D.

NGN architecture -NGN functional model

Application Servers Application/Managemen

Management Servers t Part

Parlay/LDAP
SNMP

Session Part
Softswitches
(Call control)

MGCP
Megaco/H.248

Media Gateways Access Layer

API - Application Programming Interface

 D. NGN architecture (Cntd.)

Softswitch
Application Server Services
Network Management
Server

PSTN, GSM, ATM, ...

IP network Media Gateway Transport

Multiservice Access
 D. ITU-T NGN architecture (Y.1001) and
corresponding protocols

IP Network IW Functions PSTN/ISDN

Softswitch includes MGC, SG, Call Agent
Media Gateway is protocol converter
Media Gateway Controller is master
Intelligent
controller of a media gateway Database (ID)
Intelligent Database - Network directory,
Billing, Call records ID/MGC . ID/SG . API
Parlay/OSA/LDAP

. Signaling
Gateway (SG)
.
H.323/SIP/SIP-T/ CC7
SIGTRAN SG/MGC . ISUP (MTP)

MGC/MGC MG Controller
. (MGC)
.
MGC/MG MGCP/Megaco(H.248)

. Media Gateway
(MG)
.
RTP Packet Flow TDM Flow (Voice)
(Voice/Data/MM)
 D. NGN architecture possible NGN configuration

Network
Application Server Manager
IB AAA
SNMP
API RADIUS
(PARLAY/LDAP)
SIP/SIP-T Softswitch
SG H.323/BICC
SIGTRAN
SG SS7
SIGTRAN
ISUP ISUP/MTP
STP Switch STP
Switch SS7
SS7 Softswitch
SIP PSTN/ISDN
PSTN/ISDN
MGC
MGCP/Megaco/H.248
Gatekeeper/
Proxy Server
Media
Gateway Media
Gateway Mobile Networks/
H.323/SIP
IMS
Core IP Network (QoS)

.323/ IP Network
 E. NGN building blocks
Media Gateway - protocol converter
Media Gateway Controller - master controller of a
media gateway
Softswitch = MGC + SG
Signaling Gateway
Application Server Information Database (ID) -
Network directory, Billing, Call records, Authentication,
authorization, and accounting (AAA)
Network Manager Operation, Administration,
Management (OAM); provides network elements
management from a centralized web interface
 E. NGN building blocks (Cntd.)
Application
Server
IB AAA SOFTSWITCH
Signaling Gateway
Media Gateway Controller Network
Gatekeeper Manager

DSL
POTS
ISDN
PRI
V5.x (VoIP)
Multiservice Access
Multiplexer Media
Gateway
 E. Main NGN building blocks (Cntd.)
Media Gateway (IETF RFC 3015)

Media gateway (MG) protocol converter between different types

of networks (Example MG between circuit-switched voice
network - TDM flows, and the IP network - RTP packet flows.
MG processes incoming calls via requests to the Application
Server using HTTP.

The media gateway (MG) terminates IP and circuit-switched

traffic. MGs relay voice, fax, modem and ISDN data traffic over the
IP network using Quality of Service enabled IP technology.
 Media Gateway (IETF RFC 3015)
All types of traffic (voice, data, video)

Control (from Media Gateway Controller): MGCP, Megaco/H.248

Interfaces: STM-1to transport network, E1 to PSTN; Eth-Fast/Gb to

IP network

Voice Packetization/Compression (Codecs: ITU-T G.711, G.723.1, G.726,

G.729A

Echo cancellation: ITU-T G.165, G.168

QoS via DiffServ and ToS bits marking

Mapping addresses: E.164 IP address

 Softswitch
Signaling Gateway
Signaling Gateway (SG) offers a consolidated signaling
interface - SS7 signaling point for the NGN platform.
Also, SG supports a SIGTRAN interface (IETF SS7 telephony
signaling over IP) as well as IP Proxy functions (SIP).

Media Gateway Controller

MGC acts as the master controller of a media gateway
Supervises terminals attached to a network
Provides a registration of new terminals
Manages E.164 addresses among terminals
 Signaling Gateway Function
Several millions BHCA
Several hundreds controlled trunk ports
Control: MGCP, MEGACO, SIP
Signaling: ISUP, H.323, SIP, SIP-T, INAP, SIGTRAN
Mgmt: SNMP

IP Signaling SIGTRAN ISUP SS7 Signaling

IP Network Signaling Gateway PSTN

 Application Server
Application Server (AS) consists a number of modular
application building blocks; server generates VoiceXML pages.
(VoiceXML is a standards-based scripting language for developing
voice-enabled software applications)

The modular design of the next generation communications

platform makes it easy to deploy enhanced services such as
unified communications solutions, multimedia messaging services,
and presence & availability management applications.
 Application Server
Application Server generates application documents (VoiceXML pages) in
response to requests from the Media Gateway via the internal Ethernet
network.

The application server leverages a web application infrastructure to interface

with data stores (messages stores, user profile databases, content servers)
to generate documents (e.g., VoiceXML pages).

AS provide interoperability between applications like WAP, HTML, and voice

allowing the end user to simultaneously input voice command and receive
presentation via WAP or HTML.
 Parlay
Parlay is an evolving set of specifications for industry-standard
application programming interfaces (APIs) for managing network
"edge" services:
call control
messaging
content-based charging.

Parlay specifications are being developed by the Parlay Group, a

consortium of member companies that include AT&T, BT, Cisco, IBM,
Lucent, Microsoft, Nortel Networks, and others.
Use of the Parlay specifications is expected to make it easier to add
new cross-platform network applications so that users need not
depend solely on the proprietary offerings of carriers.
The Parlay Group is not a standards group itself, but sees itself as a
facilitator of needed interfaces. Application program interfaces are or
will be defined for:
 Parlay
Authentication
Integrity management
Operations, administration, and maintenance (OA&M)
Discovery (of the closest provider of a service)
Network control
Mobility
Performance management
Audit capabilities
Generic charging and billing
Policy management
Mobile M-commerce/E-commerce
Subscriber data/user profile/virtual home environment (VHE)
The Parlay APIs are said to complement and encourage use of the
Advanced Intelligent Network (AIN) protocols.
Authentication, Authorization, Accounting (AAA)
Authentication, Authorization, Accounting (AAA) is a term
for a framework for intelligently controlling access to computer
resources, enforcing policies, auditing usage, and providing the
information necessary to bill for services. These combined
processes are considered important for effective network
management and security.
As the first process, authentication provides a way of identifying
a user, typically by having the user enter a valid user name and
valid password before access is granted. The process of
authentication is based on each user having a unique set of
criteria for gaining access. The AAA server compares a user's
authentication credentials with other user credentials stored in a
database. If the credentials match, the user is granted access to
the network. If the credentials are at variance, authentication
fails and network access is denied.
 Authentication, Authorization, Accounting (AAA)

Following authentication, a user must gain

authorization for doing certain tasks. After logging into
a system, for instance, the user may try to issue
commands. The authorization process determines
whether the user has the authority to issue such
commands. Simply put, authorization is the process of
enforcing policies: determining what types or qualities
of activities, resources, or services a user is permitted.
Usually, authorization occurs within the context of
authentication. Once you have authenticated a user,
they may be authorized for different types of access or
activity.
 Authentication, Authorization, Accounting (AAA)
The final term in the AAA framework is accounting, which
measures the resources a user consumes during access. This
can include the amount of system time or the amount of data a
user has sent and/or received during a session. Accounting is
carried out by logging of session statistics and usage
information and is used for authorization control, billing, trend
analysis, resource utilization, and capacity planning activities.
Authentication, authorization, and accounting services are often
provided by a dedicated AAA server, a program that performs
these functions. A current standard by which network access
servers interface with the AAA server is the Remote
Authentication Dial-In User Service (RADIUS).
 RADIUS
Remote Authentication Dial-In User Service (RADIUS) is a
client/server protocol and software that enables remote access
servers to communicate with a central server to authenticate
dial-in users and authorize their access to the requested
system or service. RADIUS allows a company to maintain user
profiles in a central database that all remote servers can share.
It provides better security, allowing a company to set up a policy
that can be applied at a single administered network point.
Having a central service also means that it's easier to track
usage for billing and for keeping network statistics. Created by
Livingston (now owned by Lucent), RADIUS is a de facto
industry standard used by a number of network product
companies and is a proposed IETF standard.
 F. NGN protocols and mechanisms
Signaling Protocols
H.323
SIP
MGCP
Megaco/H.248
SIP-T
SIGTRAN
Mechanisms (QoS, Resource Allocation)
MPLS
IntServ
DiffServ
 VoIP protocols:
1. H.323, ITU-T
H.323 - first call control standard for multimedia networks.
Was adopted for VoIP by the ITU in 1996
H.323 is actually a set of recommendations that define how
voice, data and video are transmitted over IP-based networks
The H.323 recommendation is made up of multiple call control
protocols. The audio streams are transacted using
the RTP/RTCP
In general, H.323 was too broad standard without sufficient
efficiency. It also does not guarantee business voice quality
 VoIP protocols:
2. SIP - Session Initiation Protocol, IETF (Internet
Engineering Task Force)

SIP - standard protocol for initiating an interactive user session

that involves multimedia elements such as video, voice, chat,
gaming, and virtual reality. Protocol claims to deliver faster call-
establishment times.
SIP works in the Session layer of IETF/OSI model. SIP can
establish multimedia sessions or Internet telephony calls. SIP
can also invite participants to unicast or multicast sessions.
SIP supports name mapping and redirection services. It makes
it possible for users to initiate and receive communications and
services from any location, and for networks to identify the
users wherever they are.
 VoIP protocols :
2. SIP - Session Initiation Protocol, IETF (Internet
Engineering Task Force) (Cntd)
SIP client-server protocol, Rq from clients, Rs from servers.
Participants are identified by SIP URLs. Requests can be sent
through any transport protocol, such as UDP, or TCP.
SIP defines the end system to be used for the session, the
communication media and media parameters, and the called
party's desire to participate in the communication.
Once these are assured, SIP establishes call parameters at
either end of the communication, and handles call transfer and
termination.
The Session Initiation Protocol is specified in IETF Request
for Comments (RFC) 2543.
 VoIP protocols :
3. MGCP/Megaco/H.248

MGCP - Media Gateway Control Protocol, IETF

[Telcordia (formerly Bellcore)/Level 3/Cisco]

MGCP control protocol that specifically addresses the

control of media gateways

Megaco/H.248 (IETF, ITU) - standard that combines

elements of the MGCP and the H.323, ITU (H.248)

The main features of Megaco - scaling (H.323) and

multimedia conferencing (MGCP)
 SIP-T
SIP-T (SIP for telephones, previously SIP-BCP-T) is a
mechanism that uses SIP to facilitate the interconnection of the
PSTN with IP. SIP-T defines SIP functions that map to ISUP
interconnection requirements.
This is intended to allow traditional IN-type services to be
seamlessly handled in the Internet environment. It is essential
that SS7 information be available at the points of PSTN
interconnection to ensure transparency of features not
otherwise supported in SIP. SS7 information should be
available in its entirety and without any loss to the SIP network
across the PSTN-IP interface.
 SIGTRAN
SIGTRAN (for Signaling Transport) is the standard
Telephony Protocol used to transport Signaling System 7
signals over the Internet. SS7 signals consist of special
commands for handling a telephone call.
Internet telephony uses the IP PS connections to
exchange voice, fax, and other forms of information that
have traditionally been carried over the dedicated CS
connections of the public switched telephone network
(PSTN). Calls transmitted over the Internet travel as
packets of data on shared lines, avoiding the tolls of
PSTN.
 SIGTRAN
A telephone company switch transmits SS7 signals to a SG. The gateway,
in turn, converts the signals into SIGTRAN packets for transmission over IP
to either the next signaling gateway.

The SIGTRAN protocol is actually made up of several components (this is

what is sometimes referred to as a protocol stack):
standard IP
common signaling transport protocol (used to ensure that the data
required for signaling is delivered properly), such as the Streaming
Control Transport Protocol (SCTP)
adaptation protocol that supports "primitives" that are required by
another protocol.
 SIGTRAN

The IETF Signaling Transport working group has

developed SIGTRAN to address the transport of
packet-based PSTN signaling over IP Networks,
taking into account functional and performance
requirements of the PSTN signaling. For
interworking with PSTN, IP networks will need to
transport signaling such as Q.931 or SS7 ISUP
messages between IP nodes such as a
Signaling Gateway and Media Gateway
Controller or Media Gateway. Applications of
SIGTRAN include Internet dial-up remote access
and IP telephony interworking with PSTN.
 SCTP
TCP transmits data in a single stream (sometimes called a byte
stream) and guarantees that data will be delivered in sequence to
the application or user at the end point. If there is data loss, or a
sequencing error, delivery must be delayed until lost data is
retransmitted or an out-of-sequence message is received. SCTP's
multi-streaming allows data to be delivered in multiple, independent
streams, so that if there is data loss in one stream, delivery will not
be affected for the other streams. For some transmissions, such as
a file or record, sequence preservation is essential. However, for
some applications, it is not absolutely necessary to preserve the
precise sequence of data. For example, in signaling transmissions,
sequence preservation is only necessary for messages that affect
the same resource (such as the same channel or call). Because
multi-streaming allows data in error-free streams to continue
delivery when one stream has an error, the entire transmission is
not delayed.
 G. NGN as converged networks:
concluding remarks
PSTN Data networks

Switch Switch
Switch
Switch

Flexible
QoS bandwidth

Services Effective transmission

SOFTSWITCH

Voice services for IP-users

VoIP