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Ref.5 NGN Protocols

Ref.5 NGN Protocols


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Technology White Paper

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 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.


ISDN Q.931




User Adaoption












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)
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controls the delivery of packetised media streams over RTP.
MGC/Softswitch IN
Switch INAP



SIP-I/SIP-T, H.323

SS7 TDM Local/ Transit Switch




H.248 SIP

H.248 SIP



TDM Local/ Transit Switch

IP Analog/ ISDN

IP H.248


H.248 AGW

Analog/ ISDN

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 Model

Common Signalling Transport Protocol

Standard Internet Protocol (IP)

Figure 3: SIGTRAN Protocol Stack Model

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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 multistreaming 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 orderof-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.
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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 Signalling Gateway Media Gateway Controller





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

Nodal Interworking Function

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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


Nodal Interworking Function






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


Nodal Interworking Function






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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) Nodal Interworking Function ( NIF ) Media Gateway Controller



ISDN Access Signaling




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 V 5.2 Access Media Gateway (AGW)
Nodal Interworking Function




V 5.2

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.
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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/ H.248 Trunk Media Gateway (TMG)

Megaco/ H.248

Megaco/ H.248

Megaco/ H.248 Trunk Media Gateway (TMG)

PSTN/PLMN Media stream

Packetised Media stream (RTP)

PSTN/PLMN Media stream

PSTN/PLMN Media stream

Access Media Gateway (AGW)

Packetised Media stream (RTP)

Access Media Gateway (AGW)

PSTN/PLMN Media stream

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




Figure 5: RTP Architecture

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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

• 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

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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 Server 1 Registrar Server 2

Proxy/Redirect Server 1 User Agent SIP Signalling

Proxy/Redirect Server 2 User Agent

RTP Media Stream
PSTN Gateway Router


Policy 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.
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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:
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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

IP Network


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.

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Technology White Paper on NGN Protocols

ANNEXURE-1 NGN Protocols S. No. 1. Protocol H.248 (Megaco) ITU-T/IETF RFC No. ITU-T Recommandation H.248 IETF RFC 3015 ITU-T Recommendation n Q.1912.5 Title Media Gateway Control Protocol Megaco Protocol Version 1.0 Inter working between Session Initiation Protocol (SIP) and Bearer Independent Call Control Protocol or ISDN User Part. Session Initiation Protocol for Telephones (SIP-T): Context and Architecture Real Time Protocol, Real Time Control Protocol Remarks Media Gateway Control Protocols



Communication between two Soft switches






IETF RFC 3550, RFC3551

Delivery of Packetised Media Streams over IP

7. 8. 9. 10.

Sigtran SCTP M2PA M2UA

IETF RFC 2719 IETF RFC 2960. IETF RFC 4165 IETF RFC 3331.



IETF RFC 4666.

12. 13.



Framework Architecture for Signalling Transport Stream Control Transport Protocol MTP2 Peer to Peer Adaptation protocol MTP2 User Adaptation Layer protocol MTP3 User Adaptation Layer protocol SCCP user Adaptation protocol ISDN User Adaptation Layer Protocol

Signalling Interface with SS7 Network



IETF RFC 3807 V5.2 User Adaptation Layer Protocol

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Technology White Paper on NGN Protocols

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

End of document

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