PREFACE

Presently various telecom services are provided through separate

networks. Technological advancements in telecommunications are forcing a

trend towards unification of networks & services setting up the stage for the

emergence of Next Generation Networks (NGN). In the next generation

networks, multiple access networks can connect customers to a core network,

which is predominantly based on IP technology. NGN promises to provide

number of significant benefits and opportunities both for the service providers

and the end-users by providing new innovative services and applications

through a common platform.

With the efficient and cheaper IP technology forcing telecommunications

networks to migrate to Next Generation Networks, triple play (voice, data and

video) would become a basic service. Traffic of different services of data,

television and subsequently voice would be simply enclosed in Internet protocol

packets, transmitted over these networks. These networks can later support

any number of additional value-added services and transmit them also as IP

packets.

As an early application and driver of NGN, VOIP is proliferating fast and

is expected to result in significant penetration in the matured telecom markets.

In India, till some time back IP telephony was permitted only in a restrictive

manner i.e. PC-to-PC, IP device-to-IP device and PC-to-Phone (abroad). Now

with recent guidelines, Govt. has permitted UASPs (telecom access providers) to

provide phone-to-phone Internet Telephony viz. unrestricted VOIP and therefore

this is likely to proliferate in India also. VOIP is likely to have a big impact on

the traditional circuit switched telephony, initially on fixed lines followed by

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mobile, driving consumer prices and margins down, forcing far-reaching

changes in industry.

2
What is the NGN?

The Next Generation Network (NGN) is a popular phrase used to describe the

network that will replace the current PSTN network around the world today

used to carry voice, fax, modem signals, etc.

The ITU defined the term NGN in Recommendation Y.2001 as follows:

A packet-based network able to provide telecommunication services and able to

make use of multiple broadband, QoS-enabled transport technologies and in

which service-related functions are independent from underlying transport-

related technologies. It offers unrestricted access by users to different service

providers. It supports generalized mobility which will allow consistent and

ubiquitous provision of services to users.

ITU-T draft Y.2001 goes on to characterize the NGN by the following

fundamental aspects:

Packet-based transfer.

Separation of control functions among bearer capabilities, call/session,

and application/service.

Decoupling of service provision from network, and provision of open

interfaces.

Support for a wide range of services, applications, and mechanisms

based on service building blocks (including real time/streaming/non-real

time services and multi-media).

Broadband capabilities with end-to-end QoS (Quality of Service).

Interworking with legacy networks via open interfaces.

Generalized mobility.

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 Access to different service providers, independent of any access or

transport technologies. Unrestricted access by users to different service

providers.

A variety of identification schemes.

Unified service characteristics for the same service as perceived by the

user.

Converged services between Fixed/Mobile.

Independence of service-related functions from underlying transport

technologies.

Support of multiple last mile technologies.

Compliant with all Regulatory requirements; for example, concerning

emergency communications, security, privacy, and so forth.

From a practical perspective, NGN involves three main architectural changes

that need to be looked at separately:

In the core network, NGN implies a consolidation of several (dedicated or

overlay) transport networks each historically built for a different service

into one core transport network (often based on IP and Ethernet). It

implies amongst others the migration of voice from a switched

architecture (PSTN) to VoIP, and also migration of legacy services such as

X.25, Frame Relay (either commercial migration of the customer to a new

service like IP VPN, or technical emigration by emulation of the "legacy

service" on the NGN).

In the wired access network, NGN implies the migration from the "dual"

legacy voice next to xDSL setup in the local exchanges to a converged

setup in which the DSLAMs integrate voice ports or VoIP, allowing to

remove the voice switching infrastructure from the exchange.

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 In cable access network, NGN convergence implies migration of constant

bit rate voice to CableLabs PacketCable standards that provide VoIP and

SIP services. Both services ride over DOCSIS as the cable data layer

standard.

Next Generation Network (NGN) is a broad term to describe some key

architectural evolutions in telecommunication core and access networks that

will be deployed over the next 5-10 years. The general idea behind NGN is that

one network transports all information and services (voice, data, and all sorts

of media such as video) by encapsulating these into packets, like it is on the

Internet. NGNs are commonly built around the Internet Protocol, and therefore

the term "all-IP" is also sometimes used to describe the transformation towards

NGN.

One of the most important aspects of NGN is the deliberate separation of

the access provider from the "service" provider. For those that do not

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understand what this means, it means that the access provider (the service

provider that provides you, the customer, with access to the NGN) may be

different than the service provider that provides you with various services,

such as voice and video communication, e-mail, stock quotes, or other services.

In a NGN there is a more defined separation between the transport

(connectivity) portion of the network and the services that run on top of that

transport. This means that whenever a provider wants to enable a new service,

they can do so by defining it directly at the service layer without considering

the transport layer - i.e. services are independent of transport details.

Increasingly applications, including voice, will tend to be independent of the

access network (de-layering of network and applications) and will reside more

on end-user devices (phone, PC, Set-top box).

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 Next Generation Networks are based on Internet technologies including

Internet Protocol (IP) and Multi-protocol Label Switching (MPLS). At the

application level, Session Initiation Protocol (SIP) seems to be taking over from

ITU-T H.323.

Initially H.323 was the most popular protocol, though its popularity

decreased in the "local loop" due to its original poor traversal of NAT and

firewalls. For this reason as domestic VoIP services have been developed, SIP

has been far more widely adopted. However in voice networks where everything

is under the control of the network operator or telco, many of the largest

carriers use H.323 as the protocol of choice in their core backbones. So really

SIP is a useful tool for the "local loop" and H.323 is like the "fiber backbone".

With the most recent changes introduced for H.323, it is now possible for

H.323 devices to easily and consistently traverse NAT and firewall devices,

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opening up the possibility that H.323 may again be looked upon more favorably

in cases where such devices encumbered its use previously. Nonetheless, most

of the telcos are extensively researching and supporting IMS, which gives SIP a

major chance of being the most widely adopted protocol.

For voice applications one of the most important devices in NGN is a Soft

switch - a programmable device that controls Voice over IP (VoIP) calls. It

enables correct integration of different protocols within NGN. The most

important function of the Soft switch is creating the interface to the existing

telephone network, PSTN, through Signalling Gateways (SG) and Media

Gateways (MG). However, the Soft switch as a term may be defined differently

by the different equipment manufacturers and have somewhat different

functions.

One may quite often find the term Gatekeeper in NGN literature. This

was originally a VoIP device, which converted (using gateways) voice and data

from their analog or digital switched-circuit form (PSTN, SS7) to the packet-

based one (IP). It controlled one or more gateways. As soon as this kind of

device started using the Media Gateway Control Protocol (and similars), the

name was changed to Media Gateway Controller (MGC).

A Call Agent is a general name for devices/systems controlling calls.

The IP Multimedia Subsystem (IMS) is a standardised NGN architecture

for an Internet media-services capability defined by the European

Telecommunications Standards Institute (ETSI) and the 3rd Generation

Partnership Project (3GPP).

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NGN ARCHITECTURE

1. INTRODUCTION
At present separate networks exist for voice, data, mobile, Internet etc.

Over the years, network operators have been looking for a service independent

network architecture which can facilitate rapid and economical introduction of

new services. The explosion of the Internet and popularity of Internet

multimedia services emphasized the need to shift towards packet based core

networks from the present circuit-switched networks.

NGN is envisaged to facilitate the convergence of voice, data and video

networks into a single unified packet-based multi-service network capable of

providing futuristic services. Converging voice, data and video services onto a

common network infrastructure with a universal addressing scheme and

mobility will provide unprecedented cost savings and performance advantages.

An aim of NGN is to support PSTN/ISDN replacement. Therefore, the

NGN provides support for PSTN/ISDN emulation as well as PSTN/ISDN

simulation.

2. NGN ARCHITECTURE
The NGN architecture incorporates:

Support for multiple access technologies: The NGN architecture offers

the configuration flexibility needed to support multiple access technologies.

Distributed control: This will enable adaptation to the distributed

processing nature of packet-based networks and support location transparency

for distributed computing.

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 Open control: The NGN control interface is open to support service

creation, service updating, and incorporation of service logic provision by third

parties.

Independent service provisioning: The service provisioning process is

separated from transport network operation by using the above-mentioned

distributed, open control mechanism. This is intended to promote a competitive

environment for NGN development in order to speed up the provision of

diversified NGN services.

Support for services in a converged network: This is needed to generate

flexible, easy-to-use multimedia services, by tapping the technical potential of

the converged, fixed-mobile functional architecture of the NGN.

Enhanced security and protection: This is the basic principle of an open

architecture. It is imperative to protect the network infrastructure by providing

mechanisms for security and survivability in the relevant layers.

Functional entity characteristics: Functional entities incorporate the

following principles:

Functional entities may not be distributed over multiple physical units but

may have multiple instances.

Functional entities have no direct relationship with the layered architecture.

However, similar entities may be located in different logical layers.

One of the main characteristics of NGN is the decoupling of services and

networks. Decoupling can provide the advantage of services to be offered

separately and to evolve independently. A functional NGN architecture is shown

in Figure 1. It is horizontally layered network architecture instead of the

present vertically separated networks for each service. It uses Internet Protocol

(IP) based transport for all services including voice.

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Figure 1 NGN Architecture Using Softswitch

AGW: Access Gateway, AN: Access Network, IAD: Integrated Access

Gateway, LMG: Line Media Gateway, MGC: Media Gateway Controller, MS:

Media Server, SG: Signalling gateway, OSS: Operations Support Systems, SCP:

Service Control Point, TMG: Trunk Media Gateway,

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Access Layer
Access layer of the NGN architecture has following functions:

Connecting subscribers (Legacy/IP based), AN & PABX and trunks from

PSTN, ISDN, and PLMN etc.

Converting the format of information (circuit to packet or packet to

circuit) before transmitting it.

The components on the access layer are:

IP terminals: The IP terminals refer to IP phones, IP PBX and software

phones. They are typically intelligent terminals based on either H.323 or

SIP protocol. IP terminals do not require to convert media as the voice

already digitalized with IP terminals

Integrated access device (IAD): It is a device used to access subscribers

(Analog, ADSL, IP) in the NGN. It accesses data of the subscriber

terminals, voice services and video services to the packet network.

Access Gateway (AGW): It acts as the line side interface to the core IP

network and connects subscribers with analog subscriber access,

integrated services digital network (ISDN) subscriber access, V5

subscriber access, PABX and x digital subscriber line (xDSL) access.

Access Network (AN): The access network provides connectivity between

the customer premises equipment and the access gateways in the service

providers network.

The access network includes access-technology dependent functions, e.g., for

W-CDMA technology and xDSL access. Depending on the technology used for

accessing NGN services, the access network includes functions related to:

1. Cable access

2. xDSL access
 3. Wireless access (e.g. IEEE 802.11 and 802.16 technologies, and

3G RAN access)

4. Optical access

SIP phone: It is a multimedia device working in the Session Initiation

Protocol (SIP).

H.323 phone: It is a multimedia device working in the H.323 protocol.

Signalling gateway (SG): The SG provides the signalling interface between

the IP network and the PSTN signalling network. It terminates SS7 links

and provides Message Transport Part (MTP) Level 1 and Level 2

functionality. Each SG communicates with its associated circuit switch

(CS) to support the end-to-end signalling for calls.

Trunk Media gateway (TMG): It resides between the circuit switched (CS)

network and the IP network. It converts format between pulse code

modulation (PCM) signal flow and IP media flow. It supports functions

such as packetisation, echocontrol etc It can have integrated signalling

gateway functionality also. The MGW can connect with devices, such as

the PSTN exchange, private branch exchange (PBX), access network

devices and base station controller (BSC).

Session Border Controller (SBC): Session Border Controller is locatedat

the administrative boundary of an IP network for enforcing policy on

multimedia sessions. Session policy may be defined to manage security,

service level agreements, network device resources, network bandwidth,

inter-working and protocol interoperability between networks.

Session Border Controller (SBC) provides functions as below:

Inter-working

Security
 Management of service level agreements

Overload control

Network Address Translation and Firewall Traversal

Lawful Interception

Quality of Service (QoS) management

Protocol Translation

Call accounting

Session Border Controller functions can be logically split into two types:

signalling-related functions and media-related functions, according to whether

these functions are co-located or not.

Transport Layer

The transport functions provide the connectivity for all components and

physically separated functions within the NGN. These functions provide

support for the transfer of media information, as well as the transfer of control

and management information.

The transport layer is composed of devices, such as routers and layer 3

switches that are located in the backbone network and in the MAN. The

primary function of the IP core network is to provide routing and transport of

IP packets. It adopts the packet switching technology and provides subscribers

with a common, integrated platform of data transport, which ensures:

High reliability

QoS assurance

High capacity

Control Layer

The network control layer adopts the software switching or softswitching

technology to achieve:
 Primary real-time call control

Connection control

The softswitch, also known as Media Gateway Controllers (MGC), Call

Servers (CS) and Call Agents is the core device in the NGN. The Softswitch is

located in the service providers network and handles call control and signalling

functions, typically maintaining call state for every call in the network. A

Softswitch interacts with Application Servers to provide services that are not

directly hosted on Softswitch. Important functions of Softswitch are:

Call control

Media gateway access control

Signalling Gateway Control

Border Gateway control

Resource allocation

Protocol processing

Routing

Authentication

Charging

Softswitches also act as Signalling Switching Point (SSP) to provide

access to IN services to SIP users.

The softswitch also provides services such as:

Basic voice services

Multimedia services

Service Layer

The service layer provides value-added services and operation support

functions. The components of the service layer are:

 OSS (Operation Support System): It includes an integrated charging system,

and network operation & management system, which conducts centralised

management on the NGN components.

Application server: It produces and manages logics of value-added services

and intelligent network (IN) services, providing a platform for a third party to

develop services through open APIs. The application server is the result of

separating service from call control. It helps to develop supplementary services.

Media Server (MS): It processes media streams in the basic and enhanced

services. It provides functions of service tone playing, conference service,

interactive voice response (IVR), recorded announcement and advanced tone

service.

Service control point (SCP): It is the core component in the traditional IN,

which is used to store subscriber data and service logics. The SCP starts a

service logic based on the call events reported from the service switching point

(SSP). It then, queries the service database and the subscriber database using

the started service logic and sends proper call control instructions to the SSP

on the next action. This helps to realize various intelligent calls, which is the

main function of SCP.

Video server: It schedules and manages video conferences, and provides

video conferences to NGN users.

 3. NGN PROTOCOLS
The figure 2 shows the control and media streams in NGN environment.

The media streams consist of audio, video or data, or a combination of any of

them. Media stream conveys user or application data (i.e., a payload) but not

control data. It is transported through RTP/RTCP. Control signalling messages

are transported by control streams using signalling protocols like SIGTRAN,

H.248, H.323, SIP etc.

Figure 2 Protocols Used in NGN

 IAD: Integrated Access Gateway, IN: Intelligent Network TMG: Trunk

Media Gateway, SG: Signalling gateway, OSS: Operations Support Systems

H.323
H.323 is an ITU Recommendation that defines "packet-based multimedia

communications systems." It defines a distributed architecture for creating

multimedia applications, including VoIP. The H.323 protocol is best known as

the original call signalling protocol that made real time voice and video over IP

possible. Being the first solution to work, H.323 is the most widely deployed

protocol in the market and through its veteran status and wide acceptance

provides telecommunication equipment with the benefits of a highly mature

and completely interoperable signalling solution.

SIP
Designed by the IETF, the Session Initiation Protocol (SIP) is an

application-layer control protocol for multimedia communication over IP

network. It is used for creating, modifying and terminating two party sessions,

multiparty sessions and multicast sessions (one sender and many receivers).

These sessions include audio, video and data for multimedia conferences,

instant messaging, Internet telephone calls, distance learning, telemedicine,

multiparty real time games etc.

SIP defines telephone numbers as URLs (Uniform Resource Locators), so

that web pages can contain them. This allows a click on a link to initiate a

telephone call. These addresses take the form of user@host, similar to e-mail

addresses. The user part, which is left of the @ sign, may be user name or a

telephone number and host part, which is right of the @ sign, is a domain

name or IP address. SIP addresses may be obtained out-of-band, learned via

media gateways, recorded during earlier conversations, or guessed (since

theyre often similar to E-mail addresses. SIP may be used in conjunction with

other call setup & signalling protocols and has a verity of other features like

caller reachability, call screening, encryption and authentication etc.

MGCP
Media Gateway Control Protocol (MGCP) is a control protocol that uses

text or binary format messages to setup, manage, and terminate multimedia

communication sessions in a centralised communications system. This differs

from other multimedia control protocol systems (such as H.323 or SIP) that

allow the end points in the network to control the communication session.

MGCP is specified in RFC 2705. MGCP is, in essence, a master/slave protocol,

where the MGs are expected to execute commands sent by the MGCs.

H.248
H.248 is an ITU Recommendation that defines Media Gateway Control

Protocol. It is the result of a joint collaboration between the ITU and the IETF.

It is also referred to as IETF RFC 2885 (MEGACO), which defines a centralised

architecture for creating multimedia applications, including VoIP. In many

ways, H.248 builds on and extends MGCP. It is used as a media gateway

control protocol between a Media Gateway Controller (MGC) and a Media

Gateway (MG). The ITU-T, the IETF, the International Softswitch Consortium

(ISC), and other standardization organizations are optimizing the H.248

protocol currently. Telecommunication equipment vendors are investing much

in the development and application of the H.248 protocol. Compared to the

MGCP protocol, the H.248 protocol is more flexible and can support more types

of access technologies and mobility of terminations.

SIGTRAN
 SIGTRAN (Signalling Transport) is a protocol stack defined by the

SIGTRAN workgroup of the Internet Engineering Task Force (IETF) for transport

of switched circuit network (SCN) signalling over IP networks.

This protocol stack supports the inter-layer standard primitive interface

defined in SCN signalling protocol hierarchy model so as to ensure utilization of

the existing SCN signalling application without modification. It uses the

standard IP transport protocol as the transmission bottom layer, and satisfies

the special transmission requirements of SCN signalling by adding its own

functions.

PARLAY/ JAIN
Parlay/JAIN is a suite of open, standard, APIs designed to facilitate easier

access to core network capabilities from outside of the network. The opening up

of the network in a secure manner by such APIs allows the existence of new

business models, which allow applications to be developed and provided by

vendors outside of the network operators domain.

4. IP MULTIMEDIA SUBSYSTEM (IMS)

The IP Multimedia Subsystem (IMS) is a global, access-independent and
standard-based IP connectivity and service control architecture that enables

various types of multimedia services to end-users using common Internet-

based protocols. IMS is envisaged to be the heart of Next generation Network

(NGN). It is standardised by 3GPP (3rd Generation Partnership Project) and

subsequently adopted and endorsed by 3GPP2, for 3G mobile networks.

International Standards Organisations like ITU, ETSI etc. are working to adopt

it for other types of access.

 IMS allows applications in IP-enabled devices to establish peer-to-peer

and peer-to-content connections easily and securely. Once established, the IP

connection can be used to exchange all types of communication media,

including voice, video, content and more. The IMS provides a full suite of

network capabilities for authentication of clients, network-to-network interfaces

and administrative functions such as charging. Session Initiation Protocol (SIP)

is a signalling protocol that handles the setup, modification and tear-down of

multi-media sessions and RTP (Real-time Transport Protocol) provides

transport of media streams.

Figure 3 shows a converged communication network for the fixed mobile

environment. It is the IMS which introduces multimedia session control in the

packet-switched domain and at the same time brings circuit-switched

functionality in the packet switched domain. The IMS is a key technology for

such network consolidation.

Figure 3: IP Multimedia Subsystem (IMS) in Converged Networks
 5. CONCLUSION
With the gradual evolution towards NGNs, the time and direction of

change will have to be regulated with well defined specifications and with

NGN-legacy threshold point. Operators from around the globe are

implementing NGN strategies and planning to invest billions of dollars in the

rollout of NGN. Service providers are making strategies to begin rolling out

NGN based networks to take advantage of fast & flexible service creation and

provisioning capabilities, while also providing for legacy interworking and

combinational services that make use of most of the existing investments.

Operators can then build networks toward the all-IP vision offering rich

multi-access multimedia services.

IMS provides business-focused evolution options for delivering

attractive, easy-to-use, reliable and profitable multimedia services. It also

enables operators to achieve Fixed Mobile Convergence (FMC).

 SIP

Introduction
As the Internet became more popular in the 1990s, network programs

that allowed communication with other Internet users also became more

common. Over the years, a need was seen for a standard protocol that could

allow participants in a chat, videoconference, interactive gaming, or other

media to initiate user sessions with one another. In other words, a standard

set of rules and services was needed that defined how computers would

connect to one another so that they could share media and communicate.

The Session Initiation Protocol (SIP) was developed to set up, maintain, and

tear down these sessions between computers.

By working in conjunction with a variety of other protocols and

specialized servers, SIP provides a number of important functions that are

necessary in allowing communications between participants. SIP provides

methods of sharing the location and availability of users and explains the

capabilities of the software or device being used. SIP then makes it possible

to set up and manage the session between the parties. Without these tasks

being performed, communication over a large network like the Internet

would be impossible. It would be like a message in a bottle being thrown in

the ocean; you would have no way of knowing how to reach someone directly

or whether the person even could receive the message.

Beyond communicating with voice and video, SIP has also been

extended to support instant messaging and is becoming a popular choice

thats incorporated in many of the instant messaging applications being

produced. This extension, called SIMPLE, provides the means of setting up a

session in much the same way as SIP. SIMPLE also provides information on

the status of users, showing whether they are online, busy, or in some other

state of presence. Because SIP is being used in these various methods of

communications, it has become a widely used and important component of

todays communications.

Understanding SIP
SIP was designed to initiate interactive sessions on an IP network.

Programs that provide real-time communication between participants can

use SIP to set up, modify, and terminate a connection between two or more

computers, allowing them to interact and exchange data. The programs that

can use SIP include instant messaging, voice over IP (VoIP), video

teleconferencing, virtual reality, multiplayer games, and other applications

that employ single-media or multimedia. SIP doesnt provide all the

functions that enable these programs to communicate, but it is an

important component that facilitates communication between two or more

endpoints.

You could compare SIP to a telephone switchboard operator, who uses

other technology to connect you to another party, set up conference calls or

other operations on your behalf, and disconnect you when youre done. SIP

is a type of signaling protocol that is responsible for sending commands to

start and stop transmissions or other operations used by a program. The

commands sent between computers are codes that do such things as open a

connection to make a phone call over the Internet or disconnect that call

later on. SIP supports additional functions, such as call waiting, call

transfer, and conference calling, by sending out the necessary signals to

enable and disable these functions. Just as the telephone operator isnt

concerned with how communication occurs, SIP works with a number of

components and can run on top of several different transport protocols to

transfer media between the participants.

Overview of SIP
One of the major reasons that SIP is necessary is found in the nature

of programs that involve messaging, voice communication, and exchange of

other media. The people who use these programs may change locations and

use different computers, have several usernames or accounts, or

communicate using a combination of voice, text, or other media (requiring

different protocols).This creates a situation thats similar to trying to mail a

letter to someone who has several aliases, speaks different languages, and

could change addresses at any particular moment.

SIP works with various network components to identify and locate

these endpoints. Information is passed through proxy servers, which are

used to register and route requests to the users location, invite another

user(s) into a session, and make other requests to connect these endpoints.

Because there are a number of different protocols available that may be

used to transfer voice, text, or other media, SIP runs on top of other

protocols that transport data and perform other functions. By working with

other components of the network, data can be exchanged between these

user agents regardless of where they are at any given point.

It is the simplicity of SIP that makes it so versatile. SIP is an ASCII- or

text-based protocol, similar to HTTP or SMTP, which makes it more

lightweight and flexible than other signaling protocols (such as H.323). Like

HTTP and SMTP, SIP is a request-response protocol, meaning that it makes

a request of a server, and awaits a response. Once it has established a

session, other protocols handle such tasks as negotiating the type of media

to be exchanged, and transporting it between the endpoints. The reusing of

existing protocols and their functions means that fewer resources are used,

and minimizes the complexity of SIP. By keeping the functionality of SIP

simple, it allows SIP to work with a wider variety of applications.

 The similarities to HTTP and SMTP are no accident. SIP was modeled

after these text-based protocols, which work in conjunction with other

protocols to perform specific tasks. As well see later in this chapter, SIP is

also similar to these other protocols in that it uses Universal Resource

Identifiers (URIs) for identifying users. A URI identifies resources on the

Internet, just as a Uniform Resource Locator (URL) is used to identify Web

sites. The URI used by SIP incorporates a phone number or name, such as

SIP: user@syngress.com, which makes reading SIP addresses easier. Rather

than reinventing the wheel, the development of SIP incorporated familiar

aspects of existing protocols that have long been used on IP networks. The

modular design allows SIP to be easily incorporated into Internet and

network applications, and its similarities to other protocols make it easier to

use.

RFC 2543 / RFC 3261

The Session Initiation Protocol is a standard that was developed by the

Internet Engineering Task Force (IETF). The way that IETF develops a

standard is through recommendations for rules that are made through

Request for Comments (RFCs).The RFC starts as a draft that is examined by

members of a Working Group, and during the review process, it is developed

into a finalized document. The first proposed standard for SIP was produced

in 1999 as RFC 2543, but in 2002, the standard was further defined in RFC

3261. Additional documents outlining extensions and specific issues related

to the SIP standard have also been released, which make RFC 2543 obsolete

and update RFC 3261.The reason for these changes is that as technology

changes, the development of SIP also evolves. The IETF continues developing

SIP and its extensions as new products are introduced and its applications

expand.

SIP and Mbone

Although RFC 2543 and RFC 3261 define SIP as a protocol for setting up,

managing, and tearing down sessions, the original version of SIP had no

mechanism for tearing down sessions and was designed for the Multicast

Backbone (Mbone). Mbone originated as a method of broadcasting audio and

video over the Internet. The Mbone is a broadcast channel that is overlaid on

the Internet, and allowed a method of providing Internet broadcasts of

things like IETF meetings, space shuttle launches, live concerts, and other

meetings, seminars, and events. The ability to communicate with several

hosts simultaneously needed a way of inviting users into sessions; the

Session Invitation Protocol (as it was originally called) was developed in

1996.

The Session Invitation Protocol was a precursor to SIP that was

defined by the IETF MMUSIC Working group, and a primitive version of the

Session Initiation Protocol used today. However, as VoIP and other methods

of communications became more popular, SIP evolved into the Session

Initiation Protocol. With added features like the ability to tear down a

session, it was a still more lightweight than more complex protocols like

H.323. In 1999, the Session Initiation Protocol was defined as RFC 2543,

and has become a vital part of multimedia applications used today.

SIP Functions and Features

When SIP was developed, it was designed to support five specific

elements of setting up and tearing down communication sessions. These

supported facets of the protocol are:

User location, where the endpoint of a session can be identified

and found, so that a session can be established

User availability, where the participant thats being called has

the opportunity and ability to indicate whether he or she wishes

to engage in the communication

 User capabilities, where the media that will be used in the

communication is established, and the parameters of that

media are agreed upon

Session setup, where the parameters of the session are

negotiated and established

Session management, where the parameters of the session are

modified, data is transferred, services are invoked, and the

session is terminated

Although these are only a few of the issues needed to connect parties

together so they can communicate, they are important ones that SIP is

designed to address. However, beyond these functions, SIP uses other

protocols to perform tasks necessary that allow participants to communicate

with each other, which well discuss later in this chapter.

User Location
The ability to find the location of a user requires being able to translate a

participants username to their current IP address of the computer being

used. The reason this is so important is because the user may be using

different computers, or (if DHCP is used) may have different IP addresses to

identify the computer on the network. The program can use SIP to register

the user with a server, providing a username and IP address to the server.

Because a server now knows the current location of the user, other users

can now find that user on the network. Requests are redirected through the

proxy server to the users current location. By going through the server,

other potential participants in a communication can find the user, and

establish a session after acquiring their IP address.

User Availability
The user availability function of SIP allows a user to control whether he or

she can be contacted. Users can set themselves as being away or busy, or

available for certain types of communication. If available, other users can

then invite the user to join in a type of communication (e.g., voice or

videoconference), depending on the capabilities of the program being used.

User Capabilities
Determining the users capabilities involves determining what features are

available on the programs being used by each of the parties, and then

negotiating which can be used during the session. Because SIP can be used

with different programs on different platforms, and can be used to establish

a variety of single-media and multimedia communications, the type of

communication and its parameters needs to be determined. For example, if

you were to call a particular user, your computer might support video

conferencing, but the person youre calling doesnt have a camera installed.

Determining the user capabilities allows the participants to agree on which

features, media types, and parameters will be used during a session.

Session Setup
Session setup is where the participants of the communication connect

together. The user who is contacted to participate in a conversation will have

their program ring or produce some other notification, and has the option

of accepting or rejecting the communication. If accepted, the parameters of

the session are agreed upon and established, and the two endpoints will

have a session started, allowing them to communicate.

Session Management
Session management is the final function of SIP, and is used for modifying

the session as it is in use. During the session, data will be transferred

between the participants, and the types of media used may change. For

example, during a voice conversation, the participants may decide to invoke

other services available through the program, and change to a video

conferencing. During communication, they may also decide to add or drop

other participants, place a call on hold, have the call transferred, and finally

terminate the session by ending their conversation. These are all aspects of

session management, which are performed through SIP.

SIP URIs
Because SIP was based on existing standards that had already been proven

on the Internet, it uses established methods for identifying and connecting

endpoints together. This is particularly seen in the addressing scheme that

it uses to identify different SIP accounts. SIP uses addresses that are similar

to e-mail addresses. The hierarchical URI shows the domain where a users

account is located, and a host name or phone number that serves as the

users account. For example, SIP:myaccount@madeupsip.com shows that

the account myaccount is located at the domain madeupsip.com. Using this

method makes it simple to connect someone to a particular phone number

or username.

Because the addresses of those using SIP follow a

username@domainname format, the usernames created for accounts must

be unique within the namespace. Usernames and phone numbers must be

unique as they identify which account belongs to a specific person, and used

when someone attempts sending a message or placing a call to someone

else. Because the usernames are stored on centralized servers, the server

can determine whether a particular username is available or not when a

person initially sets up an account.

 URIs also can contain other information that allows it to connect to a

particular user, such as a port number, password, or other parameters. In

addition to this, although SIP URIs will generally begin with SIP:, others will

begin with SIPS:, which indicates that the information must be sent over a

secure transmission. In such cases, the data and messages transmitted are

transported using the Transport Layer Security (TLS) protocol, which well

discuss later in this chapter.

SIP Architecture
Though weve discussed a number of the elements of SIP, there are still a

number of essential components that make up SIPs architecture that we

need to address. SIP would not be able to function on a network without the

use of various devices and protocols.The essential devices are those that you

and other participants would use in a conversation, allowing you to

communicate with one another, and various servers may also be required to

allow the participants to connect together. In addition to this, there are a

number of protocols that carry your voice and other data between these

computers and devices. Together, they make up the overall architecture of

SIP.

SIP Components
Although SIP works in conjunction with other technologies and protocols,

there are two fundamental components that are used by the Session

Initiation Protocol:

User agents, which are endpoints of a call (i.e., each of the

participants in a call)

SIP servers, which are computers on the network that service requests

from clients, and send back responses

User Agents
User agents are both the computer that is being used to make a call, and

the target computer that is being called. These make the two endpoints of

the communication session. There are two components to a user agent: a

client and a server. When a user agent makes a request (such as initiating a

session), it is the User Agent Client (UAC), and the user agent responding to

the request is the User Agent Server (UAS). Because the user agent will send

a message, and then respond to another, it will switch back and forth

between these roles throughout a session.

Even though other devices that well discuss are optional to various

degrees, User Agents must exist for a SIP session to be established. Without

them, it would be like trying to make a phone call without having another

person to call. One UA will invite the other into a session, and SIP can then

be used to manage and tear down the session when it is complete. During

this time, the UAC will use SIP to send requests to the UAS, which will

acknowledge the request and respond to it. Just as a conversation between

two people on the phone consists of conveying a message or asking a

question and then waiting for a response, the UAC and UAS will exchange

messages and swap roles in a similar manner throughout the session.

Without this interaction, communication couldnt exist.

Although a user agent is often a software application installed on a

computer, it can also be a PDA, USB phone that connects to a computer, or

a gateway that connects the network to the Public Switched Telephone

Network. In any of these situations however, the user agent will continue to

act as both a client and a server, as it sends and responds to messages.

SIP Server
The SIP server is used to resolve usernames to IP addresses, so that

requests sent from one user agent to another can be directed properly. A

user agent registers with the SIP server, providing it with their username
and current IP address, thereby establishing their current location on the

network. This also verifies that they are online, so that other user agents

can see whether theyre available and invite them into a session. Because

the user agent probably wouldnt know the IP address of another user agent,

a request is made to the SIP server to invite another user into a session. The

SIP server then identifies whether the person is currently online, and if so,

compares the username to their IP address to determine their location. If the

user isnt part of that domain, and thereby uses a different SIP server, it will

also pass on requests to other servers.

In performing these various tasks of serving client requests, the SIP

server will act in any of several different roles:

Registrar server

Proxy server

Redirect server

Registrar Server
Registrar servers are used to register the location of a user agent who has

logged onto the network. It obtains the IP address of the user and associates

it with their username on the system. This creates a directory of all those

who are currently logged onto the network, and where they are located.

When someone wishes to establish a session with one of these users, the

Registrar servers information is referred to, thereby identifying the IP

addresses of those involved in the session.

Proxy Server
Proxy servers are computers that are used to forward requests on behalf of

other computers. If a SIP server receives a request from a client, it can

forward the request onto another SIP server on the network. While
functioning as a proxy server, the SIP server can provide such functions as

network access control, security, authentication, and authorization.

Redirect Server
The Redirect servers are used by SIP to redirect clients to the user agent

they are attempting to contact. If a user agent makes a request, the Redirect

server can respond with the IP address of the user agent being contacted.

This is different from a Proxy server, which forwards the request on your

behalf, as the Redirect server essentially tells you to contact them yourself.

The Redirect server also has the ability to fork a call, by splitting the

call to several locations. If a call was made to a particular user, it could be

split to a number of different locations, so that it rang at all of them at the

same time. The first of these locations to answer the call would receive it,

and the other locations would stop ringing.

Stateful versus Stateless

The servers used by SIP can run in one of two modes: stateful or

stateless. When a server runs in stateful mode, it will keep track of all

requests and responses it sends and receives. A server that operates in a

stateless mode wont remember this information, but will instead forget

about what it has done once it has processed a request. A server running in

stateful mode generally is found in a domain where the user agents resides,

whereas stateless servers are often found as part of the backbone, receiving

so many requests that it would be difficult to keep track of them.

Location Service
The location service is used to keep a database of those who have

registered through a SIP server, and where they are located. When a user

agent registers with a Registrar server, a REGISTER request is made (which

well discuss in the later section). If the Registrar accepts the request, it will

obtain the SIP-address and IP address of the user agent, and add it to the

location service for its domain. This database provides an up-to-date catalog

of everyone who is online, and where they are located, which Redirect

servers and Proxy servers can then use to acquire information about user

agents. This allows the servers to connect user agents together or forward

requests to the proper location.

SIP Requests and Responses

Because SIP is a text-based protocol like HTTP, it is used to send

information between clients and servers, and User Agent clients and User

Agent servers, as a series of requests and responses. When requests are

made, there are a number of possible signaling commands that might be

used:

REGISTER Used when a user agent first goes online and

registers their SIP address and IP address with a Registrar

server.

INVITE Used to invite another User agent to communicate, and

then establish a SIP session between them.

ACK Used to accept a session and confirm reliable message

exchanges.

OPTIONS Used to obtain information on the capabilities of

another user agent, so that a session can be established

between them. When this information is provided a session isnt

automatically created as a result.

SUBSCRIBE Used to request updated presence information on

another user agents status. This is used to acquire updated

information on whether a User agent is online, busy, offline, and

so on.
 NOTIFY Used to send updated information on a User agents

current status.This sends presence information on whether a

User agent is online, busy, offline, and so on.

CANCEL Used to cancel a pending request without terminating

the session.

BYE Used to terminate the session. Either the user agent who

initiated the session or the one being called can use the BYE

command at any time to terminate the session.

When a request is made to a SIP server or another user agent, one of a

number of possible responses may be sent back. These responses are

grouped into six different categories, with a three-digit numerical response

code that begins with a number relating to one of these categories. The

various categories and their response code prefixes are as follows:

Informational (1xx) The request has been received and is being

processed.

Success (2xx) The request was acknowledged and accepted.

Redirection (3xx) The request cant be completed and

additional steps are required (such as redirecting the user agent

to another IP address).

Client error (4xx) The request contained errors, so the server

cant process the request

Server error (5xx) The request was received, but the server

cant process it. Errors of this type refer to the server itself, and

they dont indicate that another server wont be able to process

the request.

Global failure (6xx) The request was received and the server is

unable to process it. Errors of this type refer to errors that

 would occur on any server, so the request wouldnt be forwarded

to another server for processing.

There are a wide variety of responses that apply to each of the

categories. The different responses, their categories, and codes are

shown in Table below.

Listing of Responses, Response Codes, and Their Meanings

Response Code Response Category Response Description

100 Informational Trying
180 Informational Ringing
181 Informational Call is being forwarded
182 Informational Queued
200 Success OK
300 Redirection Multiple choices
301 Redirection Moved permanently
302 Redirection Moved temporarily
303 Redirection See other
305 Redirection Use proxy
380 Redirection Alternative service
400 Client Error Bad request
401 Client Error Unauthorized
402 Client Error Payment required
403 Client Error Forbidden
404 Client Error Not found
405 Client Error Method not allowed
406 Client Error Not acceptable
407 Client Error Proxy authentication Required
408 Client Error Request timeout
409 Client Error Conflict
410 Client Error Gone
411 Client Error Length required
413 Client Error Request entities too large
414 Client Error Request-URI too large
415 Client Error Unsupported media type
420 Client Error Bad extension
480 Client Error Temporarily not available
481 Client Error Call transaction does not exis
 482 Client Error Loop detected
483 Client Error Too many hops
484 Client Error Address incomplete
485 Client Error Ambiguous
486 Client Error Busy here
500 Server Error Internal server error
501 Server Error Not implemented
502 Server Error Bad gateway
503 Server Error Service unavailable
504 Server Error Gateway time-out
505 Server Error SIP version not supported
600 Global Failures Busy everywhere
603 Global Failures Decline
604 Global Failures Does not exist anywhere
606 Global Failures Not acceptable

IP TAX IN BSNL

The current generation network of BSNL, popularly known as PSTN is

mainly circuit switching based network and it is divided into a hierarchical

architecture viz. Level I TAX exchanges, then Level-II exchanges and then

tandem/local exchanges. The PSTN network is mainly optimized for voice

calls and not much suited for data services. We have a separate network for

data services.

Today the world over trend is for a single converged network for all

type of services viz. voice, data, video which is called Next Generation

Network and is a packet switching based network. To change over from

current generation network to next generation network we have to move in a

step-by-step manner to safeguard our existing network infrastructure and

investment and therefore we have to follow an evolutionary path. IP TAX is

the first step towards the Evolution of Current Generation Network to Next

generation Network. In other words IP TAX is the replacement of existing

Level I TAX exchanges to IP based network (Packet switching network) and

rest the entire network still remaining circuit switched network. The other
reasons why we should evolve our existing network to NGN are that the

existing circuit switched networks have following problems:

Slow to develop new features and capabilities.

Expensive upgrades and operating expenses.

Proprietary vendor troubles

Large power and cooling requirements.

Limited migration strategy to new technology.

Model obsolescence.

Generic reference diagram for IP TAX is as below:

Based on the above GR the implementation plan is as below:

Setting up Two Soft Switches at New Delhi and Chennai and

Signalling Gateways at New Delhi, Chennai, Kolkota and Bangalore

Providing Trunk Media Gateways (TMGs) at 21 Level-1 locations

Providing one Announcement Servers in each IP domain i.e. one at

New Delhi and one at Chennai.

Billing interface to Centralized Billing Server at Chennai.

NMS at Chennai with FCAPS (Fault, Configuration, Accounting,

Performance, Security) capabilities.

No separate NTP server is being used in IP TAX, the existing NTP server of

our data network will be used for synchronization.

The protocols used are:

Between Softswitch and media gateway H.248

Between two softswitches - SIP(T) or BICC

Between Softswitch and Signaling gateway - sigtran

Between softswitch and Application server- sip, parley etc.

IP TAX CAPACITY (KC)

Agra 4
Ahmedabad 8
Ambala 4
Bangalore 12
Bhopal 4
Chennai 16
Coimbatore 4
Cuttack 4
Ernakulam 4
Guwahati 4
Hyderabad 8
Jaipur 4
Jalandhar 4
Kolkata 16
Lucknow 4
Mumbai 40
Nagpur 4
New Delhi 40
Patna 8
Raipur 4
Rajkot 4
Total 200

Overview of NOC Site Chennai

Overview of Soft Switch Site - New Delhi
Overview of SGW Site Bangalore & Kolkota

Overview of Other 17 Level-I Sites

 NGN Field trial in BSNL by C-DoT
C-DoT is conducting field trials of NGN in BSNL network. The trial set up is

as shown below.
 The main components used are

1. Softswich consists of

VocalTec Essentra CX

VocalTec Essentra EX and

VocalTec Essentra BAX

2. Media Gateway

Audio Codes Mediant 8000 and

Audio Codes Mediant 2000

3. Signaling Gateway

Dialogic SS7G22

4. OSS

VocalTec Essentra OSS

VocalTecs Essentra series Soft switch

Essentra BAX
Service providers are looking for ways to leverage the burgeoning broadband

access market in order to introduce new revenue generating IP

communications services. Enabling the delivery of residential and hosted

enterprise VoIP services over any broadband infrastructure, VocalTecs

Essentra BAX Broadband Access Server helps service providers take

advantage of evolving IP opportunities. Based on over a decade of leadership

in VoIP solutions, Essentra BAX, VoIP Application Server, offers service

providers a cost-effective entry into broadband VoIP services, with the

capability of scaling up to millions of subscribers over time. The carrier-

grade turnkey solution supports both traditional subscriber calling features,

including call waiting and call forward, and a range of cutting-edge features

such as call screening and click-to-dial.

 For enterprise customers, Essentra BAX supports VoIP virtual private

networks (VPNs) and an array of IP-Centrex features. Equipped with a Web-

based subscriber self-provisioning interface, the access server enables

subscribers to control their own services, thereby reducing the operating

expenses (OPEX) of service providers.

Essentra CX: A Core Control Softswitch, Essentra CX is a flexible multi-

protocol softswitch, featuring intelligent routing, control of third-party

gateways and seamless migration of H.323 networks to NGN. Operators of

wholesale and retail long-distance networks are migrating their legacy

trunking networks to VoIP-based next-generation networks (NGNs) to reduce

operating costs and enhance service flexibility. At the same time, VoIP

service providers are looking for a cost-effective solution to connect their

networks to the PSTN. Essentra CX, SIP-SS7 media gateway controller is a

scalable, carrier-grade SIP-based MGC that offers high-quality voice

services, carrier-grade reliability and maximum service flexibility. With its

open interfaces to non-proprietary media devices and application servers,

Essentra CX facilitates the creation of best-of-breed network solutions.

Essentra CX is offered in two configuration options:

1. An integrated media gateway and signaling gateway controlled by the

Essentra CX, a solution well suited for deployments requiring fewer E1/T1

spans and signaling links.

2. The Essentra SG signaling gateway and a separate media gateway both

controlled by the Essentra CX, this solution enables increased E1/T1

capacity and signaling links.

Essentra CX enables smooth migration to NGNs, while maintaining

seamless connectivity to PSTN/SS7 services. In addition to controlling the

media gateways, Essentra CX features intelligent call control, an advanced

routing engine, and PSTN/SS7 signaling support. A flexible and scalable

solution, Essentra CX facilitates the deployment and migration of existing

networks to NGNs. Taking advantage of a large SS7 protocol library and

supporting industry-standard control protocols, including Megaco/H.248

and MGCP, Essentra CX provides service providers with the capability to

seamlessly route voice and data calls between the PSTN and SIP based

packet networks. Supporting both traditional and converged services,

Essentra CX is the ideal SIP-SS7 solution for today's carriers.

Essentra EX: A Peering Manager, Essentra EX is an edge network device

enabling interconnection and peering of VoIP networks, dynamic call

routing, vendor and protocol interworking, accounting and topology hiding.

As an increasing number of carriers deploy VoIP networks the need for direct

interconnection between these networks is becoming paramount. The

Essentra EX Peering Manager provides a simple and cost-effective solution

for wholesale operators, clearinghouses, bilateral traffic exchange carriers,

operators of NLD/ILD networks and other service provider requiring secured

interconnection between disparate VoIP networks. This border element

facilitates peering between SIP and/or H.323 networks by fully addressing

carriers requirements in the areas of protocol interworking, accounting,

billing, security and routing. With full support for SIP-H.323 interworking,

Essentra EX can be deployed seamlessly with any VocalTec or third-party

softswitch platform. Essentra EX provides enhanced call-routing capabilities

enabling the support of complex routing policies that maximize network

performance and reduce transport costs in both small and large voice over

packet networks. A SIP-based service layer ensures that all service/routing

logic and network topology is hidden from the networking layer, accelerating

time to market and simplifying network management.

Essentra OSS: An Operational Support Server. Essentra OSS is a
centralized, Web-based system, enabling remote element management,

service configuration, monitoring and provisioning for Essentra products.

Media Gateway

Audio Codes Mediant 8000

Interfaces 9 STM-1 optical/copper ports or up to 240 E1

Mediant 2000

Signaling Gateway Dialogic SS7G22

 NGN Services
Class 5
Calling Line and Calling Name Identification Services
CLIP, CLIR, CNIP
Per Call CLIR
Per call UnCLIR
Name Enrichment Service
Dialling Facilities
Telephone User Interface Codes (TUIs)
Quick Dial Codes
Speed Dial Keys
Emergency Protection
Call Redirection Services
Call Forwarding Unconditional (CFU)
Call Forwarding on Busy (CFB)
Call Forwarding No Answer (CFNA)
Call Forward Un Reachable (CFNR)
Calls Forwarding can be done through
Feature Code Activation
Personal Voice Portal
Personal Web Portal
A phone Number
Contact Name
Voice Mail
Additional Call Forwarding services
Source Based Call Forwarding
Denying Incoming Call Forward calls
2 Party Loop Detection
3 Party Loop Detection
Anonymous Call Rejection
Three Party Control
Call Hold
Call Waiting
 Call Toggle / Swap
3 way conference
Call Transfer After Answer
Call Transfer during Ring (Blind Call Transfer)
Centralized Music On Hold
Code Restriction List
Network CRL
Subscription / Pre Defined CRL
Voice Mail Service
Deposit
Consultation
Direct Mode
Authentication Mode
Voice mail service number
Express messaging
Personalized Voice Mail Greeting
Voice Message Waiting Indication
Mail Call Back
E-mail Notification
Personal Web Portal
Personal Voice Portal
Video Mail
Video Calls
Do Not Disturb
Ring Hold Call
Conference Bridging
Camp on Busy/ Automatic Redial
Billing codes
Traffic Management System (TMS)
 NGN Services
Class 4
Alternate routing
Authentication
Basic traffic control
Call failure handling
Destination call routing
Dynamic routing
NLD/ILD
Lawful intercepts
Least Cost Routing (LCR)
Routing according origin/destination
Signaling Transfer Point (STP)
Time-based routing
SS7 messages manipulation
SIP-T support
Continuity check tests
Traffic measurement
E.164 extended
 Acronyms
3G 3rd Generation mobile telephony
3GPP 3rd Generation Partnership Project
ACM Association for Computing Machinery
ADSL Asymmetric Digital Subscriber Line
AGCF TISPAN PES) Access Gateway Control Function
AGW (TISPAN PES) Access GateWay
ALG Application-Level Gateway
AMR Adaptive Multi Rate
AMR-WB AMRWide Band
ANSI American National Standards Institute
A-RACF (TISPAN NGN) Access Resource and Admission Control Function
AS Application Server
ASP Application Service Provider
B2BUA (SIP) Back-to-back User Agent
BER Bit Error Rate
BGCF (3GPP IMS) Breakout Gateway Control Function
BGF (TISPANNGN) Border Gateway Function
BGP (IETF RFC 4271) Border Gateway Protocol
BHCA Busy Hour Call Attempts
BICC Bearer Independent Call Control
BRAS Broadband Remote Access Server
BS IEEE WiMAX) Base Station
CAMEL (3GPP GSM) Customized Applications of Mobile Enhanced Logic
CAP (3GPP GSM) CAMEL Application Part
CAPS Call Attempts Per Second
C-BGF (TISPANNGN) Core Border Gateway Function
CDMA Code Division Multiple Access
CDMA2000 (3GPP2) 3G packet data using CDMA
CDR Call Data Record
CHAM (3GPP OSA) Charging Control & AccountManagement
CI (3GPP GSM) Cell Identity
CMIP (ITU-T X.700) Common Management Information Protocol
CMTS Cable ModemTermination System
COPS (IETF RFC 2748) Common Open Policy Service
CORBA (OMG) Common Object Request Broker Architecture
CPE (eSG) Control Plan Editor
CPU Central Processing Unit
CSAPS Call or SMS Attempts Per Second
CSCF (3GPP IMS) Call Session Control Function
CSLAM Combined Subscriber Line Access Multiplexer
CSP Communication Service Provider
DIAMETER (IETF RFC 3588) Evolution of RADIUS
DNS (IETF RFC 1035) Domain Name Service
DSCP (IETF RFC 2474) DiffServ Code Point
DSL Digital Subscriber Line
DSLAM Digital Subscriber Line Access Multiplexer
DVB (ETSI) Digital Video Broadcasting
EAP (IETF 3748) Extensible Authentication Protocol
EDGE (3GPP) Enhanced Data rates for GSM Evolution
ENUM (IETF RFC 3761) tElepnoneNUmberMapping
ETSI European Telecommunications Standards Institute
FDD Frequency Division Duplex
FMC Fixed-Mobile Convergence
FW FireWall
GERAN (3GPP GPRS) GPRS Enhanced Radio Access Network
GGSN (3GPP GPRS) Gateway GPRS SupportNode
GPRS (3GPP) General Packet Radio Service
GSM (3GPP) Global System for Mobile communications
GTP (3GPP 09.60) GPRS Tunneling Protocol
HA (3GPP2 CDMA2000) Home Agent
HA High Availability
HLR (3GPP GSM) Home Location Register
HSDPA (3GPP) High Speed Downlink Packet Access
HSPA (3GPP)High Speed Data Access (HSDPA HSUPA)
HSS (3GPP IMS) Home Subscriber Server
HSUPA (3GPP)High Speed Uplink Packet Access
I/O Input/Output
I-BCF (TISPAN NGN) Interconnection Border Control Function
I-BGF (TISPAN NGN) Interconnection Border Gateway Function
IEEE Institute of Electrical and Electronics Engineers
IETF Internet Engineering Task Force
IM-MGW (3GPP IMS) IP Multimedia Media GateWay
IMPS (OMA) InstantMessaging & Presence Protocol
IMS (3GPP) IP Multimedia Subsystem
IN (ITU-T Q.12XX) Intelligent Network
INAP (SS7) Intelligent Network Application Part
IP (IETF) Internet Protocol
ISC (3GPP IMS) IMS Service Control interface
ISDN (ITU-T) Integrated Services Digital Network
ISIM (3GPP) IMS Subscriber Identity Module
ISO International Standards Organization
ISUP (ITU-T SS7) ISDN User Part
ITU-T International Telecommunications Union, Telecommunications
standardization sector
IUA (IETF SIGTRAN) ISDNUser Adaptation
L2TF (TISPANNGN) Layer 2 Termination function (at IP Edge)
LA (3GPP GSM) Location Area
LAN Local Area Network
LDAP (IETF RFC 4510) LightweightDirectory Access Protocol
M2PA (IETF SIGTRAN) MTP2 Peer Adaptation
M3UA (IETF SIGTRAN) MTP3 User Adaptation
MAN Metropolitan Area Network
MGC Media Gateway Controller
MGCF (3GPP IMS) Media Gateway Control Function
MGCP (IETF RFC 3435) Media Gateway Control Protocol
MGW Media GateWay
MGWF Media GateWay Function
MIB Management Information Base
MIP (IETF RFC 3344) Mobile IP
MOML (IETF) Media Objects Markup Language
MOS Mean Opinion Score
MML Man-Machine Language
MMS (OMA) Multimedia Messaging Service
MMSC (OMA) MMS Center
MPSP Media Pilot Service Provider
MRF (3GPP IMS) Media Resource Function
MRFC (3GPP IMS) Media Resource Function Controller
MRFP (3GPP IMS) Media Resource Function Point
MSC (3GPP GSM) Mobile Switching Center
MSML (IETF) Media Session Markup Language
MSRP (IETF SIMPLE)Message Session Relay Protocol
MTP (ITU-T SS7 Q.70X) Message Transfer Part
MVNO Mobile Virtual Network Operator
NAPTR (IETF RFC 3403) Naming Authority Pointer Resource
NASS (TISPANNGN) Network Attachment SubSystem
NAT Network Address Translation
NGN (IETF) Next Generation Network
NMS (TMN) Network Management System
NP Number Portability
OFDM Orthogonal Frequency Division Multiplexing
OFDMA Orthogonal Frequency Division Multiple Access
OMA Open Mobile Alliance
OMG Object Management Group
ORB (OMG CORBA) Object Request Broker
OSA (3GPP) Open Service Access
OTA Over-The-Air
PAN Personal Access Network
PBX Private Branch eXchange
PCF (3GPP IMS) Policy Control Function
PCPSP Personal Contact Page Service Provider
PCU 3GPP GPRS) Packet Control Unit
PDF (3GPP IMS) Policy Decision Function
PDP (3GPP GPRS) Packet Data Protocol
PDP (IETF COPS) Policy Decision Point
PDSN (3GPP2 CDMA2000) Packet Data ServingNode
PEF (IETF COPS) Policy Enforcement Point
PES (TISPAN) PSTN Emulation Subsystem
PKI Public Key Infrastructure
POTS Plain Old Telephone Service (delivered by a PSTN)
PPP (IETF RFC 1661) Point to Point Protocol
PPPoE (IETF RFC 2516) PPP over Ethernet
PRI (ITU-T ISDN) Primary Rate Interface
PSTN Public Switched Telephone Network
QoS (IETF IP)Quality of Service
RA (3GPP GPRS) Routing Area
RACS (TISPANNGN) Resource & Admission Control Subsystem
RADIUS (IETF RFC 2864) Remote Authentication Dial-In User Suite
RCEF (TISPAN NGN) Resource Control Enforcement Function (at IP
edge)
RGW (TISPAN PES) Residential GateWay
RIP (IETF RFC 2453) Routing Information Protocol
RNC (3GPP UMTS) Radio Network Controller
RSS Really Simple Syndication
RSVP (IETF RFC 2205) Resource ReSerVation Protocol
RTCP (IETF RFC 3550) Real Time Control Protocol
RTP (IETF RFC 3550) Real Time Protocol
RTSP (IETF RFC 2326) Real Time Streaming Protocol
SBC Session Border Controller
SCCP (ITU-T SS7 Q.71X) Signaling Connection Control Part
SCCP Skinny Client Control Protocol
SCF (3GPP OSA) Service Capability Feature
SCP (ITU-T IN) Service Control Point
SCS (3GPP OSA) Service Capability Server
SCTP (IETF RFC 2960) Stream Control Transmission Protocol
SDP (IETF RFC 2327) SessionDescription Protocol
SGSN (3GPP GPRS) Serving GPRS SupportNode
SIGTRAN (IETF) Signaling TRANsport
SIM (3GPP GSM) Subscriber Identity Module
SIMPLE (IETF) SIP for Instant Messaging and Presence Leveraging
Extensions
SIP (IETF) Session Initiation Protocol (RFC 3261)
SMS (3GPP) Short Message Service
SMSC (3GPP) SMS Center (SMS Interworking and Gateway
Center)
SNMP (IETF) Simple Network Management Protocol
SOAP (W3C) Simple Object Access Protocol
SP Service Provider
SPDF (TISPAN NGN) Service Policy Decision Function
SS (IEEE WiMAX) Subscriber Station
SS7 (ITU-T) Signaling System No. 7
SUA (IETF SIGTRAN) SCCP User Adaptation
TCAP (ITU-T SS7 Q.77X) Transaction Capabilities Application Part
TCP (IETF RFC 793) Transmission Control Protocol
TDD Time Division Duplex
TDM Time Division Multiplexing
TGCF Trunking Gateway Control Function ( a MGCF)
TGW (TISPAN PES) Trunking GateWay ( a MGW)
TIA Telecommunications Industry Association
TISPAN Telecoms and Internet converged Services and Protocols for
Advanced Networks
TLS (IETF RFC 4346) Transport Layer Security
TPS Transactions Per Second
TRIP (IETF RFC 3219) Telephony Routing over IP
UA (IETF SIP) User Agent
UDP (IETF RFC 768) User Datagram Protocol
UE (3GPP IMS) User Entity
UMA (3GPP) Unlicensed Mobile Access
UML (OMG) Unified Modeling Language
UMTS (3GPP) Universal Mobile Telecommunications System
USIM (3GPP) Universal Subscriber Identity Module
USSD (3GPP GSM) Unstructured Supplementary Service Data
UTRAN (3GPP UMTS) UMTS Terrestrial Radio Access Network
V5UA (IETF SIGTRAN) V5 interface User Adaptation
VAS Value-Added Service
VPN Virtual Private Network
WIN (TIA IS-826) Wireless IN
WPAN Wireless PAN
XCAP (IETF SIMPLE) XML Configuration Access Protocol
XML (W3C) eXtensible Markup Language
XMPP (IETF RFC 3920-3923) eXtensible Messaging and Presence
Protocol
W3C World Wide Web Consortium
WAN Wide Area Network
WiFi (IEEE 802.11) Wireless Fidelity
WiMAX (IEEE 802.16-2004) Worldwide Interoperability for Microwave
Access
WIN (TIA IS-826) Wireless Intelligent Network