP voice packets receive the preferential treatment they require. In general, QoS provides
better (and more predictable) network service by providing the following features:
Supporting dedicated bandwidth
Improving loss characteristics
Avoiding and managing network congestion
Shaping network traffic
Setting traffic priorities across the network

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QoS Methods

There are a number of different methods that can be used to control the QoS of a voice connection; these include:
Classification and Marking
Link Efficiency
Congestion Management
Congestion Avoidance
Classification and Marking
QoS classification and marking is Differentiated Services (DiffServ).
all traffic classified into a specific traffic classification (for example, Voice Traffic or Data Traffic).
Differentiated Service Codepoint (DSCP). prioritising which traffic to process first
Link Efficiency
The most commonly known mechanisms include IP header and payload compression.
Congestion Management
The more congested a link, the less likely a packet will be able to get through in a timely manner required by VoIP
Congestion management mechanisms attempt to control the amount of congestion faced by traffic by processing the traffic in a variety of
different ways, The most common methods include:
FIFO
Priority Queuing (PQ)
Custom Queuing (CQ)
Weighted Fair Queuing (WFQ)
Class Based Weighted Fair Queuing (CBWFQ)
Low Latency Queuing (LLQ)
Congestion Avoidance
Congestion avoidance is another method of QoS; the most common of the techniques used is called Weighted Random Early Detection
(WRED). Basically, WRED attempts to predict that congestion will be forthcoming, and when this happens packets are selectively dropped to
avoid congestion.

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6
Wide Area Network 13

The Router

The router is a layer 3 device in the OSI model, which means the router uses the network
address (layer 3 addressing) to make routing decisions regarding forwarding the data
packets.

Remember, the OSI model separates the network responsibilities into different layers. In
the OSI model, the layer 3 or Network layer responsibilities include handling of the network
address.

The network address is also called a logical address rather than a physical address such as
the MAC address. The logical address describes the IP address location of the network
and the address location of the host in the network.

Basically the router is configured to know how to route data packets entering or exiting the
LAN. This differs from the bridge and the layer 2 switch which use the Ethernet address for
making decisions regarding forwarding data packets and only know how to forward data to
hosts physically connected to their ports.

Routers are used to interconnect LANs in a campus network.

The networks can use the same protocol (e.g. Ethernet) or they can be used to interconnect
LANs that are using different layer 2 technologies such as a Ethernet and Token Ring.

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The Router

Routers also make it possible to interconnect to LANs around the country and the
world and interconnect to many different networking protocols.

Routers have multiple port connections for connecting to the LANs and by definition, a
router must have a minimum of three ports.

The common symbol used to represent a router in a networking drawing is provided.


The arrows pointing in and out indicate that data enters and exits the routers through
multiple ports. The router ports are bi-directional, meaning that data can enter and
exit the same router port.

Often the router ports are called the router interface, the physical connection where
the router connects to the network.

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Router Interface

Cisco 2800 Series Router - Interface

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Router Interface

Console inputThis input provides an RS-232 serial


communications link into the router for initial router
configuration. This router is also showing a USB input.

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Router Interface

Auxiliary inputThis input is used to connect a dial-in modem


into the router. The auxiliary port provides an alternate way to
remotely log into the router if the network is down. This port
also uses an RJ-45 connection.
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Router Interface

Serial Input - This is a serial connection and it has a built-in CSU/DSU.


This interface is used to provide a T1 connection to the communications
carrier. (Note: The CSU/DSU function is presented in Chapter 10.) This
type of connection (RJ-45) replaces the older cabling using V.35 cable
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Router Interface

FastEthernet Input These are 100 Mbps interfaces for the


router.

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Router Interface

Voice Interface Card (VIC2-4FXO) This interface shows 4 phone


line connections. This router can be programmed as a small PBX (Private
Branch Exchange) for use in a small office.

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Router Interface

Modem - This interface has two RJ-11 jacks and has two V.90 analog
internal modems. These modems can be used to handle both incoming
and outgoing modem calls.

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Router Interface

FXS/DID - This interface is a four-port FXS and DID voice/fax interface


card. FXS is a Foreign Exchange Interface that connects directly to a
standard telephone. DID is Direct Inward Dialing is a feature that enables
callers to directly call an extension on a PBX.
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12
Router show Command

Example : Show IP Interface


access-lists List access lists
Arp ARP table
Flash System Flash information
History Display the session history
interfaces Interface status and configuration
IP IP information
Line TTY line information
Protocols Active network routing protocols
privilege Show current privilege level
startup-config Contents of startup configuration
running-config Current operating configuration

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Cisco Router Configuration Commands

Requirement Cisco Command


Set a console password to cisco Router(config)#line con 0
Router(config-line)#login
Router(config-line)#password cisco
Set a telnet password Router(config)#line vty 0 4
Router(config-line)#login
Router(config-line)#password cisco
Stop console timing out Router(config)#line con 0
Router(config-line)#exec-timeout 0 0
Set the enable password to cisco Router(config)#enable password cisco
Set the enable secret password to cisco. Router(config)#enable secret cisco
This password overrides the enable password
and is encypted within the config file
Enable an interface Router(config-if)#no shutdown
To disable an interface Router(config-if)#shutdown
Set the clock rate for a router with a DCE cable to 64K Router(config-if)clock rate 64000
Set a logical bandwidth assignment of 64K to the serial interface Router(config-if)bandwidth 64
Note that the zeroes are not missing
To add an IP address to a interface Router(config-if)#ip addr 10.1.1.1 255.255.255.0
To enable RIP on all 172.16.x.y interfaces Router(config)#router rip
Router(config-router)#network 172.16.0.0
Disable RIP Router(config)#no router rip
To enable IRGP with a AS of 200, to all interfaces Router(config)#router igrp 200
Router(config-router)#network 172.16.0.0
Disable IGRP Router(config)#no router igrp 200
Static route the remote network is 172.16.1.0, with a mask of 255.255.255.0, Router(config)#ip route 172.16.1.0 255.255.255.0 172.16.2.1 5
the next hop is 172.16.2.1, at a cost of 5 hops

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Routers - Review

A router routes data based on the destination network address or logical address
rather than the physical address used by the layer 2 devices such as the switch
and the bridge.

Information exchanged with bridges and layer 2 switches requires that the MAC
address for the hosts be known.

Routed networks such as most enterprise and campus networks use IP


addressing for managing the data movement. Enterprise network is a term
used to describe the network used by a large company.

The use of the network or logical address on computers allows the information to
be sent from a LAN to a destination without requiring that the computer know the
MAC address of the destination computer. Remember, final delivery of data
packets is based on knowing the MAC address of the destination.

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Router A connects directly to the LAN A
switch via FastEthernet port FA0/0.
Router A also
connects directly to Router B via the
FastEthernet port FA0/1 and connects to
Router C via FastEthernet port FA0/2.

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Router B connects directly to the LAN B


switch via FastEthernet port FA0/0. Router
B also connects to the LAN C switch via
FastEthernet port FA0/1. Router B
connects directly to Router A via
FastEthernet port FA0/2 and connects to
Router C via FastEthernet port FA0/3.

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15
Serial Ports
The serial ports (s0, s1, s2 ) are not being used to interconnect the routers
in this example campus network. The serial interfaces are typically used to
interconnect LANs that connect through a data communications carrier such
as a telephone company.

Routing Tables
Delivery of the information over the network is made possible by the use of an
IP address and routing tables. Routing tables keep track of the routes to
used for forwarding data to its destination.

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The Console Port

The routers console port is used as the initial interface for configuring the router.
It is a slow speed serial communications link and it is the only way to
communicate with the router until the router interfaces have been configured.

Specifically, the console connections is an RS-232 serial communications port that


uses an RJ-45 jack to connect to its interface.

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DB-9 DB-25

The RS-232 protocol running on the console port is the same


communications protocol format used on a computers (COM1, COM2)
port, however, the connector for the serial communications port on
the computer is either a DB-9 or DB-25 type connector. The DB-25
serial connection is seldom used.

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Hyperterminal

A serial communications software package such as Microsofts


Hyperterminal can be used for establishing the communications
link to the routers console input. The settings for the serial
interface on Ciscos console port are provided.

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Enter a name for your
connection such as
CiscoRouter and select
an icon to be associated
with the connection.
Click OK when done.

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The Connect To menu lets you


specify how you are making the
serial connection to the router.
This example is showing the
connection is configured to use the
computers COM2 serial port.
Change the Connect using
parameter to match the connection
(COM1, COM2, ) you have
made on your computer.

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The next menu is the properties
menu for your serial connection.
This menu is labeled COM2
Properties since the Connect using
COM2 parameter was specified in
the previous menu.
The COM2 properties will have to
be set to match these settings. The
COM2 Properties menu with the
settings entered is shown

Wide Area Network 37

You should see the image shown when a connection has been established. If
the text does not display Press RETURN to get started then press enter to see
if the router resets itself. Another possible screen you might see may only have
the Router> prompt. Press enter and if the Router> remains then you are
connected. If this doesnt correct the displayed text the router may need to be
restarted.
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The Routers Privileged EXEC Mode

Configuring a router interface requires that the privileged mode be entered on the router.

The privileged mode allows full access for configuring the router interfaces and configuring
routing protocol.

This chapter focuses on general configuration steps for the router and configuring the
routers interfaces, both Ethernet and serial.

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Privileged Mode (Router #)

The privileged mode is entered using the command


enable at the Router> prompt as shown. The #
sign after the router name indicates you are in the
privileged mode (e.g. Router#).

Use caution once you have entered the privileged


mode in a router. It is easy to make mistakes and
incorrectly entered router configurations will affect
your network.

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Configuration Mode

The next commands examined require that the routers terminal configuration
mode be entered. To do this, enter the command configure terminal
(abbreviated conf t) at the RouterA# prompt to enter the routers configuration
mode.

RouterA#conf t
Enter configuration commands, one per line. End with Ctrl/Z.
RouterA(config)#

Or

RouterA#configure terminal
Enter configuration commands, one per line. End with Ctrl/Z.
RouterA(config)#

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Password Protection

Password protection for the privileged (enable) mode is configured by


setting the enable secret. The steps are as follows. Enter the routers
configure terminal mode by entering configure terminal or conf t at the
Router# prompt. Enter the command enable secret [your-password]
<enter>. An example is shown.

Router# conf t
Router(config)#
Router(config)# enable secret my-secret

This sets the password for entering the routers privileged mode to my-
secret. The password for entering the routers privileged mode must now
be entered to gain access to the routers privileged mode.

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Setting the Line Console Password

The router has three line connections that can be used by a user to gain access
to the router. The line connections available on a router can be displayed using
the line ? command at the Router(config)# prompt. The available line
connections typically are:

aux Auxiliary line (modem back-up)


console Primary terminal line (console port)
vty Virtual terminal
(for a telnet connection)

The console (primary terminal line) is the console port and vty is the virtual
terminal used for telnet connections. The following steps demonstrate how to
configure password protection for the console port and the virtual terminal.

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Passwords

Router(config)# line console 0


Router(config-line)# login
Router(config-line)# password [my-secret2]

Router(config)# line vty 0 4


Router(config-line)#login
Router(config-line)# password [my-secret3]

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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
thus, 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 high-quality reproduction of audio
Analog cable television systems are designed to deliver multiple
channels of broadcast video with no interruptions or loss

Wide Area Network 45

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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23
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

Wide Area Network 47

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 below illustrates the organization of a real-time playback system
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24
Delayed Playback and Jitter Buffers

Wide Area Network 49


49

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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25
Real-Time Transport Protocol (RTP)

Figure (below) illustrates an RTP packet header:


See how the sequence number, timestamp, and source identifier fields
appear in an RTP packet header

Wide Area Network 51

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 (below) illustrates the three levels of encapsulation that are used when an RTP
message is transferred over a single network

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26
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:
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
Wide Area Network 53

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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27
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

Wide Area Network 55

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, both 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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28
Components of an IP Telephone System

Table (below) lists main components of an IP telephone system

Wide Area Network 57

Components of an IP Telephone System

Figure (below) illustrates how they are used to interconnect networks

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29
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)

Wide Area Network 59

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:
SIP Terminology and Concepts
H.323 Terminology and Concepts
ISC Terminology and Concepts

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30
Components of an IP Telephone System
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

Wide Area Network 61

Components of an IP Telephone System


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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Components of an IP Telephone System
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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Components of an IP Telephone System


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)
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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Summary of Protocols and Layering

Because multiple groups have proposed protocols for VoIP, competing protocols exist
at most layers of the protocol stack
Figure (below) lists some of the proposed protocols along with their position in the
Internet 5-layer reference model

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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:
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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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
Table (below) illustrates the basic layering in the H.323 standard

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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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SIP Characteristics and Methods

SIP defines six basic message types and seven extensions


Message types are known as SIP method
Table (below) lists the basic SIP methods

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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 below lists a sequence of messages sent SIP Session
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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SIP Session 1

An example of the
messages
exchanged by SIP
to manage a
telephone call

An Example SIP
Session

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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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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 Domain Name System 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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Key Terms

RS-232 Privileged Mode

DB-9 enable

DB-25 Router#

Console Cable configure terminal (conf t)

COM1, COM2, Router(config)#

no shutdown (no shut) Router(config-line)#

show ip interface brief (sh Router(config-if)#


ip int brief)
DCE
DTE

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Braden, R., Clark, D., Shenker, S. Integrated Services in the Internet
Architecture: an Overview. June 1994. URL:ftp://ftp.rfc-editor.org/in-
notes/rfc1633.txt (14 March 2003).
Elachi, Joanna. Standards Snapshot: The State Of The Big 3 in VoIP
Signaling, Protocols 27 November 2002.
URL:http://www.commweb.com/article/COM20001127S0008 (13 Feb
2003).

http://www.networktutorials.info/

http://docwiki.cisco.com/wiki/Introduction_to_WAN_Technologies
http://technet.microsoft.com/en-us/library/bb742465.aspx
www.networktutorials.info/

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