Application/Managemen

t Part

Session Part
(Call control)

Access Layer

Parlay/LDAP
SNMP

MGCP
Megaco/H.248


D. NGN architecture -NGN functional model
Application Servers
Management Servers
Softswitches

Media Gateways
API - Application Programming Interface

D. NGN architecture (Cntd.)
IP network
Softswitch
Application Server
Network Management
Server
Media Gateway
Multiservice Access
PSTN, GSM, ATM, ...
Services
Transport
D. ITU-T NGN architecture (Y.1001) and
corresponding protocols
PSTN/ISDN
IP Network IW Functions
Intelligent
Database (ID)
Signaling
Gateway (SG)
.
MG Controller
(MGC)
Media Gateway
(MG)
.
.
.
. ID/SG
CC7
ISUP (MTP)

MGC/MG
RTP Packet Flow
(Voice/Data/MM)
TDM Flow (Voice)
. .
.
MGC/MGC
SG/MGC
ID/MGC
H.323/SIP/SIP-T/
SIGTRAN

API
Parlay/OSA/LDAP
MGCP/Megaco(H.248)
.
Softswitch includes MGC, SG, Call Agent
Media Gateway is protocol converter
Media Gateway Controller is master
controller of a media gateway
Intelligent Database - Network directory,
Billing, Call records

D. NGN architecture possible NGN configuration

SG
Media
Gateway

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

Softswitch
MGC
SNMP
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

Application
Server


Gatekeeper
Network
Manager
SOFTSWITCH
Signaling Gateway
Media Gateway Controller

Multiservice Access
Multiplexer
DSL
POTS
ISDN
PRI
V5.x

(VoIP)
Media
Gateway
E. NGN building blocks (Cntd.)
IB
AAA
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



IP
SCTP/IP MTP




SS7
SGW
Signaling Gateway Function
SIGTRAN ISUP IP Signaling SS7 Signaling
IP Network PSTN Signaling Gateway
Several millions BHCA
Several hundreds controlled trunk ports
Control: MGCP, MEGACO, SIP
Signaling: ISUP, H.323, SIP, SIP-T, INAP, SIGTRAN
Mgmt: SNMP



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
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.


Voice services for IP-users
PSTN
Switch
Switch

Switch

Switch
Services
QoS
VoIP
Data networks
Flexible
bandwidth
Effective transmission
G. NGN as converged networks:
concluding remarks
SOFTSWITCH