Technology White Paper

NGN PROTOCOLS
INTRODUCTION
NGN architecture is characterised by the separation of service, transport and control layers,
which are inter connected by open interfaces and use standards protocols. Legacy TDM
networks are interconnected with NGN via interfaces based on open standards and protocols.
This paper on ‘NGN Protocols’ describes some of the standard protocols used in NGN
architecture.
A protocol is set of rules that govern the control connections, communications and data
HT TH HT TH

transfer between two computing devices. A protocol stack denotes a specific combination of
protocols that work together.
A protocol stack typically used in NGN is shown in figure 1.

APPLICATION LAYER PROTOCOLS

ISDN SS7 ISUP SS7 V5.2

· Q.931 SCCP MTP3

User Adaoption H.323 H.248 SIP RTCP Media

(IUA, M3UA, M2UA, M2PA SUA, V5UA)

Sigtran RTP

SCTP TCP UDP

IPv4/IPv6

Network Layer Protocols

Fig 1: Protocol stack for NGN

Figure 2 shows how the protocols shown in figure 1 are used for signalling and media
streams in NGN environment.
Control signalling messages are transported using SIGTRAN, H.248, SIP, H.323 etc. which
are summerised in Annexure -1.
Media streams, which consist of audio, video or data, or a combination of any of them,
convey user or application data (i.e., a payload) but not control data. These are transported
through RTP (Real-time Transport Protocol). RTCP (Real-time Transport Control Protocol)
Switching Division, TEC
 Technology White Paper on NGN Protocols
controls the delivery of packetised media streams over RTP.
MGC/Softswitch
IN
Switch MGC/Softswitch

INAP IP SIP-I/SIP-T, H.323

SS7 SIGTRAN SIGTRAN SS7

TDM TDM
Local/ SG SG Local/
H.248 H.248
Transit Transit
Voice SIP SIP
Switch MG MG Voice Switch

IP IP
H.248
H.248
Analog/ RTP/RTCP Analog/
ISDN ISDN
AGW AGW

AGW: Access Gateway, MG: Media Gateway, SG: Signalling gateway, MGC: Media Gateway Controller, TDM: Time Division Multiplex

Figure 2: Protocols in NGN

2. SIGTRAN (Signalling Transport)

Sigtran refers to a protocol stack for transporting Switched Circuit Network (SCN) signalling
protocols (SS7, ISDN, V5.2 etc.) over an IP network. It encapsulates and carries SCN
protocols over IP networks. SIGTRAN is defined in IETF RFC 2719.
The SIGTRAN protocol stack consists of three components, a standard IP stack, a common
signalling transport protocol and an Adaptation layer as shown in figure 3.

Application Layers

Adaptation Protocol
SIGTRAN
Architectural
Common Signalling Transport Protocol Model

Standard Internet Protocol (IP)

Figure 3: SIGTRAN Protocol Stack Model

Switching Division, TEC Page 2 of 15

 Technology White Paper on NGN Protocols
A standard IP stack: The function of IP stack is to deliver IP packets where they are
supposed to go. IP is considered as the most suitable protocol for transportation of
messages. It provides an efficient way to transport user data.
A common signalling transport protocol: A protocol that supports reliable transfer of data
is required for signalling transport functions. A new protocol, SCTP (Stream Control
Transmission Protocol) is designed by IETF to transport SCN signalling messages over IP
networks. It operates on the top of IP at the same level as TCP. Although TCP provides
reliable transfer of data through acknowledgement mechanism and order of transmission
delivery through sequence mechanism, but it imposes several limitations for new emerging
applications. For example some applications require reliable transfer of data without
sequence mechanisms and some require partial ordering of data. Moreover TCP is not
appropriate for real time applications since it adds unnecessary delay.
SCTP's basic service is connection oriented reliable transfer of messages between peer
SCTP users. SCTP supports multi-homing and multi-streaming.
• Multi-homing: is the ability of an association (i.e a connection) to support multiple IP
addresses or interfaces at a given end point. In case of network failures, use of more than
one address could allow re-routing of packets, and also provide an alternate path for
retransmissions. A single port number is used across the entire address list at an endpoint
for a specific session.
• Multi-streaming: Allows for multiple virtual connections on the same physical line. Each
user application might be assigned its own stream (virtual connection). 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. The SCTP user
can specify at association startup time the number of streams to be supported by the
association.
SCTP transport service can be fragmented into following functionalities:
• Acknowledged error-free non-duplicated transfer of user data.
• Data fragmentation to conform to discovered path MTU size.*
• Sequenced delivery of user messages within multiple streams, with an option for order-
of-arrival delivery of individual user messages.
• Optional bundling of multiple-user messages into a single SCTP packet.
• Network-level fault tolerance through supporting of multi-homing at either or both ends of
an association.
An Adaptation layer: The function of the adaptation layer is to provide the same functions to
its upper layer in IP network as provided by the corresponding layer in SCN protocol stack.
Fore example M3UA provides the equivalent set of primitives at its upper layer as provided by
the MTP3 to its local MTP3 users at the SS7 signalling end point. In this way ISUP/SCCP
layers are not aware that the expected services are offered remotely from an MTP3 layer at
an SG and not by a local MTP3 layer and vice versa. Protocols defined for this layer are
M2PA, M2UA, M3UA, SUA, IUA and V5UA, however, exact use of Adaptation Layer Protocol
depends upon the application and implementation in the network.
* When one IP host has a large amount of data to send to another host, the data is transmitted as a series of IP datagrams. It is usually preferable that these
datagrams be of the largest size that does not require fragmentation anywhere along the path from the source to the destination. This datagram size is
referred to as the Path MTU (PMU), and it is equal to the minimum of the MTUs of each hop in the path. When needed, SCTP fragments user messages to
ensure that the SCTP packet passed to the lower layer conforms to the path MTU On receipt, fragments are reassembled into complete messages before
being passed to the SCTP user.
Switching Division, TEC Page 3 of 15
 Technology White Paper on NGN Protocols

2.1 M2PA (MTP2 peer-to-peer Adaptation Layer Protocol)

MP2A defines the protocol supporting the transport of SS7 MTP3 messages over IP using the
services of the SCTP. M2PA operates similar to MTP2 so as to provide peer-to-peer
communication between SS7 endpoints.
SS7 Signalling end point Media Gateway Controller
Signalling Gateway

S7AP S7AP

SCCP/ SCCP/
ISUP ISUP
MTP3
MTP3 MTP3
MTP2 MTP2 M2PA M2PA
SCTP SCTP
MTP1 MTP1
IP
IP

IP
SS#7 NETWORK

2.2 M2UA (MTP2 User Adaptation Layer Protocol)

M2UA is a protocol for the backhauling of SS7 MTP3 messages over IP using the services of
SCTP. This protocol is used between a Signalling Gateway (SG) and a Media Gateway
Controller (MGC).
SS7 Signalling end point Signalling Gateway Media Gateway Controller

SCCP
/ISUP SCCP
Nodal Interworking /ISUP
MTP3 Function MTP3

MTP2 MTP2 M2UA M2UA

SCTP SCTP
MTP1 MTP1
IP IP

SS#7 IP
NETWORK

Switching Division, TEC Page 4 of 15

 Technology White Paper on NGN Protocols
2.3 M3UA (MTP3 User Adaptation Layer Protocol)

M3UA supports the transport of any SS7 MTP3-User signalling (i.e, ISUP and SCCP
messages) to an IP Signalling Point (IPSP) using the services of SCTP.

SS7 Signalling end point Signalling Gateway Media Gateway Controller

SCCP Nodal Interworking SCCP

/ISUP Function /ISUP
MTP3 MTP3 M3UA M3UA

MTP2 MTP2 SCTP SCTP

MTP1 MTP1 IP IP

IP
SS#7 NETWORK

2.4 SUA (Signalling Connection Control Part User Adaptation Layer protocol)

SUA defines a protocol for the transport of any SS7 SCCP-user signalling message such as
TCAP (Transaction Capabilities Application Protocol) and RANAP (Radio Access Network
Application Protocol) over IP using SCTP services.

SS7 Signalling end point Signalling Gateway Media Gateway Controller

S7AP Nodal Interworking S7AP

Function

SCCP SCCP SUA SUA

MTP3 MTP3 SCTP SCTP

MTP2 MTP2
IP IP
MTP1 MTP1

IP
SS#7 NETWORK

Switching Division, TEC Page 5 of 15

 Technology White Paper on NGN Protocols
2.5 IUA (ISDN User Adaptation Layer Protocol)

IUA defines an adaptation module that is suitable for the transport of ISDN Q.921-User
Adaptation Layer (e.g., Q.931) messages.
ISDN End Point Access Media Gateway (AGE) Media Gateway Controller

Q.931 Nodal Interworking Q.931

( NIF ) Function

IUA IUA

SCTP SCTP
Q.921
Q.921
IP IP

ISDN Access Signaling IP

NETWORK

2.6 V5UA (V5.2-User Adaptation Layer Protocol)

V5UA protocol is used to deliver V5.2 messages over IP using the Stream Control
Transmission Protocol (SCTP).

Access Network Access Media Gateway (AGW) MGC

Nodal Interworking Function V5.2

V 5.2
V5UA V5UA
LAPV5
LAPV5 SCTP SCTP
IP IP
V 5.2 IP
NETWORK

3.0 H.248/Megaco: Media Gateway Control Protocol

H.248/Megaco is a signalling protocol between Media Gateway and Media Gateway

Controller (also known as Call Agent or Softswitch). This protocol is the result of joint work of
IETF and ITU. H.248 is the name given to it by the ITU and Megaco by IETF.

Switching Division, TEC Page 6 of 15

 Technology White Paper on NGN Protocols
H.248/MEGACO is designed to provide a centralised architecture where a centralised device
Media Gateway Controller, handles switching logic and call control functions. The Media
Gateway performs conversion of media (voice/video information) format as required from
TDM to IP networks and vice versa, and transmits them. This architecture closely resembles
the existing PSTN architecture and services.
H.248 enables the creation, modification and deletion of media streams across a media
gateway, including the capability to negotiate the media formats to be used.

Media Gateway Controller /Call

Agent / Softswitch

Megaco/ Megaco/
H.248 Megaco/ Megaco/ H.248
H.248 H.248
Trunk Media Trunk Media
Gateway (TMG) Gateway (TMG)
PSTN/PLMN Packetised PSTN/PLMN
Media stream Media stream Media stream
(RTP)

Access Media Access Media

Gateway Gateway
PSTN/PLMN Packetised PSTN/PLMN
(AGW) Media stream (AGW)
Media stream Media stream
(RTP)

Figure 4: H.248/Megaco Architecture

3.1 How H.248 works?

When an Access Media Gateway (AGW) detects an off hook condition, it informs the MGC
that a call has arrived.
MGC responds with a command to instruct the AGW to connect dial tone on the line and
receive DTMF tones indicating the number dialled.
After receiving the dialed digits, AGW sends the digits to MGC which analyses the digits to
determine how to route the call.
For a terminating call to the same network, MGC instructs the appropriate AGW to connect
the called number.
AGW connects the called number and sends the status of called line to the MGC.
If the called line is off hook, MGC instructs the AGW(s) to establish a two-way communication
channel.
AGW converts the format of the media streams coming from PSTN/PLMN and connect them
to the appropriate port using RTP stream, as instructed by MGC,
There are two basic component concepts in H.248/Megaco namely terminations and
contexts.

Switching Division, TEC Page 7 of 15

 Technology White Paper on NGN Protocols
Terminations: Terminations represent media streams entering or leaving the media gateway.
These are used to represent flows on the packet network, such as an RTP stream.
A termination is given a name or termination ID, by the media gateway. Some terminations,
which typically represent ports on the gateway, such as time slot in an E1, remain active all
the time. Other terminations such as RTP flows, called ephemeral, are created when needed
and released after use.
Context: Context is an association between a numbers of terminations for the purpose of
sharing media between those terminations. Terminations can be added to contexts, removed
from contexts, or moved from one context to another. A termination can exist in only one
context at any time, and terminations in a given gateway can only exchange media if they are
in the same context. The normal, "active" context might have a physical termination (one time
slot in an E1) and one RTP stream connecting the gateway to the IP network Contexts are
created and released by the gateway through the command from MGC.
A context ID, assigned by the MG, identifies a context and is unique within a single MG. The
context itself has certain attributes, including the topology, which indicates the flow of media
between terminations.
H.248/Megaco uses a series of commands to manipulate terminations and contexts as below:
• Add - adds a termination to a context and may be used to create a new context at the
same time.
• Subtract - removes a termination from a context and may result in the context being
released if no terminations remain.
• Move - moves a termination from one context to another
• Modify - changes the state of the termination.
• Notify - The Notify command allows the Media Gateway to inform the Media Gateway
Controller of the occurrence of events in the Media Gateway
• Audit Value and Audit Capabilities - return information about the terminations,
contexts, and general gateway state and capabilities.
• Service Change - creates a control association between a gateway and a gateway
controller and also deals with some fail over situations.

Descriptors: Descriptors form the parameters of the command and/or response and provide
additional information to qualify a given command or response.
Packages: A gateway may implement terminations that have different characteristics.
Variations in terminations are accommodated in the protocol by allowing terminations to have
optional properties, events, signals and statistics implemented by media gateway. Such
options are grouped into Packages, and typically a termination realizes a set of such
Packages. Examples of packages are; Tone Detection Package, DTMF Generator Package
etc. MGC can audit a termination to determine which packages it supports.
Transactions: H.248 protocol involves a series of transactions between MGC’s and MG’s.
Each transaction involves sending a ‘Transaction Request’ by the initiator of the transaction
and sending of ‘Transaction Reply’ by the responder. A Transaction Request consists of a
number of commands and Transaction Reply consists of a corresponding number of

Switching Division, TEC Page 8 of 15

 Technology White Paper on NGN Protocols
responses. There may be multiple commands within a single transaction and multiple
transactions within a Message.
H.248 protocol uses transactions, commands, terminations, contexts, descriptors and
packages to create, modify and release a media session call.

4.0 RTP: Real-Time Transport Protocol

The Real-time Transport Protocol (RTP) is an internet protocol which provides end-to-end
HT TH HT TH

delivery of real-time data such as audio, video and text over IP. RTP itself does not guarantee
HT TH

real-time delivery of data, but it does provide mechanisms for the sending and receiving
applications to support streaming data. Typically, RTP runs on top of the UDP protocol,
HT TH HT TH

although the specification is general enough to support other transport protocols. RTP is
defined in IETF RFC 3550 and 3551.
RTP packets include a sequence number, so that the application using RTP can detect the
occurrence of lost packets and present the received packets to the user in the correct order.
RTP packets also include a time-stamp that corresponds to the time at which the packet was
sampled from its source media stream. The destination application can use this time-stamp to
synchronise multiple streams with each other and to reduce delay and jitter. RTP may also
run over another suitable transport protocol like TCP.
4.1 How RTP Works?
As shown in figure 5, the multimedia applications which consist of multiple audio, video, text
etc. are sent to RTP library which multiplexes the streams and encodes them into packets.

Application
Media
Encapsulation

RTP RTCP

Data Control

UDP

IP

Ethernet

Figure 5: RTP Architecture

Switching Division, TEC Page 9 of 15

 Technology White Paper on NGN Protocols
A header is attached to these RTP packets. These RTP packets are sent to UDP, where a
UDP header is attached. The combined packet is the sent to IP, where an IP header is
attached and the resulting IP datagram is routed to the destination. At the destination, the
various headers are used to pass the packet to the appropriate application.
RTP can use many different voice and video coding standards (e.g. G.711 G.723, G.729,
H261 etc.). RTP includes a mechanism which allocates a payload type number to various
coding schemes and provides high-level descriptions of the coding techniques. Thus pay load
data can be correctly interpreted at receiving end by knowing which coding scheme is being
used.

4.2 RTCP: Real Time Transport Control Protocol (or RTP Control Protocol)

RTCP is the control protocol for RTP (Real-time Transport Protocol). RTCP's primary function
is to provide feedback on delay, jitter, bandwidth, congestion and other network properties.
This information is used to improve quality of service. RTCP also handles interstream
synchronization.

4.3 User Datagram Protocol (UDP)

UDP is a transport protocol, provides connectionless transport service with single-shot

datagram type service.
It receives the packets from the application and sends to IP to route to the far end. At the
receiving end, it passes the incoming data packets from IP to the appropriate application.
UDP does not provide acknowledgement, sequencing, flow control, message continuation, or
other sophisticated attributes. Therefore there is no guarantee that anything sent from an
application that uses UDP will actually get to the desired destination.
It is simple, efficient and ideal for applications that require a quick, one-shot transmission of a
piece of data or a simple request/response. That is why it is preferred for real time
applications like voice over IP (VoIP).

5.0 Session Initiation Protocol (SIP)

SIP is an ASCII-based application-layer control (signaling) protocol for creating, modifying,

and terminating sessions with one or more participants. These sessions include Internet
telephone calls, multimedia distribution and multimedia conferences. SIP is defined in IETF
RFC 3261.
SIP provides the capabilities to:
• Determine the location of the target end point.
• Determine the media capabilities of the target end point via Session Description Protocol
(SDP).
• Determine the availability of the target end point. If a call cannot be completed because
the target end point is unavailable, it returns a message indicating why the target end
point was unavailable.
• Establish a session between the originating and target end point if the call can be
completed.

Switching Division, TEC Page 10 of 15

 Technology White Paper on NGN Protocols
• Handle the transfer and termination of calls. SIP supports the transfer of calls from one
end point to another.

5.1 Components of SIP

SIP is a peer-to-peer protocol. The peers in a session are called User Agents (UAs). A user
agent can function in one of the following roles:
• User agent client (UAC): A client application that initiates the SIP request.
• User agent server (UAS): A server application that accepts a SIP request and returns a
response to the request.
Typically, a SIP end point is capable of functioning as both a UAC and a UAS.
From an architecture standpoint, the physical components of a SIP network can be grouped
into two categories: clients and servers. Figure 6 illustrates the architecture of a SIP network.

Location Server
Registrar Registrar
Server 1 Server 2

Proxy/Redirect Server 1 Proxy/Redirect Server 2

User SIP Signalling User

Agent Agent

RTP Media Stream

PSTN Gateway
Router

PSTN/ PLMN Policy Internet

Server

Figure 6: Architecture of SIP Network

5.2 SIP Clients

SIP Phones act both as UAS and UAC. Softphones (PCs that have phone capabilities
installed) and SIP phones can initiate SIP requests and respond to requests.
Gateways provide call control. They translate between audio and video codecs and performs
call setup and clearing on both sides of network.
5.3 SIP Servers
SIP servers include:
• Proxy server: The proxy server receives SIP requests from a client and then forwards
the requests on the client's behalf.
Switching Division, TEC Page 11 of 15
 Technology White Paper on NGN Protocols
• Redirect server: Redirect server provides the client with information about the address
where the request has to be sent. The client then sends the request to the address given
by the redirect server.
• Registrar server: Registrar server processes requests from UACs for registration of their
current location.
• Location Server: A Location Server is used by a SIP redirect or proxy server to obtain
information about a called party's possible location.
• Policy Server: The Policy Server is designed to use Common Open Policy Service to
provide Quality of Service (QoS), bandwidth reservation for calls or call segments that
are transmitted over the network. The Policy Server uses open interfaces to interface
with clearinghouses for reserving bandwidth and authorising the use of a network for
inter network calls.

5.4 SIP Message Syntax

SIP Message syntax is text-based. These messages are either requests from a client to a
server or responses from a server to a client.

5.5 SIP-T and SIP-I

SIP-T (SIP for telephones) is a protocol defined by IETF that allows SIP to be used for ISUP
call setup between SS7-based public switched telephone networks and SIP-based IP
telephony networks. SIP-T carries an ISUP message payload in the body of a SIP message.
The SIP header carries translated ISUP routing information. SIP-T also specifies the use of
the SIP INFO method for effecting IN-call ISUP signalling in IP networks.
SIP-I (SIP ISUP mapping) is a protocol defined ITU (Q.1912.5) which specifies
recommendations for interworking of ISUP/BICC and SIP. It is more accurate and explicitly
defines the parameters between PSTN and SIP. It also defines the supplementary services
for telecommunication interconnection. SIP-I is widely accepted by manufacturers, carriers
and organizations (e.g. 3GPP).

6.0 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.
H.323 based network has following elements: (Refer Figure 6)

• Gateway: It connects IP network to PSTN/PLMN and provides translation of both call

control signalling and media from the PSTN/PLMN to IP network and vice versa.
• Gatekeeper: It controls the IP terminals under its jurisdiction and governs access to the
Gateway.
• H.323 terminal: H.323 terminal is typically an IP phone or any IP device such as a PC.
H.323 is not a single protocol; it refers to a suite of protocols required for network integration.
The following protocols are defined:
Switching Division, TEC Page 12 of 15
 Technology White Paper on NGN Protocols

• H.225 RAS: Registration, Authentication and Status protocol. It manages a channel

between the Gatekeeper and an H.323 endpoint to provide secure access control.

Gateway

H.323

Gatekeeper

H.323

IP PSTN/
Network PLMN

H.323 IP Device
PSTN/PLMN
Subscribers

Figure 6: H.323 based network

• H.225 Call Signalling: It is used for establishing connections between H.323 endpoints.
• H.245 Control Signalling: It runs between H.323 endpoints, allowing exchange of control
messages.
• RTP: The Real Time Protocol, which carries packetised media between H.323 endpoints.
• RTCP: The RTP control protocol (RTCP), to monitor the quality of service and to convey
information about the participants in an on-going session.

Switching Division, TEC Page 13 of 15

 Technology White Paper on NGN Protocols
ANNEXURE-1

NGN Protocols

S. Protocol ITU-T/IETF Title Remarks

No. RFC No.
1. H.248 ITU-T Media Gateway Control Media Gateway
(Megaco) Recommandation Protocol Control
H.248 Megaco Protocol Version Protocols
IETF RFC 3015 1.0
3. SIP-I ITU-T Inter working between
Recommendation Session Initiation Protocol
n Q.1912.5 (SIP) and Bearer Communication
Independent Call Control between two
Protocol or ISDN User Soft switches
Part.
4. SIP-T IETF RFC 3372 Session Initiation Protocol
for Telephones (SIP-T):
Context and Architecture
6. RTP,RTCP IETF RFC 3550, Real Time Protocol, Real Delivery of
RFC3551 Time Control Protocol Packetised
Media Streams
over IP
7. Sigtran IETF RFC 2719 Framework Architecture
for Signalling Transport
8. SCTP IETF RFC 2960. Stream Control Transport
Protocol
9. M2PA IETF RFC 4165 MTP2 Peer to Peer
Adaptation protocol
10. M2UA IETF RFC 3331. MTP2 User Adaptation
Layer
protocol Signalling
11. M3UA IETF RFC 4666. MTP3 User Adaptation Interface with
Layer SS7 Network
protocol
12. SUA IETF RFC 3868 SCCP user Adaptation
protocol
13. IUA IETF RFC 4233 ISDN User Adaptation
Layer Protocol

14. V5UA IETF RFC 3807 V5.2 User Adaptation

Layer Protocol

Switching Division, TEC Page 14 of 15

 Technology White Paper on NGN Protocols

ABBREVIATIONS

AGW Access media Gateway

IETF Internet Engineering Task Force
IP Internet Protocol
ISDN Integrated Services Digital Network
ITU International Telecommunication Union
ITU-T ITU Telecommunication Sector
IUA ISDN User Adaptation Layer
M2PA MTP2- User Peer-to-Peer Adaptation Layer
M2UA Message Transfer Part User Adaptation Layer 2
M3UA Message Transfer Part User Adaptation Layer 3
MGW Media Gateway
MTU Maximum Transmission Unit
NGN Next Generation Network
PSTN Public Switched Telephone Network
RFC Request For Comments
RTCP Real Time Control Protocol
RTP Real Time Protocol
SCCP Signalling Connection Control Part
SCN Switched Circuit Network
SCTP Simple Control Transmission Protocol
SCTP Stream Control Transport Protocol
SDP Session Description Protocol
SIP Session Initiation Protocol
SS7 Common Channel Signalling No.7
SUA SCCP User Adaptation
TDM Time Division Multiplex
TEC Telecommunication Engineering Centre
TMG Trunk Media Gateway
UDP User Datagram Protocol
V5UA V 5.2 User Adaptation Layer Protocol

End of document

Switching Division, TEC Page 15 of 15