Voice over Internet Protocol

(VoIP)

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 Main Contents
• Overview of TCP/IP
• Internet Protocol.
• VoIP Network
• VoIP Media Protocols (RTP and RTCP)
• VoIP Signaling Protocols
– H.323 Protocol Stack
– SIP and SDP
– Stream Control Transmission Protocol (SCTP)
– Media Gateway Control Protocol (MGCP)
• Quality of Service
• IP Telephony Market
• Lab Practice

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 Main Contents
 Overview of TCP/IP
• Internet Protocol.
• VoIP Network
• VoIP Media Protocols (RTP and RTCP)
• VoIP Signaling Protocols
– H.323 Protocol Stack
– SIP and SDP
– Stream Control Transmission Protocol (SCTP)
– Media Gateway Control Protocol (MGCP)
• Quality of Service
• IP Telephony Market
• Lab Practice

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 Internetworking Technology

The main goal of TCP/IP(internetworking technology) is to

connect multiple physical networks into a large ,uniform
communication system.

Different physical networks may have different frame formats.

Token-
ring

Ethernet
Router
LAN 1 LAN 2
LAN = Local Area Network
TCP = Transmission Control Protocol
IP = Internet Protocol 4
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 • Different addressing schemes

• Different Maximum packet sizes

• Different Network access mechanisms
• Different time outs
• Error Recovery
• Status Reporting
• connection, connectionless.
and so on…
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 OSI Vs TCP/IP protocol stacks
OSI = Open System Interconnection
FTP = File Transfer Protocol
ASCII =American Standard Code for Information Interchange

Application
FTP
Application
Presentation ASCII/Binary

Session
TCP
Transport
Transport

Network IP Network

Link
Ethernet Link
Physical

The 7-layer OSI Model The 4-layer Internet model

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 UDP = User Datagram Protocol
(Using UDP, programs on networked computers can send short
messages sometimes known as datagrams to one another).
SEGMENT =Network designers create segments to physically
separate related computers into groups. This grouping can
improve network performance and security.

Protocol Stack

App

Transport TCP / UDP Data Hdr TCP Segment

Network IP Data Hdr IP Datagram

Link

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 Application Layer
IANA = The Internet Assigned Numbers Authority (IANA is
one of the primary governing bodies for Internet networking )
RFC = Request For Comments

Every application uses a unique port number.

Ports numbers are assigned by IANA.

RFC 1700 which tells the information about port

numbers is now replaced by RFC 3232.
16
Port numbers are in the range of 0 – 65535(2 - 1)

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 Port Assignment (1)

The port numbers are divided into three ranges:

The Well Known Ports:
The Well Known Ports are those from 0 through 1023.
The Registered Ports:
The Registered Ports are those from 1024 through 49151
The Dynamic and Private Ports:
The Dynamic and Private Ports are those from 49152
through 65535
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 Port Assignment (2)

The complete list of port numbers are mentioned at

website of IANA.

http://www.iana.org/assignments/port-numbers

The few well-known port numbers are;

HTTP 80
FTP 21
TELNET 23 HTTP: Hypertext Transfer Protocol
FTP = File Transfer Protocol
TELNET = TELetype NETwork (a network protocol used on
the Internet or local area network LAN connections)

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 Port Assignment (3)

The Trojans also use port numbers may be within the network
(un-used ports) and outside the network (used ports).
The lower ports are often used by Trojans that steals password
and either mail the passwords to attackers or hide them in FTP-
directories.

The higher ports are often used by Remote Access Trojans that
can be reached over the network.

TROJANS = A destructive program that masquerades as a

benign application. Unlike viruses, Trojan horses do not
replicate themselves but they can be just as destructive. One of
the most insidious types of Trojan horse is a program that
claims to rid your computer of viruses but instead introduces
viruses onto your computer.

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 Port Assignment (4)

The list of most popular Trojans are available at the

website of Simovits Consulting.

http://www.simovits.com/nyheter9902.htm
l
At operating system level we use TCP/IP filters to
overcome the attacks of Trojans.

Filters = A program that accepts a certain type of data as

input, transforms it in some manner, and then outputs the
transformed data.

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 Internet Protocol

IP ensures two important mechanisms ;

addressing and fragmentation

IP doesn’t guarantee
• reliability.
• flow control.
• sequencing.
Fragmentation = Refers to the condition of, say, a disk in

which files are divided into pieces scattered around the disk.

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 Fragmentation and Re-assembly

• Different packet sizes

• When to re-assemble
– At destination
• Results in packets getting smaller as data traverses
internet
– Intermediate re-assembly
• Need large buffers at routers
• Buffers may fill with fragments
• All fragments must go through same router
– Inhibits dynamic routing
Buffers = A temporary storage area, usually in RAM. The
purpose of most buffers is to act as a holding area, enabling the
CPU to manipulate data before transferring it to a device.

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 IP Fragmentation (1)

• IP re-assembles at destination only

• Uses fields in header
– Data Unit Identifier (ID)
• Identifies end system originated datagram
– Source and destination address
– Protocol layer generating data (e.g. TCP)
– Identification supplied by that layer
– Data length
• Length of user data in octets

Octets = A group of eight

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 IP Fragmentation (2)

– Offset
• Position of fragment of user data in original
datagram
• In multiples of 64 bits (8 octets)
– More flag
• Indicates that this is not the last fragment

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 Example of IP Fragmentation

208/8 = 26

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 IP Header
IHL = Internal Header Length
Time To Live = Time-to-live (TTL) is a value in an Internet Protocol (IP) packet that tells a network
router whether or not the packet has been in the network too long and should be discarded. For a
number of reasons, packets may not get delivered to their destination in a reasonable length of time.
Header Checksum =16 bits. A checksum on the header only. Since some header fields change (e.g., time
to live), this is recomputed and verified at each point that the internet header is processed.
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 IP Header Fields (1)
Versions:
indicates 4 bits version number.
IHL (Internet Header Length) 4 bits:
Defines the length of the header in multiple of 32 bits(4 octets).
The min. value is 5 (x 4 octets)=20 octets.
Type of service (8 bits):
It specifies reliability, precedence, delay and throughput.
Total Length (16 bits):
Total datagram length in octets. Throughput = The amount of data transferred from one place
to another or processed in a specified amount of time. Data
transfer rates for disk drives and networks are measured in
terms of throughput. Typically, throughputs are measured in
kbps, Mbps and Gbps.

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 IP Header Fields (2)
Identification (16 bits):
It identifies the datagram uniquely on the basis of source
address , Destination address and protocol.
Flags (3 bits):
Only two bits are currently defined.
More and Don’t fragment flags.
Fragment offset: Hops = An intermediate connection in a string of connections
linking two network devices. On the Internet, for example,
most data packets need to go through several routers before
Multiple of 64 bits (8 octets). they reach their final destination. Each time the packet is
forwarded to the next router, a hop occurs. The more hops, the
longer it takes for data to go from source to destination.

TTL (Time to Live):

Max no of hops that a datagram can travel.
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 IP Header Fields (3)

Header Checksum(16 bits):

It is used for error detection and it is computed at every node.
Source/Destination Address (32 bits):
Options (variable):
Padding (variable):
Data (variable):
Max size is 65535

Padding = Filling in unused space.

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 ICMP : Internet Control Message Protocol (1)

Internet Protocol has no error control and flow control

mechanism.
In case of network problem , IP use the services of Internet
Control Message Protocol (ICMP).

The few ICMP messages are;

Source Quench - for sender to reduce data rate
Destination unreachable - router has to fragment a don’t
fragment datagram
Time Stamp: to record the inter-arrival packet time.

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 ICMP : Internet Control Message Protocol (2)
ICMP Message
20bytes 4bytes

IP ICMP ICMP
Header Header Data

Type Code Checksum

1byte 1byte 2bytes

Types and Codes

• Echo Request (type=8, code=0)
• Echo Reply(type=0, code=0)
• Destination Unreachable(type=3, code=0)
• Time Exceeded(type=11, code=0) : Time-to-Live =0
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 Type Description 18 Address mask reply.

0 Echo reply. 19 Reserved (for security).

1 Reserved. 20
- Reserved (for robustness experiment).
2 Reserved. 29
3 Destination unreachable.
30 Traceroute
4 Source quench.
31 Conversion error
5 Redirect.
32 Mobile Host Redirect.
6 Alternate Host Address.
33 IPv6 Where-Are-You.
7  
34 IPv6 I-Am-Here.
8 Echo request.
35 Mobile Registration Request.
9 Router advertisement.
36 Mobile Registration Reply.
10 Router solicitation.
37 Domain Name request
11 Time exceeded.
38 Domain Name reply
12 Parameter problem.
39 SKIP Algorithm Discovery Protocol.
13 Timestamp request.
40 Photuris, Security failures
14 Timestamp reply.
41 Experimental mobility protocols.
15 Information request.
42
16 Information reply. - Reserved.
255
17 Address mask request.

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 Transmission Control Protocol (1)

TCP offers connection-oriented service, confirms sequential

delivery of all the segments ,offers error control
(acknowledgements) and flow control(window).
That’s why TCP is slow but more reliable than UDP.
UDP offers connectionless service without
acknowledgements , error control and flow control
Connectionless=In telecommunication, connectionless describes
Connection Oriented =In telecommunications, connection-
communication between two network end points in which a
oriented describes a means of transmitting data in which the
message can be sent from one end point to another without
devices at the end points use a preliminary protocol to establish
prior arrangement. The device at one end of the
an end-to-end connection before any data is sent. Connection-
communication transmits data to the other, without first
oriented protocol service is sometimes called a "reliable"
ensuring that the recipient is available and ready to receive the
network service, because it guarantees that data will arrive in
data. The device sending a message simply sends it addressed
the proper sequence. Transmission Control Protocol (TCP) is a
to the intended recipient. If there are problems with the
connection-oriented protocol.
transmission, it may be necessary to resend the data several
times. The Internet Protocol (IP) and User Datagram Protocol
(UDP) are connectionless protocols.

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 Transmission Control Protocol (2)

Sequence Number = TCP uses a 32-bit sequence number that

counts bytes in the data stream.
URG = URG flag indicating valid data in the Urgent Pointer TCP Header
field
ACK = Whenever a TCP packet is sent, an acknowledgment,
or ACK, is returned
PSH = Push function (High priority data for the application)
RST = Reset the connection
SYN = sends a synchronization
FIN = No more data from sender
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 Transmission Control Protocol (3)
Data Offset:
Number of 32-bit words in the header.
Reserved (6 bits):
Reserved 6 bits for future use.
Flags (6 bits):
URG: set when the urgent data sends
ACK: for acknowledgement
PSH: all the data to be sent is segmented by the sender.
RST: reset connection
SYN: first segment transmitted with SYN flag set.
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 Transmission Control Protocol (4)

FIN: no more data from sender.

URGENT POINTER:sequence no of last octet in case of urgent data,
Indicates the data offset in no.of bytes.

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 Connection establishment
To establish a connection, the three-way (or 3-step) handshake occurs:

The active open is performed by sending a SYN to the server.

In response, the server replies with a SYN-ACK.
Finally the client sends an ACK (usually called SYN-ACK-ACK) back to the server.
At this point, both the client and server have received an acknowledgement of the connection.

Example:

The initiating host (client) sends a synchronization (SYN flag set) packet to initiate a connection. Any SYN packet
holds a Sequence Number. The Sequence Number is a 32-bit field in TCP segment header. For example let the
Sequence Number value for this session be x (Send SYN seq=x).

The other host receives the packet (Receive SYN segment), records the Sequence Number of x from the client, and
replies with an acknowledgment and synchronization (SYN-ACK)…. Receive SYN +ACK segment

The Acknowledgment Number is a 32-bit field in TCP segment header. It contains the next sequence number
that this host is expecting to receive (x + 1)…. ACK x+1

The host also initiates a return session. This includes a TCP segment with its own initial Sequence
Number value of y (Send SYN seq=y). The initiating host responds with a next Sequence Number (x+1) and a simple
Acknowledgment Number value of y + 1, which is the Sequence Number value of the other host + 1.

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 Transmission Control Protocol (5)

Client Host Client Host

Send SYN seq=x Send FIN seq=x

Receive SYN segment Receive FIN segment
Send SYN seq=y, Send ACK x+1
Receive ACK segment
Receive SYN ACK x+1
+ACK segment Send FIN seq=y,
Receive FIN ACK x+1
+ ACK segment
Send ACK y+1
Send ACK y+1
Receive ACK segment Receive ACK segment

Establishing a TCP Connection Closing a TCP Connection

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 Connection termination

The connection termination phase uses, at most, a four-way

handshake, with each side of the connection terminating
independently. When an endpoint wishes to stop its half of
the connection, it transmits a FIN packet, which the other
end acknowledges with an ACK. Therefore, a typical
teardown requires a pair of FIN and ACK segments from
each TCP endpoint.

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 Transmission Control Protocol (6)

Client Host

Send Packet 1
Start Timer Packet Lost Packet should arrive
Timer ACK should be sent
ACK would normally
Arrive at this time

Time Expires

Retransmit Packet1
Timer Start Timer Receive Packet 1
Send AXK 1
Receive ACK 1
Cancel Timer

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 User Datagram Protocol

UDP Header

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 Main Contents
• Overview of TCP/IP
• Internet Protocol.
• VoIP Network
• VoIP Media Protocols (RTP and RTCP)
• VoIP Signaling Protocols
– H.323 Protocol Stack
– SIP and SDP
– Stream Control Transmission Protocol (SCTP)
– Media Gateway Control Protocol (MGCP)
• Quality of Service
• IP Telephony Market
• Lab Practice

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 Main Contents
• Overview of TCP/IP
 Internet Protocol.
• VoIP Network
• VoIP Media Protocols (RTP and RTCP)
• VoIP Signaling Protocols
– H.323 Protocol Stack
– SIP and SDP
– Stream Control Transmission Protocol (SCTP)
– Media Gateway Control Protocol (MGCP)
• Quality of Service
• IP Telephony Market
• Lab Practice

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 Internet Addresses Or IP Addresses (1)
• Internet addresses are logical addresses, and are
independent of any particular hardware or network
component.
• Any machine desirous of communicating over the Internet
must have a unique IP address.
• It consists of a 4 byte (32-bit) numeric value which
identifies the network number and the device number on
the network.
• For ease of comprehension the 4 byte IP address is
represented in dotted decimal notation, where each byte
represents a value between 0 and 255, e.g., 127.46.6.11
Logical Addresses =In computer networks, a logical address
refers to a network layer address such as an IP address, an
X.25/X.121 or IPX address.
In computer architectures, a logical address or virtual address
is a memory location accessed by an application program in a
system with virtual memory such that intervening computer
hardware and/or software maps the virtual address to real
(physical) memory.
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 Understanding IP Addressing

Every computer that communicates over the Internet is assigned an IP address that
uniquely identifies the device and distinguishes it from other computers on the
Internet. An IP address consists of 32 bits, often shown as 4 octets of numbers
from 0-255 represented in decimal form instead of binary form. For example, the

IP address 168.212.226.204

in binary form is

10101000.11010100.11100010.11001100.

But it is easier for us to remember decimals than it is to remember binary numbers, so we

use decimals to represent the IP addresses when describing them. However, the binary
number is important because that will determine which class of network the IP address
belongs to. An IP address consists of two parts, one identifying the network and one
identifying the node, or host. The Class of the address determines which part belongs to the
network address and which part belongs to the node address. All nodes on a given network
share the same network prefix but must have a unique host number.

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 Internet Addresses Or IP Addresses (2)
• Domain Names. In addition, computers in a TCP/IP network each
have unique logical names like www.cnn.com. These logical
names are connected to their IP address, in this example, the IP
address of www.cnn.com is 207.25.71.5. The logical name is also
referred to as the domain name.
• When a client computer wishes to communicate with the host
computer www.cnn.com, it must translate its logical name into its
IP address. It does this via a domain name lookup query, which
asks a domain name server the IP address of the destination host.
• The IP logical numbering is comprised of a network number and
a local number. For sites connected to the Internet, the network
portion is assigned by applying to a company responsible for
maintaining the Internet.

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 Domain name
• Domain names are used in URLs to identify particular Web pages. For
example, in the URL http://www.pcwebopedia.com/index.html, the
domain name is pcwebopedia.com. Every domain name has a suffix
that indicates which top level domain (TLD) it belongs to. There are
only a limited number of such domains. For example:
• gov - Government agencies
• edu - Educational institutions
• org - Organizations (nonprofit)
• mil - Military
• com - commercial business
• net - Network organizations
• ca - Canada
• th - Thailand
InterNIC or Internet Network Information Center was the
Internet governing body primarily responsible for domain
name and IP address allocations until September 18, 1998
when this role was assumed by the ICANN body. ICANN is the
Internet Corporation for Assigned Names and Numbers.

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 Glassful IP Addressing
• When IP was first standardized in September 1981, the
specification required that each system attached to an IP-
based internet be assigned a unique, 32-bit Internet address
value.
• Some systems, such as routers which have interfaces to
more than one network, must be assigned a unique IP
address for each network interface.
The first part of an Internet address identifies the network
on which the host resides, while the second part identifies
the particular host on the given network. This created the
two-level addressing hierarchy which is illustrated in Figure

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 Classes of IP Address (1)
Class A Addressing
 first byte specifies the network portion

 remaining bytes specify the host portion

 the highest order bit of the network byte is always 0

 From 1.0.0.0 to 126.0.0.0; network values of 0 and 127 are reserved

 there are 126 class A networks


there are more than 16 million host values for each class A network

binary address start with 0, therefore the decimal number can be anywhere from 1 to 126. The first 8 bits (the first octet) identify the network
and the remaining 24 bits indicate the host within the network. An example of a Class A IP address is 102.168.212.226, where "102" identifies
the network and "168.212.226" identifies the host on that network.

1byte 3 bytes

Network
0 Host Portion
Address

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 Classes of IP Address (2)
Class B Addressing
 the first two bytes specify the network portion
 the last two bytes specify the host portion
 the highest order bits 6 and 7 of the network portion are 10
 there are more than 16 thousand class B networks
 there are 65 thousand nodes in each class B network
 From 128.0.0.0 to 191.255.0.0

binary addresses start with 10, therefore the decimal number can be anywhere from 128 to 191.
(The number 127 is reserved for loopback and is used for internal testing on the local machine.)
The first 16 bits (the first two octets) identify the network and the remaining 16 bits indicate the
host within the network. An example of a Class B IP address is 168.212.226.204 where "168.212“
identifies the network and "226.204" identifies the host on that network.

2 bytes 2 bytes

Network
10 Address Host Portion

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 Classes of IP Address (3)
• Class C Addressing
 the first three bytes specify the network portion
 the last byte specifies the host portion
 the highest order bits 5, 6 and 7 of the network portion are 110
 there are more than 2 million class C networks
 there are 254 nodes in each class C network
 From 192.0.0.0 to 223.255.255.0
binary addresses start with 110, therefore the decimal number can be anywhere from 192 to 223.
The first 24 bits (the first three octets) identify the network and the remaining 8 bits indicate the
host within the network. An example of a Class C IP address is 200.168.212.226 where "200.168.212“
identifies the network and "226" identifies the host on that network.

3 bytes 1 byte

110 Network Address Host Portion

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 Limitations of Classful Addressing

• Injudicious allocation of IP addresses.

• Standardization on a 32 bit address space which generates
only 4,294,967,296 addresses.
• Although the boundaries of Classful addressing were easy
to understand and implement, but resulted in inefficient
allocation of finite address space.

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 Problems with 2 Level Classful Hierarchy

• Internet routing tables were beginning to grow

exponentially resulting in difficulties of routing the traffic
over ever expanding global Internet.
• Local administrators had to request another network
number from the Internet before a new network could be
installed at their site.

• Eventual exhaustion of IP addresses.

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 Solution:
Three level Subnet Hierarchy
• Subnetting is a procedure for dividing a single class A,B,
or C network number into smaller pieces. It was
introduced in 1985.
• The basic idea of subnetting is to divide the standard
classful host-number field into two parts - the subnet-
number and the host-number on that subnet.

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 Subnetting
• The route from the Internet to any subnet of a given IP address is the same, no
matter which subnet the destination host is on.
• This is because all subnets of a given network number use the same network-
prefix but different subnet numbers, all of the subnets in the organization are
collected into a single routing table entry.
• This allows the local administrator to introduce arbitrary complexity into the
private network without affecting the size of the Internet's routing tables.
• Subnet structure is never visible outside of organization’s private
network.

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 Benefits of Subnetting
• The size of the global Internet routing table does not grow because the site administrator does not need
to obtain additional address space and the routing advertisements for all of the subnets are combined
into a single routing table entry.
• The local administrator has the flexibility to deploy additional subnets without obtaining a new
network number from the Internet.
• Route flapping (i.e., the rapid changing of routes) within the private network does not affect the
Internet routing table since Internet routers do not know about the reachability of the individual
subnets - they just know about the reachability of the parent network number.

Routing Tables: A routing table is a set of rules, often viewed in table format, that is used to determine where data packets traveling
over an Internet Protocol (IP) network will be directed. All IP-enabled devices, including routers and switches, use routing tables.

RIP =Routing Information Protocol

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 Extended Network Prefix- Subnet Mask (1)

• Internet routers use only the network-prefix of the destination

address to route traffic to a subnetted environment. Routers
within the subnetted environment use the extended-network-
prefix to route traffic between the individual subnets. The
extended-network-prefix is composed of the classful network-
prefix and the subnet-number known as subnet mask.

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 Extended Network Prefix- Subnet Mask (2)

• For example , if you have the address of 130.5.0.0 and you want
to use the entire third octet to represent the subnet-number, you
need to specify a subnet mask of 255.255.255.0. The bits in the
subnet mask and the Internet address have a one-to-one
correspondence.

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 IP Versions
The Internet Protocol has many versions in use. Each version
has its own definition of an IP address.
• IP v 1-3 defined and replaced
• IP v4 - current version
• IP v5 - streams protocol
• IP v6 - replacement for IP v4
–During development it was called IPng

–Next Generation

Just as there are addresses for private, or internal networks in IPv4, there are blocks of addresses set aside in IPv6 for private addresses.
With IPV6, virtually every device in the world can have an IP address: cars, fridges, lawnmowers and so on. If one's fridge stopped working,
for example, a repair specialist could identify the problem without ever visiting in person. It might even be possible to make repairs from abroad,
depending on the severity of the problem.

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 Why Change IP ?

• Address space exhaustion

 Two level addressing (network and host) wastes space
 Network addresses used even if not connected to Internet
 Growth of networks and the Internet
 Extended use of TCP/IP (Mobile IP & CATV receivers)
 Single address per host
• Requirements for new types of service
 Multimedia applications

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 Main Contents
• Overview of TCP/IP
• Internet Protocol.
• VoIP Network
• VoIP Media Protocols (RTP and RTCP)
• VoIP Signaling Protocols
– H.323 Protocol Stack
– SIP and SDP
– Stream Control Transmission Protocol (SCTP)
– Media Gateway Control Protocol (MGCP)
• Quality of Service
• IP Telephony Market
• Lab Practice

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 Main Contents
• Overview of TCP/IP
• Internet Protocol.
 VoIP Network
• VoIP Media Protocols (RTP and RTCP)
• VoIP Signaling Protocols
– H.323 Protocol Stack
– SIP and SDP
– Stream Control Transmission Protocol (SCTP)
– Media Gateway Control Protocol (MGCP)
• Quality of Service
• IP Telephony Market
• Lab Practice

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 IP Telephony Model
Gateways allow PCs
Public Switched to also reach phones
Telephone Network

PSTN
Initially, PC to PC
voice calls over the
Internet Gateway

Multimedia
PC

IP Network
Gateway
Multimedia
PC

PSTN …or phones to reach

phones

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 Why VoIP ?

 Allows voice to be an IP Application

 Converged Networks
Voice, Video & Data over an IP network

 Monetary Savings

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 TCP/IP
 Some widely used application layer services are
TELNET , FTP , HTTP , SMTP , SNMP etc.
&
H.323 , SIP , MGCP ,Megaco/H.248
 In the next slides , transport layer protocols will be
discussed.
 However, at Network Layer Internet Protocol is used.
 Protocols defined by underlying networks not by TCP/IP.
Telnet = A terminal emulation program for TCP/IP networks such as the Internet. The Telnet program runs on your computer and connects your PC to a server
on the network. You can then enter commands through the Telnet program and they will be executed as if you were entering them directly on the server console.
This enables you to control the server and communicate with other servers on the network. To start a Telnet session, you must log in to a server by entering a
valid username and password. Telnet is a common way to remotely control Web servers.
FTP = File Transfer Protocol
HTTP = hypertext transfer protocol

SMTP = simple mail transfer protocol

SNMP = Simple Network Management Protocol

H.323 = H.323 is a standard that specifies the components, protocols and procedures that provide multimedia communication services—real-time audio, video,
and data communications—over packet networks, including Internet protocol (IP)–based networks. H.323 is part of a family of ITU—T recommendations
called H.32x that provides multimedia communication services over a variety of networks
SIP = Short for Session Initiated Protocol, or Session Initiation Protocol, an application-layer control protocol; a signaling protocol for Internet Telephony.
MGCP = Media Gateway Control Protocol, developed by Telcordia and Level 3 Communications, a control and signal standards to compete with the older

Megaco = Megaco (a contraction of "Media Gateway Controller") is a signalling protocol, used between a Media Gateway and a Media Gateway Controller
(also known as a Call Agent or a Soft Switch) in a VoIP network. It defines the necessary signalling mechanism to allow a Media Gateway Controller
(Call agent) to control gateways in order to support voice/fax calls between PSTN-IP or IP-IP networks.
The protocol is defined by IETF RFC 3525 and was the result of joint work of IETF and ITU. It is also known as H.248. H.248 is the name given to it by the ITU,
Megaco is the IETF name.

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 Functional Elements

In IP-Telephony networks, following functional elements are

mostly found.

• Media Gateway (MG)

• Media Gateway Controller (MGC)

• Signaling Gateway (SG)

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 IP Telephony Model

SS7 SS7
SG SG
SS

IP
7

r
ov

ve
er

7o
IP

SS
PSTN PSTN
MGC
48

Node Node

M
.2

GC
/H

P
CP

/H
G

. 24
M

8
RTP / UDP / IP
PSTN
MG MG
PSTN
Trunks Trunks
IP Network

RTP =Real Time Transfer Protocol

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 Media Gateway

Media Gateway performs functions like

• Physical T1/E1 facility termination

• Voice Packetization
• Voice Compression using ITU recommendations
G.711,G.723.1 and G.729 etc

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 Media Gateway Controller
(MGC)
• A media gateway controller is the overall controller of the system.
• Media Gateways receives commands from the media gateway controller which define their
operations.
MGC performs core functions like
– Address Resolution (Address Resolution Protocol (ARP) is a protocol for mapping an Internet Protocol address (IP
address) to a physical machine address that is recognized in the local network.)

– Call Routing
– Resource Management
– Call Detail Record (CDR) Generation
– many more….

• A media gateway controller is also known as a Soft Switch

• In the presentation, the terms Soft switch , MGC and Call Manager will be used interchangeably.
• It handles CDR (Call Detail Record Generation)

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 Signaling Gateway (SG)

• Signaling gateway transfers signaling information between

PSTN node and Media gateway controller (MGC)

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 Media Gateway
Media Gateway Controller
Signaling Gateway

May be separate physical devices

or
Integrated in any combination
e.g. SG/MG functions
Implemented in one unit
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 H.323 = H.323 is a standard that specifies the

MGC
components, protocols and procedures that
provide multimedia communication services—
real-time audio, video, and data communications
—over packet networks, including Internet protocol
(IP)–based networks. H.323 is part of a family of
ITU—T recommendations called H.32x that provides
multimedia communication services over a variety
of networks

H.32
3 SIP = Short for Session Initiated Protocol,
or Session Initiation Protocol, an application-layer
control protocol; a signaling protocol for Internet

SIP
Telephony.

BICC Bearer Independent Call Control (BICC) is a

signaling protocol based on N-ISUP that is used
to support narrowband ISDN service over a
broadband backbone network without interfering

MGC
with interfaces to the existing network and e
nd-to-end services. Specified by the
International Telecommunications Union –
Telecommunications Standardization Sector
(ITU-T) in recommendation Q.1901, BICC
was designed to be fully compatible with
existing networks and any system capable
of carrying voice messages.

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 Decomposed Gateways

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 Conventional Gateways

In a conventional gateway the same device that provides

the media conversion from TDM to packet also provides

the signaling interworking

H.323, SIP
Gateway RTP Gateway

RTP =Real Time Transfer Protocol

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 Decomposed Gateways

A new class of gateways has been proposed that splits the

media-handling function from the signaling function

These decomposed gateways assign media function to a

Media Gateway & control function to a Media Gateway

Controller

In some quarters, this separation b/w call control &

media is known as Soft Switch Architecture
So, the MGC is also known as Soft Switch.

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 Decomposing Gateways.

MGC
Signaling
Path

Media
Path MG

Gateway Decomposed Gateways

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 Decomposed Gateways

Several control protocols exist b/w the controllers and the

media devices.
The control protocol meet widely deployed in today’s VOIP
networks is Media Gateway Control Protocol (MGCP)
developed by IETF.
MGCP has been superseded by a protocol known as
MEGACO/H.248 which is jointly developed by IETF & ITU.
In fact, it is known as MEGACO in the IETF commonly & as
H.248 within the ITU.
The terms MEGACO and H.248 are interchangeable.

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 Separation off Media and Call Control

Call Agent or SIP Call Agent or

Media Gateway H.323 Media Gateway
Controller Controller
(MGC) (MGC)

Media Gateway Control Protocol, (Megaco) MGCP, MGCP,

Megaco Megaco

Media
Media
Media
Media Media over
Media Gateway Media Media Gateway over
over over
over (MG) overIP
IP (MG) IP
IP
IP
IP

Media Gateway Media Gateway

(MG) (MG)
Institute of Communication Technologies , Islamabad. 70
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 SIP   H.323
     
PHILOSOPHY

"New World" - a relative of Internet protocols "Old World" - complex, deterministic and
 
- simple, open and horizontal vertical
IETF   ITU
Carrier-class solution addressing the wide Borne of the LAN - focusing on enterprise
 
area conferencing priorities
CHARACTERISTICS
A simple toolkit upon which smart clients and H.323 specifies everything including the
applications can be built. It re-uses Net   codec for the media and how you carry the
elements (URLs, MIME and DNS) packets in RTP
Leaves issues of reliability to underlying Assumes fallibility of network - an
 
network unnecessary overhead
SIP messages are formatted as text. (Text Binary format doesn't sit well with the
 
processing lies behind the web and email) internet - this adds complexity
SIP allows for standards-based extensions to Extensions are added by using vendor-
 
perform specific functions. specific non-standard elements
Hierarchical URL style addressing scheme
  Addressing scheme doesn't scale well
that scales
Minimal delay - simplified signalling scheme
  Possibilities of delay (up to 7 or 8 seconds!)
makes it faster
Slim and Pragmatic   The suite is too cumbersome to deploy easily

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 Main Contents
• Overview of TCP/IP
• Internet Protocol.
• VoIP Network
• VoIP Media Protocols (RTP and RTCP)
• VoIP Signaling Protocols
– H.323 Protocol Stack
– SIP and SDP
– Stream Control Transmission Protocol (SCTP)
– Media Gateway Control Protocol (MGCP)
• Quality of Service
• IP Telephony Market
• Lab Practice

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 Main Contents
• Overview of TCP/IP
• Internet Protocol.
• VoIP Network
 VoIP Media Protocols (RTP and RTCP)
• VoIP Signaling Protocols
– H.323 Protocol Stack
– SIP and SDP
– Stream Control Transmission Protocol (SCTP)
– Media Gateway Control Protocol (MGCP)
• Quality of Service
• IP Telephony Market
• Lab Practice

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 UDP Vs TCP
• Which one is the most suitable VoIP transport
Protocol?
• UDP is simple ,fast ,connection-less and unreliable
• TCP is slow, reliable and connection-oriented.

TCP Protocol UDP Protocol

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 UDP Vs TCP

• A late retransmission of voice samples is of no use.

• Latency is more important than ensuring the delivery
of each voice packet.
• Therefore ,UDP is the logical choice to carry voice.
• However, TCP is the logical choice for signaling due
to its reliability.

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 Real Time Protocol (RTP)

→ Although , UDP is the logical choice to carry voice but in

voice networks , some additional information is required.

→ RTP gives receiving stations information that is not in the

connectionless UDP/IP streams

→ RTP is the standard for transmitting Delay-sensitive

traffic (voice, video) across packet-based networks.

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 Real Time Protocol (RTP)

→ RTP provides this additional information.

1. Payload Type Identification (PT)

2. Sequence Numbering (SN)
3. Time Stamping
4. Delivery Monitoring etc

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 RTP Header

.. . . … … PT Sequence Number
(SN)
Time Stamp

Data (Variable)

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 Sequence Number
Whether the packets are arriving in proper sequential
orders
& to determine if any traffic is missing/lost
Time-stamp
The time-stamping information to determine the Inter-
arrival packet time (jitter)

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 Payload Type (PT)

Indicates the type of traffic that is carried in

the data field of the RTP message

PT Encoding Name
0 G.711(Audio)
2 G.721(Audio)
4 G.723 (Audio)
26 H.261 (Video)
Many Many more

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 Real Time Control Protocol (RTCP)
 RTCP is a companion protocol to RTP

 RTCP does not carry coded voice packets

 RTCP messages provide feedback regarding the quality of

service

 The type of information includes such details as Lost

Packets , Delay etc.

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 Real Time Control Protocol (RTCP)

 Whenever an RTP session is opened , an RTCP session is

also implicitly opened.
 This means that, when a UDP port number is assigned to
an RTP session for the transfer of voice packets, a separate
port number is assigned for RTCP messages
 An RTP port number will always be even , and the
corresponding RTCP port will be the next higher
number, and hence odd

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 IP Protocol Stack for Voice Transport

Voice Application

RTP, RTCP

UDP

IP

Layer 2-Data Link

Layer 1-Physical

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Voice Bits RTP Header UDP Header IP Header

Bytes 12 Bytes 8 Bytes 20 Bytes

40 Bytes

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 Compressed Real Time Protocol (cRTP)

To avoid the unnecessary consumption of available

bandwidth cRTP compresses 40 bytes IP/RTP/UDP header
to 2 to 5 bytes .

2 to 5 bytes Payload

IP/UDP/RTP Header

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 Voice Codecs
• Most popular voice coding standards for telephony and
packet voice ( G- series recommendations of ITU-T) are
– G.711
– G.723
– G.726
– G.728
– G.729
– many more

A codec is a device or program capable of performing encoding and decoding on a digital data stream or signal.
The word codec may be a combination of any of the following: 'Compressor-Decompressor', 'Coder-Decoder',
or 'Compression/Decompression algorithm'.

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 Voice Codecs

• G.711 recommendation specifies the the PCM for

encoding voice signals. In this case there is no voice
compression and the transmission bit rate is 64 Kbits/s
• G.723 recommendation describes a compression technique
to compress speech at 5.3 Kbits/s and 6.3 Kbits/s
• G.729 recommendation describes the algorithm for coding
speech signals at 8 Kbits/s

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 Mean Opinion Score (MOS)

• Mean opinion score (MOS) is a benchmark for quantifying the

performance of the speech codecs

• MOS tests are given to a group of listeners

• Because voice quality and sound in general are subjective to

listeners, it is important to get a wide range of listeners and
sample material when conducting a MOS

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 Mean Opinion Score (MOS)

• The listeners give each sample of speech material a rating

of
• 1 (bad ) to
• 5 (excellent)
• The scores are then averaged to get the mean opinion score

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 Main Contents
• Overview of TCP/IP
• Internet Protocol.
• VoIP Network
• VoIP Media Protocols (RTP and RTCP)
• VoIP Signaling Protocols
– H.323 Protocol Stack
– SIP and SDP
– Stream Control Transmission Protocol (SCTP)
– Media Gateway Control Protocol (MGCP)
• Quality of Service
• IP Telephony Market
• Lab Practice

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 Main Contents
• Overview of TCP/IP
• Internet Protocol.
• VoIP Network
• VoIP Media Protocols (RTP and RTCP)
• VoIP Signaling Protocols
 H.323 Protocol Stack
– SIP and SDP
– Stream Control Transmission Protocol (SCTP)
– Media Gateway Control Protocol (MGCP)
• Quality of Service
• IP Telephony Market
• Lab Practice

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 VoIP Signaling Protocols
In all telephony networks, specific signaling protocols are
invoked

 before and during a call to communicate

 a desire to setup a call,
 to monitor call progress,
 and to gracefully bring a call to conclusion

Best example is ISUP (SS7)

SS7 signaling system is used both in PSTN (Public Switched Telephone Network) and ISDN (Integrated Services Digital Network). 
ISUP is the ISDN User Part used for SS7 signalling. It's the most widely used signaling. ISUP = ISDN User Part

In VOIP system, signaling protocols also need to be used

for exactly the same reasons

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 VoIP Protocols
VOIP Signaling protocols

to establish presence, locate users, set up, modify and tear

down sessions

VOIP Media Transport Protocols

for transmission of packetized audio/video

VOIP Supporting Protocol

QoS, interdomain AAA (Authentication, Authorization and

Accounting), Address translation (Transforming one address
into another) etc
Institute of Communication Technologies , Islamabad. 93
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 VoIP Protocols

VOIP Signaling Protocols

H.323, SIP, MGCP, Megaco/H.248

VOIP Media Transport Protocols

RTP (IETF,s, adopted by ITU-T)

VOIP Supporting Protocols

RTCP,SDP
SIP = Session Initiation Protocol
MGCP = Media Gateway Control Protocol
Megaco = Media Gateway Controller
RTP = Real-time Transport Protocol
IETF = Internet Engineering Task Force
ITU-T = Telecommunication Standardization Sector of the International Telecommunications Union
RTCP = Real Time Control Protocol
SDP = Service Discovery Protocol

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 Call Processing Layered Model

The DNS and the Web support Operations

Call Processing Media Transport Support
Protocols Protocols Protocols

H.323,SIP Voice ,Video RTCP, SDP

Data
MGCP, Megaco
RTP

L-4 TCP, TCP, UDP

TCP, UDP UDP
UDP
L-3
IP

Data Link and Physical Layer

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 H.323
 The first successful set of protocols for VOIP was
developed by ITU-T. This set of protocol is known as H.323.

 Although new protocols have emerged e.g. SIP . H.323 is

still the most widely developed VOIP signaling system.

 Our focus is goanna be on

 H.323 Network Architecture

 H.323 Protocols

 H.323 Call Establishment

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 H.323 Network Architecture

H.323 network components are:

- H.323 Terminals

- Gateways

- Gatekeepers

- Multipoint Control Unit (MCU)

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 H.323 Architecture (Components)

H.323 Gatekeeper

PSTN
H.323
Terminals H.323
Gateway

H.323
Multipoint Control Unit
H.323 Zone

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 H.323 Terminal
LAN

 It is typically an end-user communications device

Endpoint on a LAN

 It supports at least one audio codec (G.711, G.729 etc)

& may optionally support video or data codecs

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 Multipoint Control Unit (MCU)

H.323
Multipoint Control Unit

1) a. MCU Allows three are more endpoints to participate in a

multipoint conference MCU can be a stand-alone device
or

b. integrated into a Gateway, Gatekeeper or H.323 terminal

2) H.323 MCU is an optional Component. MCU are not

needed unless we need multipoint conferencing

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 H.323 Gateway
PSTN
H.323 Network
H.323
Gateway

H.323 Gateway provides translation services between

H.323 network

&

another type of network such as PSTN, ISDN

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 H.323 Gateway

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 H.323 Gateway

On the H.323 side the gateway has the xtics of an H.323

terminal

On the circuit- switched side the gateway has the xtics of

a node in the circuit- switched network

A translation between the signaling protocols & media

formats of one side & those of the other side is performed
internally within the gateway

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 H.323 Gateway

Gateway
Multimedia
PSTN PC
H.323 Network

SS7 Signaling < = = > H.323 Signaling

TDM Voice < = = > Packetised Voice

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 H.323 Gateway

1. H.323 Gateway is an optional component.

2. Gateway are not needed unless interconnection of

H.323 networks with other networks (PSTN) is
required.

3. H.323 terminals can communicate directly over the

packet network without connecting to a gateway

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 H.323 Gatekeeper
H.323 Gatekeeper

1. H.323 Gatekeeper is an optional component

2. When present, it controls a collection of H.323 terminals,

gateways and MCUs the set of terminals, gateways,&
MCUs controlled by a single gate keeper is known as a
Zone.

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 H.323 Gatekeeper

It performs the following (some of) functions if present in an H.323 system

Bandwidth control/ management

Enables the gatekeeper to reject admission if the required bandwidth is not
available

Call Authorization
Enables the gatekeeper to restrict admission to certain terminals

Call Management
services include maintaining an active call list to indicate
that an endpoint is busy

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 H.323 Protocol/Protocol Stack

Audio / Video
Application Terminal / Application Control

Audio / Video H.225 H.245

H.225
Codecs RTCP Call Control
RAS
Signaling Signaling
RTP Signaling
UDP TCP
IP
Layer 2- Data Link
Layer 1-Physical
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 Internet-Phone Call Using a PC Application

1) Judy launches her Internet 2) Bob already has his I-

Phone (I-phone) application phone application
which is H.323 compatible launched

PictureTel P ic tureTel

2
PictureTel P ic tureTel

1 &3
Judy
Bob
3) Judy knows that Bob’s Internet “name” is
Bob.nextdoorneighbour.com, so she puts that into the
“who to call” section in her I-phone application and
presses return
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 Internet-Phone Call Using a PC Application
4) The I-phone
application gives Domain Name Server(DNS)
Bob.nextdoorneigh
bour.com to a DNS
machine to resolve
the DNS name and 5) The DNS
get an actual IP machine passes
address back Bob’s IP
address
P ic tureTel

2
P ic tureTel P ic tureTel

P ictureTel

1 &3
Judy 6) Judy’s I-phone application
Bob
takes Bob’s IP address and sends
an H.225 message to Bob
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 Internet-Phone Call Using a PC Application

4
5 7 ) H.225 message signals
Bob ’PC to begin ringing

Pic ture Tel

Picture Tel PictureTel

1 &3 2 ,7
PictureTel

Judy 6
Bob

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 Internet-Phone Call Using a PC Application

8) Bob clicks on the

4 Accept button, which tells
his I-phone application to
5
send back an H.225
connect message

P ic tureTel

Pic ture Tel PictureTel

1 &3 2 ,7 & 8
P ic tureTel

Judy 6
Bob

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 Internet-Phone Call Using a PC Application

4
5

Pic ture Te l

Pic ture Tel

6 P ictur eTe l

1 &3 2 ,7 & 8
P ic ture Te l

Judy
9) Judy’s I-phone Bob
application then begins
H.245 negotiation with Bob’
PC

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 Internet-Phone Call Using a PC Application

4
5

Pic ture Te l

Pic ture Tel

6 P ictur eTe l

1 &3 2 ,7 & 8
P ic ture Te l

9
Judy
Bob
10) H.245 negotiation finishes
and logical channels are opened.
Bob and Judy can now speak to
one another through a packet-
based network
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 H.323 Protocol Stack

In H.323 there are two protocols which have not yet

been discussed
6) Judy’s I-phone application takes Bob’s IP
(A). H.225
address and sends an H.225 message to Bob
7 ) H.225 message signals Bob ’PC to begin
(B). H.245 ringing
8) Bob clicks on the Accept button, which tells his
I-phone application to send back an H.225
connect message
9) Judy’s I-phone application then begins
H.323 H.245 negotiation with Bob’ PC
10) H.245 negotiation finishes and logical
H.225 H.245 channels are opened. Bob and Judy can now
speak to one another through a packet-based
network

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 H.323
(A). H.225
H.225 H.245
H.225 is a two-part protocol

1. H.225 RAS Signaling

Provides pre-call control in H.323 gatekeeper-based

networks

2. H.225 Call signaling

Used to connect, maintain, and disconnect calls between

endpoints

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 1. H.225 RAS Signaling
H.225 RAS SIGNALLING
Areas of RAS Signaling include
Gatekeeper discovery
 Gatekeeper discovery

 Registration Registration

 Admissions Admission

 Status Information
Status Information

 Bandwidth Control
Bandwidth Control

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 Gatekeeper Discovery
H.225 RAS SIGNALLING
Gatekeeper discovery is a manual or automatic process

In the manual discovery method Gatekeeper discovery

Endpoints are configured with the Gatekeeper's IP address Registration

In the auto discovery method

Admission
Endpoints discover gatekeeper through multicast messages

Status Information

Multicast is the delivery of information to a group of destinations simultaneously using

the most efficient strategy to deliver the messages over each link of the network
Bandwidth Control

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 Registration
H.225 RAS SIGNALLING

Registration is the process that enables Gateways,

Gatekeeper discovery
Endpoints, and MCUs to join a zone & inform the

gatekeeper of their IP and alias addresses Registration Registration

occurs after the discovery process

Admission

A name, usually short and easy to remember and type, that is translated
into another name or string, usually long and difficult to remember or type.
Status Information

Bandwidth Control
MCU = Multi Point Control Unit

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 Admissions
H.225 RAS SIGNALLING

Admission messages between endpoints and gatekeepers

Provide the basis for call admission Gatekeeper discovery

 ARQ: An attempt by an endpoint to iniate a call

Registration
 ACF: An authorization by the gatekeeper to admit a call

 ARJ: Denies the endpoint's request to gain access to the Admission

network for this particular call

ACF message contains the IP address of the terminating Status Information

endpoint and enables the originating endpoint to initiate
call signaling
Bandwidth Control

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 Status Procedures
H.225 RAS SIGNALLING

Status information messages are used by the gatekeeper

Gatekeeper discovery
to monitor whether the endpoint is online and offline

The following messages provides status information Registration

 Information Request (IRQ)

Admission
 Information Request Response (IRR)

Status Information

Bandwidth Control

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 Bandwidth Control
H.225 RAS SIGNALLING

Enables the gatekeeper to reject Gatekeeper discovery

admission if the required

Registration
bandwidth is not available

Admission

Status Information

Bandwidth Control

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 2. H.225 Call Signaling

H.225 Call Signaling is effectively a variant of ITU-T recommend.

.
Q.931, the ISDN layer 3 specification

It is used for the establishment & tear down of connections

between H.323 endpoints

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 H.323
(B). H.245 Signaling
H.225 H.245

Following procedures and messages are used to

enable H.245 control operations

 Capability Exchange H.245 provides a capability exchange functionality to allow the negotiation
of capabilities and to identify a set of features common to both endpoints.

 Master-Slave Determination

 Round-Trip Delay

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 H.323 Call Setup Example (with one Gatekeeper)
Call Initiation
Gatekeeper (GK) (2a) GK resolves “Bob”
to IP address through H.323
registration or external name service
(1) ARQ (An attempt by (e.g. DNS etc..)
an endpoint to initiate a (2b) Admission Policy Applied
call)
Can I call “Bob”?
(3) ACF (An authorization by
the gatekeeper to admit a call) PictureTel
PictureTe l

Pic tureTe l

Yes, use this IP Address

Bob
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 H.323 Call Setup Example (with one Gatekeeper)
Call Connection
Gatekeeper
(6) ARQ (An attempt by an endpoint
to initiate a call)

May I answer?
(7) ACF (An
authorization by the gatekeeper to

admit a call) Yes

(4) Q.931SETUP (the ISDN layer 3 specification)

PictureTel
P ictureTel

PictureTel

(5) Q.931 Call Proceeding

Bill (8) Q.931ALERTING Bob

(9) Q.931 CONNECT
(User answers)
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 H.323 Call Setup Example (with one Gatekeeper)

P ic tureTel

PictureTel
P ic tureTel

(10) H.245 connection established

Bill Bob
- Capability Exchange
- Master/Slave Determination
- Open Logical Channels
- audio, video

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 H.323 Call Setup Example (with one Gatekeeper)

PictureTel
PictureTel

Pic ture Tel

(11) Bi-directional Audio Bob

Bill

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 H.323 and SS7 Call Initiated from the SS7 side

SS7 SS7/H.323 H.323

Exchange Gateway Gatekeeper
IAM (Initial Address Message)
H.225
SETUP
H.245
Operations
H.225
ALERT
ACM (Address Complete Message)
H.225
ANM (answer message) CONN
Two-way voice Two way media flow
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 Main Contents
• Overview of TCP/IP
• Internet Protocol.
• VoIP Network
• VoIP Media Protocols (RTP and RTCP)
• VoIP Signaling Protocols
– H.323 Protocol Stack
– SIP and SDP
– Stream Control Transmission Protocol (SCTP)
– Media Gateway Control Protocol (MGCP)
• Quality of Service
• IP Telephony Market
• Lab Practice

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 Main Contents
• Overview of TCP/IP
• Internet Protocol.
• VoIP Network
• VoIP Media Protocols (RTP and RTCP)
• VoIP Signaling Protocols
– H.323 Protocol Stack
 SIP and SDP
– Stream Control Transmission Protocol (SCTP)
– Media Gateway Control Protocol (MGCP)
• Quality of Service
• IP Telephony Market
• Lab Practice

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 Session Initiation protocol ( SIP)

 SIP is considered by many to be a powerful alternative

to H.323

 Like H.323, SIP is simply a signaling protocol & it

does not carry the voice packets itself

 SIP makes use of services of RTP (Real Time

Transport Protocol) for the transport of voice packets

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 SIP Network Architecture (1)

 SIP defines two basic classes of network entities Clients

& Servers

A client is an application program that sends SIP requests

A server is an entity that responds to those requests. Thus

SIP is a client-server protocol
SIP Requests
CLIENT SERVER
Response to SIP Requests

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 SIP Network Architecture (2)

1. SIP User Agents

2. SIP Registrar

3. SIP Proxy Server

4. SIP Redirect Server

5. SIP Network Servers

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 1. SIP User Agents \ SIP End- Devices (1)

User Agents are User Applications that contain

(a) User-Agent Client (UAC)

(b) User- Agent Server (UAS)

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 SIP User Agents\SIP End- Devices (2)

A SIP device e.g. a SIP-enabled phone will function as

both user-agent client & user-agent server

Acting as a user-agent client, it is able to initiate SIP

requests

Acting as a user-agent server, it can receive & respond to

SIP requests

In practical terms, it means that it is able to iniate call &

receive calls
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 2. SIP Registrar
SIP includes the concept of user registration, whereby
a user signals to the network

it is available at a particular address

A SIP registrar is a server that accepts

SIP REGISTER request from the SIP users

Registrar usually maintain the mapping b/w

SIP URL & network addresses
at the location server database

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 Institute of Communication Technologies , Islamabad. 138
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 3. SIP Proxy Server

 A proxy server acts similarly to a proxy server used for

Web access from a corporate LAN

Client send request to proxy, which either handles those

requests itself or forwards them on to other servers

To those servers, it appears as though the message is

coming from the proxy rather than some entity hidden

behind it

The response to the request will travel through the same path

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 4. SIP Redirect Server

A redirect server is a server that accepts SIP requests, maps

the destination address to the new addresses, & returns the
translated address to the originator of the request .

Therefore , the originator of the request may send requests

to the address (es) returned by the redirect server. A
redirect server does not initiate any SIP request of its own.

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 SIP Addresses

 In a SIP network, every user is given a globally

reachable SIP URL

 SIP URLs resemble e-mail addresses

 SIP URLs are of the format

Sip: username @ domainname

Sip: ciscopress @ Cisco. COM

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 SIP Messages

There are two kinds of SIP messages

(A) Message Requests (initiated by clients)

(B) Message responses (returned from servers)

Every message contains headers that describes the details of

communications

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 (A) SIP Message Requests (SIP METHODS)

INVITE initiates sessions

ACK confirms session establishment

BYE terminates sessions

CANCEL cancels a pending INVITE

REGISTER used by clients to register location information

with SIP server

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 SIP REGISTER
This method is used by clients to register location information
with SIP REGISTRAR

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 SIP BYE message is used to terminate a media session
BYE message carry no bodies

OPTIONS message is used to discover the capabilities of

user agents and SIP servers

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 (B) SIP Message Responses
SIP message responses are sent in response to SIP message
requests

Class of SIP Response Status Code with Explanation

Informational 100 Trying
180 Ringing
181 Call is being forwarded
Success 200 OK
Redirection 302 Moved temporarily
Client-Error 404 Not found
Server-Error 500 server internal error

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 SIP Call Operation (with one Proxy server)
[email protected] Proxy Server [email protected]
Location
Server
1 INVITE 5 Ringing
[email protected] 2 [email protected]
[email protected] 3

4 INVITE
[email protected]
180 RINGING 6
180 RINGING 7
200 OK 8
200 OK 9
10 ACK
11 ACK
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 SIP Call Operation (with Redirect server)
[email protected] Redirect Server [email protected]
Location
Server
1 INVITE
[email protected] 2 [email protected]
[email protected] 3
MOVED 4
[email protected]
5 ACK
6 INVITE
[email protected]
180 RINGING 7
200 OK 8
9 ACK
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 SIP Operation in Proxy Mode

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 Session Description Protocol (SDP)

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 Session Description Protocol (SDP)

We know that SIP is used to establish sessions b/w users

which requires that the users agree on the type & coding of
the information to be shared e.g. the two users must agree on
the voice-coding scheme to be used . Which requires that
the users share session descriptions.

These session descriptions are coded according to the

Session Description Protocol (SDP)

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 Session Description Protocol (SDP)

SDP is simply a language for describing sessions

SDP contains information regarding

 The parties to be involved in a session

 Addresses and port numbers to be used

 The date and time when the session is to take

place

 The types of media streams to be shared etc.

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 Session Description Protocol (SDP)

It is possible that a session description could refer to

multiple media streams such as in a video conference

1. Where one media stream relates to coded voice

2. Another media Stream relates to coded video

Consequently SDP is structured So that it can describe

information related to the session as a whole (e.g. the name
of the session) plus information associated with each individual
media stream (e.g. the media format & the application port no.)

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 Session Description Protocol (SDP)

Some of the information included in an SDP session description

will also be included in the SIP message that carries the
SDP description

This overlap is due to the fact that SDP is designed to be used by a

range of other protocols not just SIP

SDP is designed to be used by protocols like SIP, MGCP, Megaco,

HTTP

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 SDP Information Fields

Protocol Version v
Owner/Creator & Session Identifier o
Session Name s
Connection Information c
Time Description Information t
Media Announcements m
Few more …

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 o = <username> <session id> <version> <network type> <address type> < address>

t = <start time> <stop time>

a = rtpmap:<payload type> <encoding name> <clock rate>

m = <media> <port> <transport> <media fmt>

media = audio, video, application, data, & control

port = transport port

transport = transport protocol like RTP/AVP over UDP

media format = As defined in media Payload Type (PT) in

in the RTP Audio/Video profile

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 Main Contents
• Overview of TCP/IP
• Internet Protocol.
• VoIP Network
• VoIP Media Protocols (RTP and RTCP)
• VoIP Signaling Protocols
– H.323 Protocol Stack
– SIP and SDP
– Stream Control Transmission Protocol (SCTP)
– Media Gateway Control Protocol (MGCP)
• Quality of Service
• IP Telephony Market
• Lab Practice

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 Main Contents
• Overview of TCP/IP
• Internet Protocol.
• VoIP Network
• VoIP Media Protocols (RTP and RTCP)
• VoIP Signaling Protocols
– H.323 Protocol Stack
– SIP and SDP
 Stream Control Transmission Protocol (SCTP)
– Media Gateway Control Protocol (MGCP)
• Quality of Service
• IP Telephony Market
• Lab Practice

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 Stream Control Transmission Protocol (SCTP)

Definition
Stream Control Transmission Protocol (SCTP)
[RFC2960] is an end-to-end transport protocol that
provides services heretofore unavailable from either of
the workhorse transport protocols that have supported
the Internet for more than twenty years:
reliable, connection-oriented TCP [RFC793], or
unreliable, connectionless UDP [RFC768].

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 In 1998 an IETF working group (SIGTRAN) was formed
to design a mechanism for reliably transporting call control
signaling over the Internet. SIGTRAN's goal was to create
an IP complement to the telephone switching's SS7
network.

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Why is SCTP needed given TCP and UDP are widely available?

During SIGTRAN's work, two key problems surfaced in the use of

TCP:

- Head-of-line blocking
- Multi-homing

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 Head-of-line blocking -
A problem where sending independent messages over an order-
preserving TCP connection causes delivery of messages sent later
to be delayed within a receiver's transport layer buffers until an
earlier lost message is retransmitted and arrives. These later
messages often establish independent telephone calls. For call
control signaling, the delay on later messages caused critical call
control timers to expire thus resulting in undesirable call setup
failures.

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 Multi-homing –
where a host with multiple points of attachment to the Internet,
for redundancy purposes, does not want to wait for a routing
convergence (often on the order of minutes) to communicate
critical messages to its peer communication endpoint. For call
control signaling, such delay is unacceptable when an alternate
available path exists. A TCP connection only binds a single
point of attachment at either end point.

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 Considering these problems, SIGTRAN began work on a new
transport protocol to carry its call control signaling over IP
which is not specific to any particular application.
SCTP also includes the following enhancements:

1. improved robustness to failures

2. reduced delay
3. easier coding
4. more secure
5. more flexibility

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Multi streaming - SCTP supports multiple, independent logical
streams of messages within an SCTP association. Each message sent
over an SCTP association is assigned to a particular stream. All data
within a stream is delivered in order with respect to other data in that
stream. Data in different streams have no order constraints. SCTP's
resulting parallel ordered streams provide a specific instance of
'partial ordered' delivery. It is SCTP's multi streaming service that
circumvents the head-of-line blocking problem discussed above.

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 Note:
Multi streaming also is appropriate for applications that wish to
multiplex related, yet independent data streams (e.g., voice, text,
video) over a single end-to-end association rather than open
multiple TCP connections, one for each stream.

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 SCTP's solution for Multi homing - each of the two endpoints
during an SCTP association setup can specify multiple points of
attachment. Having multiple interfaces allows data to be
automatically sent to alternate addresses when failures occur, and
most importantly, without the application even knowing a lower
level failure occurred. Such fault tolerance is unavailable for TCP,
which binds each endpoint to a single interface.
Should either endpoint interface or the link to the interface fail, all
TCP connections bound to that interface would need to timeout and
abort, thus forcing the application(s) to re-establish the
connection(s), and accurately pick up from where the aborted
connection(s) left off.

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 Message Orientation - In TCP, data sent between two
endpoints is a stream of bytes. If needed, an application must
provide message framing. In SCTP, message boundaries are
preserved. If an application sends a 100-byte message, the peer
application will receive all 100 bytes in a single read: no more,
no less. UDP provides a message-oriented service, but without
SCTP's reliability.
Un-Ordered Reliable Service - In TCP, all messages are
reliably delivered to a receiving application in exactly the
order used by the sending application. In addition to ordered
message service (and parallel ordered service discussed
above), SCTP offers the reliable delivery of messages with no
order constraints. UDP provides unordered service, but again
without SCTP's reliability.
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 Extensibility - a TCP packet is limited to 40 bytes for options. In
contrast, SCTP packets can be expanded through the use of Tag-
Length-Value(TLV) fields. Embedded within SCTP's TLV
structures are compatibility handling procedures so that
implementations remain interoperable even when one supports a
more advanced feature set than another.
Heartbeat/Keep-alive - SCTP has a default keep-alive function.
Regular heartbeats validate reachability of peer addresses, and
help maintain a Round Trip Time (RTT) estimate for each
alternate address.

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Message Time-to-Live - SCTP has an option to specify a message's
time-to-live. This feature allows a sending application to specify
how long a message is useful. If this time expires before the
message can be reliably communicated to the receiver, the sending
SCTP entity can stop trying, effectively 'undo'ing or skipping the
message. This type of reliability is called "partial reliability". Partial
reliability is of interest, for example, to the mobile community
(military and commercial) and the online gaming community where
the current "location or environment" status is only good for a short
period, and then outdated by a revised status. In this situation, being
able to discard the stale data will save on bandwidth utilization,
while providing the application with better service in the face of loss
or congestion.
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 Stronger checksum - SCTP uses a 32-bit end-to-end checksum
(CRC32) proven to be mathematically stronger than the 16-bit
ones-complement sum used by TCP and UDP. SCTP's better
checksum provides stronger verification that a message passes
end-to-end without bit errors going undetected.

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 SCTP vs. TCP vs. UDP
Services/Features SCTP TCP UDP
______________________________________________________
Connection-oriented yes yes no
Full duplex yes yes yes
Reliable data transfer yes yes no
Partial-reliable data transfer optional no no
Ordered data delivery yes yes no
Unordered data delivery yes no yes
Flow control yes yes no
Congestion control yes yes no
ECN capable yes yes no
Selective ACKs yes optional no
Preservation of message
boundaries yes no yes
Path MTU discovery yes yes no
Application PDU fragmentation yes yes no
Application PDU bundling yes yes no
Multi-streaming yes no no
Multi-homing yes no no
Protection against SYN flooding
attacks yes no n/a
Allows half-closed connections no yes n/a
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 Main Contents
• Overview of TCP/IP
• Internet Protocol.
• VoIP Network
• VoIP Media Protocols (RTP and RTCP)
• VoIP Signaling Protocols
– H.323 Protocol Stack
– SIP and SDP
– Stream Control Transmission Protocol (SCTP)
– Media Gateway Control Protocol (MGCP)
• Quality of Service
• IP Telephony Market
• Lab Practice

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 Main Contents
• Overview of TCP/IP
• Internet Protocol.
• VoIP Network
• VoIP Media Protocols (RTP and RTCP)
• VoIP Signaling Protocols
– H.323 Protocol Stack
– SIP and SDP
– Stream Control Transmission Protocol (SCTP)
 Media Gateway Control Protocol (MGCP)
• Quality of Service
• IP Telephony Market
• Lab Practice

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 MGCP

MGCP was developed to enable a central device known as

1. Media Gateway Controller
2. Soft-switch
3. Call Agent
to control endpoints or Media Gateways.

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 MGCP

SGCP IPDC

MGCP

MGCP integrates the

Simple Gateway Control Protocol (SGCP)
and
Internet Protocol Device Control (IPDC)
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 MGCP Components

Call Agent or SIP Call Agent or

Media Gateway Media Gateway
H.323
Controller Controller
(MGC) (MGC)

MGCP MGCP

Media Gateway Media Gateway

(MG) (MG)

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 MGC / Call Agent/ Softswitch

MGCP assumes that the bulk of intelligence for call control

operations resides in the call agent
The gateways receive commands from the call agents that
define their operations.
Call Agent or
Media Gateway
MGCP is concerned with several types of Gateways. Controller
(MGC)
• Trunking gateway
• Residential gateway MGCP

• Access Gateway

Media Gateway
(MG)
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 Trunking Gateway

VoIP
VoIP Telco
Telco
Backbone
Backbone

Residential Gateway

VoIP
VoIP Local
Local
Loop
Loop

Access Gateway PBX

VoIP
VoIP

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 MGCP Gateway Components

Calls Connections

Endpoint
Gateway
Endpoint Calls Connections

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 MGCP Gateway Components

Package 1 Trunk

Event a
Calls Connections

Endpoint
Event b

Package 2

Event a

Event b

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 MGCP Gateway components

Endpoint can be physical links, like an E1 trunk.

Endpoints connections can be grouped into calls,
To manage endpoints, there are MGCP packages
A package is the grouping of events and signals that are
supported by a specific type of endpoint.
Example of events are off hook and on-hook events etc.
Example of signals are Dial tone and Ringing etc.

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 MGCP Gateway components

One package might group events and signals related to a

Trunking gateway
Another package might group events and signals related
to a Residential gateway.

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 MGCP Packages
Ten basic Packages defined in MGCP are
Generic Media Package G
DTMF Package D
MF (Media Format Parameter) Package M
Trunk Package T
Line Package L
Handset Package H
RTP Package R
Network Access Server Package N
Announcement Server Package A
Script Package Script
http://www.faqs.org/rfcs/rfc3660.html
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 MGCP Packages

Each package contains a specific series of signals and

events. Some examples are:

Event Symbol Definition

hd Off-hook Transition
hu On-hook Transition
dl Dial Tone
rg Ringing
Many more …

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 Endpoint Types & Supported Packages

MGCP has specific recommendations for which packages should

be implemented on certain endpoint types

Gateway Supported Packages

Trunk Gateway G,D,T,R
Residential Gateway G,D,L,R
…

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 MGCP Control Functions

1. MGCP performs its functions in sequence of commands

& mandatory acknowledgements

2. Command contains
a request verb (action to be performed by the recipient

endpoint)
Additional parameters.
There are nine MGCP commands.
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 MGCP Commands
Verb Code Direction

CreateConnection CRCX MGC to MG

ModifyConnection MDCX MGC to MG

DeleteConnection DLCX MGC to MG

MG to MGC
NotificationRequest RQNT MGC to MG

Notify NTFY MG to MGC

AuditEndpoint AUEP MGC to MG

AuditConnection AUCX MGC to MG

ReStartInProgress RSIP MG to MGC

EndpointConfiguration EPCF MGC to MG

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Endpoint configuration
MGCs issue Endpoint configuration commands to a Gateway,

instructing the gateway about the coding characteristics

Call Agent or
expected by the endpoint. Media Gateway
Controller
Audit Endpoint and Audit Connection (MGC)

MGCs issues the Audit Endpoint and AuditConnection

commands to check on the states of an endpoint and MGCP

any associated connections and calls

Restart in-Progress
Gateways issue the restartin-Progress command
Media Gateway
to notify the MGCs that the endpoints are taken (MG)

out of service or are being placed.

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 MGCP Commands
Create connection
The MGCs issue this command to create
connection that terminates on a specified
Call Agent or
endpoint in the Gateway. Media Gateway
Controller
(MGC)
Notify Connection
The MGCs issue this command to change
MGCP
the parameters associated with a previously
established connection.
Delete Connection Media Gateway
(MG)
MGCs issue this command to delete an
existing connection.
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 MGCP Commands
Notification Request
The MGC issues the Notification Request command to
instruct the gateway to which for specific events on a
specified endpoint.
Call Agent or
Notify Media Gateway
Controller
(MGC)
The gateway responds with the notify command to
inform the MGC when specific event occurs.
MGCP

Media Gateway
(MG)
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 Over Simplified Call Flow
Call Agent
Media Gateway Controller
• When Phone A goes offhook Gateway
A sends a signal to the call agent.
• Gateway A generates dial tone and
collects the dialed digits.
• The digits are forwarded to the call
MGCP MGCP
agent.
• The call agent determines how to route
the call.
• The call agent sends commands to RTP/RTCP
Gateway B.
• Gateway B rings phone B. Gateway A Gateway B
• The call agent sends commands to both
gateways to establish RTP/RTCP
sessions. Phone A Phone B

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 H.323 MGCP
1. A user picks up phone and dials a 1. A user picks up phone and dials a number.
number. 2. The gateway notifies call agent of the phone
2. The gateway determines how to (endpoint) event.
route the call. 3. The Call agent determines capabilities,
3. The two gateways exchange routing information, and issues a command
capabilities information. to the gateways to establish RTP/RTCP
4. The terminating gateway rings the session with other end.
phone.
5. The two gateways establish
RTP/RTCP session with each other.

3 2
Call Agent/
5.RTP/ Media
RTCP 4 Gateway
1
H.323 H.323 Controller
Gateway Gateway
1
RTP/
Gateway A RTCP Gateway B

Phone
Phone Phone
Phone

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 Main Contents
• Overview of TCP/IP
• Internet Protocol.
• VoIP Network
• VoIP Media Protocols (RTP and RTCP)
• VoIP Signaling Protocols
– H.323 Protocol Stack
– SIP and SDP
– Stream Control Transmission Protocol (SCTP)
– Media Gateway Control Protocol (MGCP)
• Quality of Service
• IP Telephony Market
• Lab Practice

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 Main Contents
• Overview of TCP/IP
• Internet Protocol.
• VoIP Network
• VoIP Media Protocols (RTP and RTCP)
• VoIP Signaling Protocols
– H.323 Protocol Stack
– SIP and SDP
– Stream Control Transmission Protocol (SCTP)
– Media Gateway Control Protocol (MGCP)
 Quality of Service
• IP Telephony Market
• Lab Practice

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 Quality of Service

 The Quality of Service consideration is on

“the factors which effects the quality”

 The methods to improve the QoS for voice

traffic in the presence of Data.

 The tools to measure the quality of

service.

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 Delay /Latency (1)

The amount of time it takes for speech to exit the speakers’ mouth
and reach the listener’s ear.

• Propagation Delay

• Handling/Processing Delay

• Serialization Delay

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 Delay /Latency (2)

Propagation Delay is caused by the speed of light in

fiber or copper-based networks.

Light travels at a speed of 186,000 mps

(Miles/Sec) or 299,792,458 meters per second
Electrons travel through copper 125,000 mps

Therefore , a fiber network stretching half way around the

world (13,000 miles) induces a one way delay of about (70
ms) i.e: 13,000/186,000 = (0.07/1000) X 1000 = 70 ms

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 Delay /Latency (3)

Handling Delay caused by the devices that forward the

frame through the network.
( packetization, compression and packet switching)

Serialization delay is the amount of time it takes to actually place

a bit or byte onto an interface. Serialization delay is not covered
in depth because its influence on delay is relatively minimal.

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 Queuing Delay

When packets are held in a queue because of the congestion on an

outbound , the result is queuing delay.

Queuing delay occurs when more packets are sent out than the
interface can handle at a given interval.

Queuing Delay < 10 ms

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 G.114 recommendations defines the end-to-end delay for
good quality voice.

Acceptable delay 150 ms for good quality of voice,

according to the ITU-T recommendations.

But in certain cases that limit exceeds such as in

satellite communications (more than 500 ms).

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 Jitter

Variation of the delay in packet arrival is called jitter.

Data Traffic is very tolerant of Delays and Packet Drops,but

Voice and Video traffic is very intolerant of Delay and Packet drop.

The convergence of multimedia traffic with traditional data traffic

is set to grow.

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How to measure the quality of speech ???

Minimum Opinion Score (MOS)

is a subjective measure of speech quality.

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Other than MOS , there are so many
methods of measuring speech Quality

Like ,
PSQM P.861

PESQ P.862
PAMS
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 MOS is the best way to measure speech
quality ,as no alternative of human ear
to measure quality

than why we use other methods to

measure speech quality????

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It is difficult & time taking process to take
the opinion of
so many Listeners
for the no of voice codes
for different no.of samp/packet.

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Perceptual Speech Quality Measurement ( PSQM P.861 )

Perceptual Evaluation of Speech Quality ( PESQ P.861 )

Perceptual Analysis Measurement System (PAMS)

What is meant by Perceptual Measurement???

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 Actually the precise waveform reproduction is
not necessary to convey all the audible
information contained in the speech.

We have to focus which frequency

components are audible to human ear.

The measurements based on this, are called

Perceptual measurement and coders that use
perceptual model are called Vocoders.

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 In Perceptual model the following parameters
are the matter of consideration

Sound Masking - the high volume mask

of the low volume voice

Critical Band - the range of audible sound

spectrum is narrower at low frequencies and
wider at high frequencies.

Sound Pressure level and

Many more……
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 PSQM monitors the impairments caused by
compression and decompression but
it can’t monitor the impairments caused by
packet loss and jitter.

PESQ takes into account coding distortions, errors,

packet loss, delay and variable delay,
and filtering in analogue network components.

PAMS can measure the activity of speech.

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 What is ECHO?

Hearing your own voice in the receiver after a delay of

16 to 20 ms during conversation

is called Echo.

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 Echo cancellers are used to remove echo

In a packet based network

UserA UserB

Echo cancellation

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 Packet Loss degrades the quality of Voice.

Causes of Packet loss

Network delay much more than permissible limit.

VOIP router buffer defines the permissible limit

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 How to overcome packet loss?

VOIP router waits for a lost packet for a period of

time.

If router can’t receive the lost packet it will replay the

last received packet.

That strategy is not suitable for more than one packet

loss.

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 Digital to Analog conversions degrades the quality .

In G.729 (8 kbps), just two conversions from D/A

cause the MOS to decrease rapidly.

This result in degradation of voice quality.

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 Queuing Delay

It is just like a Queue at the highway system.

The vehicle comes first will leave first (FIFO).

The other vehicles in the queue have to wait for the

vehicles to pass which comes earlier.

At every Toll Plaza the vehicle faces a delay depending

on the length of the Queue.

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 Similarly, in packet networks, every packet offers a
delay depending on the length of the queuing packets.

The length of the Queue depends on the size of the jitter

buffer.

If the length of the Queue at any node is greater than the

size of the jitter buffer ,it will result in packet loss.

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 As today’s data networks , offer different type of services so
it is important to set different priorities to different packets.

Different Queues for packets having different priorities

These are;
WFQ-Weighted Fair Queuing.
CQ-Custom Queuing.
PQ-Priority Queuing.
CB_WFQ

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 WFQ-Weighted Fair Queuing

Weighted fair Queuing allocates equal bandwidth for all

applications .

The low volume traffic has given high priority.

WFQ has implemented at interfaces 2.048 Mbps or less.

WFQ is similar to TDM, as it equally divides BW among

different applications.

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 CQ-Custom Queuing

The percentage of available bandwidth is allocated to a

particular protocol.

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 PQ- Priority Queuing

In PQ , four different priorities of flow are available.

High , Normal, Medium and Low

Inbound traffic is assigned to one of the four

different queues.

Traffic in higher priority queue is serviced until the

queue is empty;then next priority queue is transmitted.

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 By PQ, the max BW is assigned to an application as it
needs but it starves other applications to do so.

PQ should be carefully assigned as it forces low

priority application to stop.

Remember highest priority must be given to low

volume traffic.

In case of assigning high priority to large volume

traffic, the low volume traffic may stop working.

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 CB – WFQ (class based WFQ)

High priority packets are serviced while the

packets with the low priority can be dropped.

Such type of priority is assigned at interfaces

less than 768 kbps.

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 QoS has three main criteria;

Best Effort

Differentiated Service - can deal with differing levels

of QoS on a packet by packet basis

Integrated Service - requested level of service by an

application to a network. Data is only sent after
confirmation that the service level is guaranteed to be
available.

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 Policing and Shaping
Overview

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 Traffic Policing
CAR-Committed Access Rate.
Traffic Shaping
GTS-Generic Traffic Shaping.
FRTS-Frame Relay Traffic Shaping.

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 Both Traffic Shaping and Traffic policing use the
Token bucket mechanism.
Token Bucket mechanism is different as compared
to the Leaky Bucket mechanism.

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 Leaky Bucket

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 Token Bucket

• Parameters
– r – average rate, i.e., rate at which tokens fill the
bucket
– b – bucket depth
– R – maximum link capacity or peak rate (optional
parameter)
• A bit is transmitted only when there is an available token
r bps

b bits

<= R bps

regulator
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 Committed Access Rate (CAR)

CAR examines traffic receives on an interface.

It can be configured to transmit ,drop or set

precedence.

CAR utilizes a token bucket measurement. Traffic

arriving at the bucket when sufficient tokens are
available is said to conform and the corresponding
number of tokens are removed from the bucket.
If sufficient tokens are not available, then the
traffic is said to exceed.

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 GTS and FRTS

(Generic traffic shaping and frame Relay traffic shaping)

GTS support WFQ also.

FRTS support PQ or CQ.
FRTS is implemented at DLCI (Data Link Connection
Identifiers) level and
GTS is implemented at interface and sub-interface
level

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Traffic policy apply at the IP level and Traffic Shaping
apply at the Layer-2 technologies like frame relay ,X.25
and ATM.
The advantages of Traffic Shaping is that it prevents packet
Loss.

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 Main Contents
• Overview of TCP/IP
• Internet Protocol.
• VoIP Network
• VoIP Media Protocols (RTP and RTCP)
• VoIP Signaling Protocols
– H.323 Protocol Stack
– SIP and SDP
– Stream Control Transmission Protocol (SCTP)
– Media Gateway Control Protocol (MGCP)
• Quality of Service
• IP Telephony Market
• Lab Practice

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 Main Contents
• Overview of TCP/IP
• Internet Protocol.
• VoIP Network
• VoIP Media Protocols (RTP and RTCP)
• VoIP Signaling Protocols
– H.323 Protocol Stack
– SIP and SDP
– Stream Control Transmission Protocol (SCTP)
– Media Gateway Control Protocol (MGCP)
• Quality of Service
 IP Telephony Market
• Lab Practice

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 Agenda

• Internet around the world

• IP Telephony:
– What are the issues?
– Implications for high, medium and low-priced markets
– Country case studies
• World Telecom Policy Forum 2001: IP Telephony
– Geneva, March 7-9, 2001
– Opinions and outputs
– Information resources

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 Number of Internet host computers,
in millions, and annual growth in %

104.0
135%

72.4
107%
100%
93% 95%
81% 79%
70% 43.2 67%

29.7 46%
21.8 36% 44%
14.4
2.7 5.8
0.1 0.2 0.4 0.7 1.3

Jan- Jan- Jan- Jan- Jan- Jan- Jan- Jan- Jan- Jan- Jan- Jan- Jan-
89 90 91 92 93 94 95 96 97 98 99 00 01
Source: ITU, adapted from Internet Software Consortium <www.isc.org>.

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 Top Internet markets
Top 10 countries by number of Top 10 countries by Internet user
Internet users (millions), Jan. 2001 penetration, Jan. 2001

USA 95 Iceland 60%

Japan 39 Norway 49%
China 23 Sweden 46%
Germany 20 Canada 41%
Korea (Rep.) 16 Finland 40%
UK 15 Denmark 37%
Canada 13 Korea (Rep.) 35%
Australia 7 Australia 35%
France 7 USA 35%
Italy 6 Singapore 30%

Source: ITU, Internet Software Consortium <www.isc.org>, RIPE <www.ripe.net>.

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 Inter-regional Internet backbone

357 Mbit/s

USA & 56’241 Mbit/s

Canada
19’716 Mbit/s

2’638
Asia-
Mbit/s 468 Europe
Pacific 171
Mbit/s
Mbit/s
Latin Arab
Source: TeleGeography Inc., America & States,
Global Backbone Database. Caribbean Africa
Sept. 2000.
127 Mbit/s
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 IP Telephony: What is it?

• Internet Protocol (IP) Telephony is a generic term describing voice

or fax carried over IP-based networks, such as the Internet.
• IP Telephony is important:
– In the short-term, because it cuts the cost of calls, especially if
routed over the public Internet
– In the longer-term, because telecom carriers are migrating their
separate voice and data networks to converged IP-based
networks
• Examples of IP Telephony Service Providers include Net2Phone,
Dialpad.com, iBasis etc.
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 IP Telephony: Four main stages of evolution

1. PC-to-PC (since 1994)

– Connects multimedia PC users, simultaneously online
– Cheap, good for chat, but inconvenient and low quality
2. PC-to-Phone (since 1996)
– PC users make domestic and int’l calls via gateway
– Increasingly services are“free” (e.g., Dialpad.com)
3. Phone-to-Phone (since 1997)
– Accounting rate bypass
– Low-cost market entry (e.g., using calling cards)
4. Voice/Web integration (since 1998)
– Calls to website/call centres and freephone numbers
– Enhanced voice services (e.g., integrated messaging)
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 IP Telephony traffic, in million minutes

7'000 5.5%
6'000
As percentage of int'l
5'000 outgoing traffic
3.2%
4'000
3'000
1.6%
2'000
1'000 0.2%
0.0%
0
1997 1998 1999 2000 2001
Source: ITU Internet Reports, adapted from TeleGeography Inc.
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 Percentage of IP Telephony Revenues by Region
(1998-2003)

  1998 1999 2000 2001 2002 2003

(%) (%) (%) (%) (%) (%)

North 45 43.5 42.6 40.2 38.2 37.7

America
Europe 38 37.2 36.6 36.3 35.4 34.7

Asia Pacific 9 11.2 12.1 14.5 15.2 15.9

Asia 2 2 2.3 1.7 2.8 3

MEA 2 1.9 1.8 1.4 1.6 1.6

South 3 3.1 3.4 4.5 5.3 5.5

America
Rest of World 1 1.1 1.2 1.4 1.5 1.6
 
Total 100 100 100 100 100 100

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 Pricing IP for voice services

• In competitive, low-price markets

– Main market opportunity for IP Telephony is for value-added
services.
• In markets in transition to competition
– IP Telephony offers a route towards early introduction of
competition and creates downward pressure on prices
• In high-price, monopoly markets
– Where permitted, IP Telephony creates opportunities for low-
cost calls
– Even if not permitted, IP Telephony is widely used to reduce
costs of international call termination

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 Country case studies: IP Telephony legal status

Source: Summary of ITU country case studies, available at: www.itu.int/wtpf/casestudies.

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 Country case studies: Potential price savings using
IP Telephony

Note: “IDD” refers to published prices from the incumbent operator for international direct dialling. “IP out” refers
to using the Net2Phone IP Telephony service within the country. “IP in” refers to using Net2Phone in the US to
call to the country.
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 Country case studies: The stakes in
international traffic flows

Note: “Outgoing”, “Incoming” and “Total” show international traffic in millions of minutes p.a. “Per inhabitant” and
“per main line” shows traffic in minutes p.a., “As % of revenue” shows revenue from international traffic as a % of
total telecom revenue for the country.
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 World Telecommunication Policy Forum:
What is it?

“The purpose of the forum is to provide a venue for creating a shared

vision among policy-makers worldwide on the issues arising from
the emergence of new telecommunication services and
technologies, and to consider any other policy issue in
telecommunications which would benefit from a global exchange
of views”
• “Rough consensus” in the form of opinions which are non-
prescriptive and non-binding
• Some 750 senior policy-makers, regulators and industry experts
were present, from more than 120 countries and 100 private sector
members

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 IP Telephony: What are the issues?

• Technical:
– How to define IP Telephony?
– Is quality of service comparable? Will it improve?
– How to handle numbering issues?
• Economic:
– What price and cost savings can be expected?
– How quickly will carriers migrate their networks?
– Isn’t it just a form of bypass of telecom monopolies?
• Regulatory:
– Is it voice or is it data?
– License it? Prohibit it? Restrict it? Liberalise it?
– Should IP Telephony contribute to Universal Service?

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 Opinion A: General implications of IP Telephony
• Implications for Member States
– the deployment of IP-based networks benefits users, industries, and the economy
at large, because it fosters technical and market innovation, diversity and
economic growth
– IP … could be viewed as a significant opportunity for all countries to respond to
the convergence of information and communication technologies and evolve
their networks;
• Implications for Operators
– continuing development of the Internet and IP-based networks is a significant
medium for communications and commerce;
– mobile wireless systems are expected to migrate towards an IP-based
architecture in order to deliver integrated voice, data and multimedia services, as
well as access to the Internet;
• ITU … is of the view that:
– IP Telephony applications are best supplied in a market in which consumers
have choices among multiple, alternative sources because only then will citizens,
businesses and the overall economy reap the benefits of innovation and cost
effectiveness

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 Opinion A: Invites …

• Member States and Sector Members to consider the possibility of

the introduction and deployment of IP technologies and
IP applications, including the exchange of information;
• All Member States to review their current regulatory frameworks
with a view to:
– encouraging investment, spurring innovation and advancing
development;
– achieving public policy goals in the context of a converged
communication services environment;
– considering the possibility of opening their communication
services market with respect to IP Telephony by adopting a
competition-oriented approach in order to achieve clearly
defined public policy goals, taking into account, among
other things, the concept of technology neutrality for fully-
substitutable services
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 Opinion B: Actions to assist Member States and
Sector Members
• Encourages Member States to share experiences:
– on approaches towards making any sector specific regulation technology-
neutral;
– the application of domestic competition laws as part of a pro-competitive
policy;
– establishing sustainable bases to generate financing for universal
access/service.
• Invites the ITU to:
– Carry out and update IP Telephony case studies;
– Carry out cost studies and assist Members in performing cost-benefit
analyses;
– Help Member States attract investment
• Invites ITU to carry out regional workshops on:
– IP-based technologies and network evolution
– Cost structures, pricing mechanisms, interconnection, numbering,
attracting investment, market considerations etc.
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 Opinion C: Human Resources Development

• Recognising that:
– the transition to an IP environment places new demands on the
management and operation of ITU members;
– the speed with which countries can extend the benefits of
telecommunication technologies will depend on their ability to deploy
skilled personnel;
• Invites the ITU:
– to facilitate the sharing of knowledge and views, by Member States and
Sector Members, of global trends in all new technologies, in particular IP
applications;
– to encourage Member States and Sector Members to create integrated
human resources transition plans towards IP;
– to assist Member States and Sector Members in evaluating and identifying
new and changing human resources requirements

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 Opinion D: Essential Studies to facilitate
introduction of IP Telephony
• Invites the three ITU sectors to initiate studies to facilitate the
introduction of IP Telephony on:
– compatibility and inter-operability of radio access between IP networks
and PSTNs,
– working definitions of IP telephony and Internet telephony
– whether, and to what extent, to require compatibility with the existing
international telephone service, including developing appropriate
performance metrics and QoS
– Whether, and to what extent, IP Telephony can be part of national PSTNs
and whether traffic identification and measurement need to be considered
– identifying the cost elements of international IP connectivity with respect
to the introduction of IP Telephony
• Invites the ITU-D to establish a group of experts to:
– Prepare a checklist of Factors to be considered;
– Advise on the policy impacts in developing countries;
– Prepare a report to the next World Telecom Development Conf.
Institute of Communication Technologies , Islamabad. 253
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www.ictech.edu.pk
 VONAGE-Largest IP Telephony Operator (USA)

Vonage offer services of IP telephony all over the world from rates
(as low as 6 cents/min)

US 14.99$/month to call for 500 mins in USA and canada

US 24.99$/month to call anywhere in USA and canada
unlimited.

In Pakistan the rates are;

Karachi 25 cents/min
Lahore 26 cents/min
Islamabad 27 cents/min
Institute of Communication Technologies , Islamabad. 254
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www.ictech.edu.pk
 Net-phone software for your laptop and home pc to charge
4.99 dollar per month

Subscriber Facility:

CLI with name,voice mail for there days

call waiting ,call forwarding,in network calls---
within the network no charges etc

Institute of Communication Technologies , Islamabad. 255

June 8, 2020
www.ictech.edu.pk
 Most Recently AOL launches the same services with
13.99$/month.

Institute of Communication Technologies , Islamabad. 256

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www.ictech.edu.pk
 IP TELEPHONY PERMITTED
ANGOLA GREECE NETHERLANDS
ANTIGUA AND BARBUDA GUATEMALA NEWZEALAND
ARGENTINA GUYANA PERU
AUSTRALIA HONG KONG SAR PHILIPPINES
AUSTRIA HUNGARY POLAND
BELGIUM ICELAND PORTUGAL
BHUTAN INDIA SINGAPORE
CANADA IRELAND SLOVAK REPUBLIC
CHINA ITALY SPAIN
CONGO JAPAN SRI LANKA
COSTA RICA KENYA ST. LUCIA
CYPRUS KOREA (REP) ST. VINCENT
CZECH REPUBLIC KYRGYZSTAN SWEDEN
DENMARK LUXEMBOURG SWITZERLAND
DOMINICAN REPUBLIC MADAGASCAR TONGA
ESTONIA MALAYSIA UGANDA
ETHIOPIA MALTA UNITED KINGDOM
FINLAND MEXICO UNITED STATES
FRANCE MOLDOVA VIET NAM
GAMBIA MONGOLIA
GERMANY NEPAL

Institute of Communication Technologies , Islamabad. 257

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 Main Contents
• Overview of TCP/IP
• Internet Protocol.
• VoIP Network
• VoIP Media Protocols (RTP and RTCP)
• VoIP Signaling Protocols
– H.323 Protocol Stack
– SIP and SDP
– Stream Control Transmission Protocol (SCTP)
– Media Gateway Control Protocol (MGCP)
• Quality of Service
• IP Telephony Market
• Lab Practice

Institute of Communication Technologies , Islamabad. 258

June 8, 2020
www.ictech.edu.pk
 Main Contents
• Overview of TCP/IP
• Internet Protocol.
• VoIP Network
• VoIP Media Protocols (RTP and RTCP)
• VoIP Signaling Protocols
– H.323 Protocol Stack
– SIP and SDP
– Stream Control Transmission Protocol (SCTP)
– Media Gateway Control Protocol (MGCP)
• Quality of Service
• IP Telephony Market
 Lab Practice

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 Lab Practice No 1.
3221
2221
WHITE WHITE

PAKISTAN
1/0/0 ENGLAND 1/0/0

ROUTER
ROUTER 3222

BLACK
2222
BLACK 1/0/1
1/0/1
1. Dial Tone GB (ENGLAND) 1. Dial Tone PAKISTAN
2. Signaling Loop Start 2. Signaling Loop Start
3. Note “YOUR NAME” 3. Note “YOUR NAME”
4. Duration before first digit 30 seconds 4. Duration before first digit 20 seconds
5. If caller does not attend phone for more 5. If caller does not attend phone for more
than 10 seconds then call must be than 10 seconds then call must be
disconnected. disconnected.
6. Timing gap betweeen subsequent digits 6. Timing gap betweeen subsequent digits
should be 5 seconds. should be 5 seconds.
7. Increase hookflash sensitivity. 7. Increase hookflash sensitivity.
8. Disable echo control mechanism. 8. Disable echo control mechanism.
9. Line Impedance 600 ohms. 9. Line Impedance 600 ohms.
10. Pots Dial peer. 10. Pots Dial peer.

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 Lab Practice No 2.
3221

FX0
ROUTER
FXS
FX0 3222

Outbound
1/1/1

Inbound
1/0/1

RJ-11

PSTN

PBX
RECEPTION
Institute of Communication Technologies , Islamabad. 261
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 Lab Practice No 3.

LAHORE ISLAMABAD
SIALKOT 42 51
52
` ` 000
000 000
3.1.1.3 3.1.1.2 4.1.1.3 4.1.1.2

1/0/0

ROUTER SWITCH SWITCH

1.1.1.1

1/0/1 LONDON
INTERNET 41
3.1.1.1 4.1.1.1
6.1.1.1 6.1.1.2
1.1.1.2
2.1.1.2 2.1.1.1
000
ROUTER ROUTER
7.1.1.1 8.1.1.2

001
7.1.1.2 8.1.1.1

172.16.100.100 ROUTER

PSTN
CALL MANAGER 11
SWITCH
172.16.100.1

PSTN

Institute of Communication Technologies , Islamabad. 262

June 8, 2020
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