1. Application/Management Part
D. NGN architecture -NGN functional model
Application/Management Part
Application Servers
Management Servers
Parlay/LDAP
SNMP …
Session Part
(Call control)
Softswitches
MGCP
Megaco/H.248 …
Access Layer
Media Gateways
API - Application Programming Interface

2. D. NGN architecture (Cntd.)
IP network
Softswitch
Application Server
Network Management
Server
Media Gateway
Multiservice Access
PSTN, GSM, ATM, ...
Services 1) 2)
Transport

3. 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
controller of a media gateway
Intelligent Database - Network directory,
Billing, Call records
Intelligent
Database (ID) . .
API
Parlay/OSA/LDAP
ID/MGC
ID/SG . .
Signaling
Gateway (SG)
H.323/SIP/SIP-T/
SIGTRAN .
CC7
ISUP (MTP)
SG/MGC
MG Controller
(MGC)
MGC/MGC . .
MGC/MG
MGCP/Megaco(H.248) . .
Media Gateway
(MG)
RTP Packet Flow
(Voice/Data/MM)
TDM Flow (Voice)

4. D. NGN architecture – possible NGN configuration
Network Manager
Application Server IB
AAA
SNMP
RADIUS
API (PARLAY/LDAP)
Softswitch
SIP/SIP-T
H.323/BICC SG
SIGTRAN
SS7
Switch
STP
PSTN/ISDN
SS7
Switch
STP
PSTN/ISDN
SS7
ISUP/MTP SG
SIGTRAN
ISUP
Softswitch
SIP
A PSTN 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 or, if the packet destination is not another PSTN, to a Softswitch.
MGC
MGCP/Megaco/H.248
Gatekeeper/
Proxy Server
Mobile Networks/
IMS
H.323/SIP
Media Gateway
Media Gateway
Core IP Network (QoS)
Н.323/ IP Network

5. 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

6. E. NGN building blocks (Cntd.)
Application Server IB
AAA
SOFTSWITCH
Signaling Gateway
Media Gateway Controller
Network Manager
Gatekeeper
АDSL
POTS
ISDN
PRI
V5.x
(VoIP)
Multiservice Access
Multiplexer
Media Gateway

7. 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.

8. 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 IP address

9. Softswitch Signaling Gateway Media Gateway Controller
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

10. 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
SS7 Signaling
SIGTRAN
ISUP
IP Network
Signaling Gateway
PSTN

11. 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.

12. 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.

13. 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:

14. 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.

15. 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.

16. 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.

17. 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).

18. 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.

19. F. NGN protocols and mechanisms
Signaling Protocols
H.323
SIP
MGCP
Megaco/H.248
SIP-T
SIGTRAN
Mechanisms (QoS, Resource Allocation)
MPLS
IntServ
DiffServ

20. VoIP protocols: 1. H.323, ITU-T
H 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
The first call control standard for VoIP was the H.323, which was adopted by the International Telecommunications Union (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 recommendations also included a standard called T.120, which is implemented in data collaboration tools such as Microsoft’s NetMeeting.
The H.323 recommendation is made up of multiple call control protocols. The audio streams are transacted using the real-time protocol/real-time control protocol (RTP/RTCP). However, some vendors felt that H.323 was too broad a standard and lacked efficiency. It also does not guarantee business voice quality.

21. 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.
To counter this, the Internet Engineering Task Force (IETF) Multi-party Multimedia Session Control working group came up with the Session Initiation Protocol (SIP), which claims to deliver faster call-establishment times. It also provides for ways to leverage the Internet and Web infrastructures.
The Session Initiation Protocol (SIP) is an Internet Engineering Task Force standard protocol for initiating an interactive user session that involves multimedia elements such as video, voice, chat, gaming, and virtual reality.
Like HTTP or SMTP, SIP works in the Session layer of the Open Systems Interconnection (OSI) communications model. The Application layer is the level responsible for ensuring that communication is possible. SIP can establish multimedia sessions or Internet telephony calls, and modify, or terminate them. The protocol can also invite participants to unicast or multicast sessions that do not necessarily involve the initiator. Because the 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.

22. 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.
SIP is a request-response protocol, dealing with requests from clients and responses from servers. Participants are identified by SIP URLs. Requests can be sent through any transport protocol, such as UDP, or TCP. SIP determines the end system to be used for the session, the communication media and media parameters, and the called party's desire to engage 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.

23. 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)
Most recently, Telcordia (formerly Bellcore) and Level 3, with the support of Cisco
Systems, announced the Media Gateway Control Protocol (MGCP). MGCP is a control
protocol that specifically addresses the control of media gateways (it is not a protocol
that specifies complete end-to-end communications, as H.323 does). MGCP is a
"state" protocol, in which a media gateway controller, or MGC (or call agent) acts as
the master controller of a media gateway. MGCP assumes that all call-control intelligence
is external to the gateway; H.323, by comparison, assumes that end stations are
fairly intelligent.
The ITU and the IETF have joined together to produce a new standard that combines
elements of the IETF’s MGCP and the ITU’s H.323. That standard is known as Megaco
within the IETF and H.248 within the ITU. The main features of Megaco are to allow
greater scaling than H.323 allows, and to address the technical requirements of multimedia
conferencing. Although based on MGCP, Megaco is more complex than
MGCP (for one thing MGCP does not address multimedia conferencing).
IP offers a standardized transport layer and voice is an application that rides on top
of that transport. At the applications level, the standards for voice over IP are still
evolving, which means most business voice over IP solutions today are proprietary
and do not interoperate with one another, but this will change as standards evolve.

24. 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.

25. 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.

26. 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.

27. 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.

28. 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.

29. G. NGN as converged networks: concluding remarks
PSTN
Switch
Data networks
Flexible bandwidth
Effective transmission
Services
QoS
SOFTSWITCH
Voice services for IP-users
VoIP