10
 Towards Megaco Architecture J. Peltola, J. Aromaa, A. Mustonen Satakunta Polytechnic  November 28th 2002  Abstract  —The Voice over Internet Protocol (VoIP) technology is designed to transfer voice, video and data in packet based networks. As it stands this technology is not suitable for traditional circuit switched networks. To combine these two network techniques, the packet network and the circuit switched network, a new network architecture is needed. The aim of this paper was to examine the development of combining the VoIP technology and traditional phone networks. Special attention was given to protocols associated with this convergence. In this paper ITU-T H.323 and IETF SIP (Session Initiation Protocol) VoIP protocols are studied. These protocols were the base when studying call control protocol called Megaco/H.248 in next generation networks. Megaco/H.248 is a media gateway control protocol and it is a collaborative effort of the IETF and ITU-T. Megaco specifies the master/slave architecture for decomposed gateways. Media Gateway Controller (MGC) is the master server which is responsible for call control functions and one or more Media Gateways (MG) are the slave clients which are responsible for media mixing. This paper describes how Megaco solutions will take place at Satakunta Polytechnic. The operation of the Megaco/H.248 protocol is described with examples.  Index Terms  —Voice over IP, Megaco, Network Convergence I. INTRODUCTION V OICE traffic is more and more general payload type in packet networks. There is a demand for combining the modern public switched telephone network (PSTN) and VoIP-technologies in the packet networks. As it stands VoIP solutions like SIP and H.323 do not work in the PSTN. To combine these two network architectures a new kind of architecture is defined. ITU-T and IETF created a protocol for controlling Media Gateways. This new protocol creates a new architecture called Megaco architecture which makes it possible to use services in two different networks. Centralized control enables an easy way to  provide customers with new s ervices. The idea of a decomposed gateway architecture is  based on the ETSI TIPHON project (Telecommunication and Internet Protocol Harmonization Over Networks) [9]. TIPHON defines five scenarios of how to combine two networks and how to make calls between these two networks. TIPHON project defines Interworking Functions (IWF) which include Signaling Gateway, Media Gateway and Media Gateway Controller. The Megaco protocol controls Media Gateway. So this Megaco/H.248 [4] protocol works between the Media Gateway Controller (MGC) and the Media Gateway (MG). This kind of network is easy to expand  because all the intelligence is stored in the MGC. Signaling from and to the PSTN is transported through the Signaling Gateway (SG). The protocol used to transport signaling information in the packet network is Signaling Control Transport Protocol (SCTP) [12].  This paper describes a decomposed gateway architecture and how to combine VoIP signaling  protocols like SIP and H.323 to this new network. The  paper is organized as follows: Section 2 gives an overview of VoIP protocols, like H.323 and SIP; Section 3 describes the Megaco Architecture and Megaco protocol; Section 4 presents how to combine the PSTN and packet networks and how these networks work together; Section 5 describes the Megaco solution at Satakunta Polytechnic; Section 6 describes future  plans and Section 7 is the conclusion part. II. VOIP STANDARDS The VoIP technology started to develop more rapidly after 1995. That year VocalTec introduced Internet telephony software [1]. This PC based program opened connection between two PC endpoints across the Internet. The ITU-T study group 16 prepared a VoIP standard called H.323 [2]. This standard completed in 1996. The IETF VoIP standard called Session Initiation Protocol (SIP) was ready to be published in 1999. SIP is a text based protocol and it was first defined in the RFC 2543 [3]. In the year 2000 ITU-T and IETF collaboratively published the Megaco protocol for controlling Media Gateways.  A.  H.323 The H.323 protocol was designed to support multimedia services over a LAN (Local Area Network). H.323 is an umbrella standard and it includes H.245 for control operations, H.225 for connection management and T.120 for document support for conferences. H.323 uses a binary syntax for its messages as several other ITU-T standards do. The H.323 architecture defines four major components: Terminals, Gateways, Multi-point Control 43

Towards Megaco Architecture

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Towards Megaco Architecture

J. Peltola, J. Aromaa, A. Mustonen

Satakunta Polytechnic 

 November 28th 2002

 Abstract  —The Voice over Internet Protocol (VoIP)

technology is designed to transfer voice, video and data in

packet based networks. As it stands this technology is not

suitable for traditional circuit switched networks. To

combine these two network techniques, the packet

network and the circuit switched network, a new network

architecture is needed. The aim of this paper was to

examine the development of combining the VoIP

technology and traditional phone networks. Special

attention was given to protocols associated with thisconvergence. In this paper ITU-T H.323 and IETF SIP

(Session Initiation Protocol) VoIP protocols are studied.

These protocols were the base when studying call control

protocol called Megaco/H.248 in next generation networks.

Megaco/H.248 is a media gateway control protocol and it

is a collaborative effort of the IETF and ITU-T. Megaco

specifies the master/slave architecture for decomposed

gateways. Media Gateway Controller (MGC) is the master

server which is responsible for call control functions and

one or more Media Gateways (MG) are the slave clients

which are responsible for media mixing. This paper

describes how Megaco solutions will take place at

Satakunta Polytechnic. The operation of the

Megaco/H.248 protocol is described with examples.

 Index Terms —Voice over IP, Megaco, Network

Convergence

I.  INTRODUCTION

V OICE traffic is more and more general payload type

in packet networks. There is a demand for

combining the modern public switched telephone

network (PSTN) and VoIP-technologies in the packet

networks. As it stands VoIP solutions like SIP and

H.323 do not work in the PSTN. To combine these twonetwork architectures a new kind of architecture is

defined. ITU-T and IETF created a protocol for

controlling Media Gateways. This new protocol creates

a new architecture called Megaco architecture which

makes it possible to use services in two different

networks. Centralized control enables an easy way to

 provide customers with new services.

The idea of a decomposed gateway architecture is

 based on the ETSI TIPHON project

(Telecommunication and Internet Protocol

Harmonization Over Networks) [9].  TIPHON defines

five scenarios of how to combine two networks andhow to make calls between these two networks.

TIPHON project defines Interworking Functions (IWF)

which include Signaling Gateway, Media Gateway and

Media Gateway Controller.

The Megaco protocol controls Media Gateway. So

this Megaco/H.248 [4]  protocol works between the

Media Gateway Controller (MGC) and the Media

Gateway (MG). This kind of network is easy to expand

 because all the intelligence is stored in the MGC.

Signaling from and to the PSTN is transported through

the Signaling Gateway (SG). The protocol used to

transport signaling information in the packet network is

Signaling Control Transport Protocol (SCTP) [12]. 

This paper describes a decomposed gateway

architecture and how to combine VoIP signaling

 protocols like SIP and H.323 to this new network. The

 paper is organized as follows: Section 2 gives an

overview of VoIP protocols, like H.323 and SIP;

Section 3 describes the Megaco Architecture and

Megaco protocol; Section 4 presents how to combine

the PSTN and packet networks and how these networks

work together; Section 5 describes the Megaco solution

at Satakunta Polytechnic; Section 6 describes future

 plans and Section 7 is the conclusion part.

II.  VOIP STANDARDS 

The VoIP technology started to develop more rapidly

after 1995. That year VocalTec introduced Internet

telephony software [1]. This PC based program opened

connection between two PC endpoints across the

Internet. The ITU-T study group 16 prepared a VoIP

standard called H.323 [2].  This standard completed in

1996. The IETF VoIP standard called Session Initiation

Protocol (SIP) was ready to be published in 1999. SIP is

a text based protocol and it was first defined in the RFC2543 [3].  In the year 2000 ITU-T and IETF

collaboratively published the Megaco protocol for

controlling Media Gateways.

 A.   H.323

The H.323 protocol was designed to support

multimedia services over a LAN (Local Area Network).

H.323 is an umbrella standard and it includes H.245 for

control operations, H.225 for connection management

and T.120 for document support for conferences. H.323

uses a binary syntax for its messages as several other

ITU-T standards do.

The H.323 architecture defines four majorcomponents: Terminals, Gateways, Multi-point Control

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Units (MCU) and Gatekeepers. The H.323 terminal

 provides real-time, two-way audio, video and data

communications with another H.323 terminal. Examples

of the H.323 terminals are the multimedia PC or the IP

 phone. H.323 requires that terminals must have voice

functionality and ability to communicate using the

H.225 and H.245 protocols. The Gateway provides the

appropriate translation between transmission formats(for example H.225.0 to/from H.221) and between

communications procedures. The Gateway also

 performs call setup and clearing on both the packet

network side and the PSTN side. This makes the H.323

Gateway a very important component when connecting

traffic between the PSTN and the packet network. The

Gatekeeper, which is optional in an H.323 system,

 provides call control services to the H.323 endpoints.

There is only one Gatekeeper in one Zone, and this

Gatekeeper performs all call control functions inside

this Zone. The Gatekeeper may also do for example

address translations and bandwidth management. The

MCU supports multipoint conferencing between three

or more terminals and Gateways. The MCU consists of

a multipoint controller (MC) and a multipoint processor

(MP). The MC carries out the capabilities exchange

with each endpoint in a multipoint conference. The MP

 processes these media streams and returns them to the

endpoints. Major H.323 components and examples of

connections to other networks are described in Fig. 1.

[5]

There may be two types of network servers in the SIP

environment: a proxy server and a redirect server. The

SIP proxy server acts as both a server and a client for

making requests on behalf of other clients. A SIP

redirect server is a user agent server that dictates the

client to contact an alternate address. Another

component in SIP is a registration server called

Registrar. The Registrar is a server that acceptsregistration requests. The Registrar is typically co-

located with a proxy or redirect server. The Registrar

may offer location services on the base of register

messages. The SIP components are presented in Fig. 2.

[6] The SIP and H.323 standards are used for signaling

in the VoIP networks. These protocols make it possible

to create, modify and terminate multimedia calls. Both

of these VoIP network architectures contain intelligent

components. Thus the architectures are based on

distributed intelligence. This kind of architecture is

difficult to maintain but flexible when adding

intelligence to endpoints or gateways. Developers on

the PSTN side think that this kind of architecture which

is based on distributed intelligence is complex. [7]

 B.  SIP

The Session Initiation Protocol (SIP) is anapplication-layer control protocol that can establish,

modify, and terminate multimedia sessions

(conferences) such as Internet telephony calls. SIP can

also invite participants to already existing sessions, such

as multicast conferences. SIP uses text-encoded

messages for request and responses like HTTP. SIP

consists of two major components, which are the user

agent and the network server. The user agent (UA) is a

logical entity that can act as both a user agent client

(UAC) and a user agent server (UAS). The UA

interfaces with the user and acts on behalf of the user.

The UAC initiates the call and the UAS is used to

answer the call.

IPIP

IPIP

SIP User Agent

SIP Proxy Server

and Registrar 

SIP Redirect Server 

IPIP

SIP User Agent

SIP ProxyServer 

IPIP

IPIP

SIP User Agent

SIP Proxy Server

and Registrar 

SIP Redirect Server 

IPIP

SIP User Agent

SIP ProxyServer 

 Fig. 2: Major SIP components 

PSTNPSTNIPIP

H.323 Gateway

H.323 Terminals

H.323 Gatekeeper 

MCU

H.323 Gateway

ATMATM

Fig. 1: An example of the H.323 network  

III.  MEDIA GATEWAY CONTROL 

The Media Gateway Control concept means ways to

control the device which manipulates and terminates the

media streams. These devices are called Media

Gateways (MG). This concept also means that there is

one component which includes the network intelligence.

The control functions and intelligence are made by theMedia Gateway Controller (MGC). The protocol which

is used for controlling MGs is called Megaco/H.248. [8]

The Megaco/H.248 standard is the result of the

cooperation between IETF and ITU-T. Lots of other

organizations helped gain this coal. The ETSI TIPHON

 project and the Multiservice Switching Forum (MSF)

were significant when this protocol was being

developed. Previous versions of Media Gateway

Control Protocols have been introduced since 1998.

Simple Gateway Control Protocol (SGCP) was the first

 protocol for controlling Gateways. Also Internet

Protocol Device Control (IPDC) was announced in1998. These two protocols combined and the resulting

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 protocol was named Media Gateway Control Protocol

(MGCP). ITU-T made Media Device Control Protocol

(MDCP) as a competing proposal for the MGCP. ITU

renamed the MDCP to H.GCP. Finally a new kind of

connection model was developed and the

Megaco/H.248 development began. Megaco/H.248 was

approved by IESG in May 2000 and by the ITU-T in

June 2000. [8]

The concentration of Interworking functionality

consists of three decomposed devices. This is because

one device with all functions needed for Interworking is

not scalable or efficient enough. The three components

are: [9]

-  Media Gateway (MG) handling the conversion and

mixing of the media streams

-  Media Gateway Controller (MGC) managing and

controlling the connections and providing call

 processing and

-  Signaling Gateway (SG) providing the signaling

mediation function.

These decomposed components are shown in Fig. 3.

The decomposition offers advantages such as

efficiency, robustness, flexibility and scalability. The

intelligence for call control logic and network signaling

is moved to more generic computing resources (MGC).

The MGC allows the network operator tightly to control

and manage the communications and to provide new

services for customers without making any changes to

MGs. [10]

When different functionalities take place in different

components a special protocol between the components

is needed. The Megaco protocol addresses the MGC↔ 

MG communication. Signaling information from and to

the PSTN is carried out using the SCTP [12] as a

signaling transport protocol. Section 4 describes more

accurately the communication between the MGC and

the MG. Fig. 4 shows the VoIP network, the PSTN and

the decomposed Gateways which mix media streams

 between these two networks. The call control protocol

Megaco is also mentioned. [10]

IV.  MEGACO/H.248 PROTOCOL 

The Megaco/H.248 protocol is a master/slave

 protocol. In this architecture the MGC is a master and

one or more MGs are slaves. The Megaco protocol

 provides means for describing and controlling the

connections of media streams. The Megaco/H.248

 protocol is based on a connection model. The

connection model includes two concepts: Contexts and

Terminations. The connection model can be

manipulated via commands and extended with specific

 packages. [4], [10]

PSTNPSTN IPIPVoIP

Terminal

P   C   M   

Telephone

Exchange

Subscriber MG

SGSignaling (ISUP)

  S  C  T   P

  M e g  a c

 o  /  H.  2 4  8

MGC

MGRTP

H   .3   2   3    o  r  S   I   P   

    R     T    P

M    e    g   a  c   o   /    H    

. 2   4    8   

Fig.4: Gateway distribution and control 

Termination

RTP Stream

Termination

SCN Bearer

Channel

Context

Context

Context

Termination

RTP Stream

Termination

RTP Stream

Termination

SCN Bearer

Channel

Termination

SCN Bearer

Channel

Termination

SCN Bearer

Channel

 Null Context

*

*

*

Media Gateway

Termination

RTP Stream

Termination

SCN Bearer

Channel

Context

Context

Context

Termination

RTP Stream

Termination

RTP Stream

Termination

SCN Bearer

Channel

Termination

SCN Bearer

Channel

Termination

SCN Bearer

Channel

 Null Context

*

*

*

Media Gateway

Termination

RTP Stream

Termination

SCN Bearer

Channel

Context

Context

Context

Termination

RTP Stream

Termination

RTP Stream

Termination

SCN Bearer

Channel

Termination

SCN Bearer

Channel

Termination

SCN Bearer

Channel

 NullContext

*

*

*

Media Gateway

Fig.5: Example of Connection Model described in Megaco/H.248

standard 

SG

MGC

MG

Media ControlSignal ing Control

 Fig. 3: Physical decomposition of MG, MGC and SG 

 A.  The Connection Model

The connection model for the Megaco protocoldescribes the logical entities within the MG that can becontrolled by the MGC. The main abstractions used inthe connection model are Terminations and Contexts.[4]

A Termination sources and/or sinks one or morestreams. In a multimedia conference, a Termination can be multimedia and sources or sinks multiple mediastreams. [4]

A Context is an association between a collection ofTerminations. It describes the topology (who hears/seeswhom) of associated Terminations and the mediamixing if more than one Terminations are involved inthe association. There is a special type of Context, thenull Context, which contains all Terminations that arenot associated to any other Termination (e.g. idle linesin the access gateway). These concepts are shown inFig. 5. [4]

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Terminations can be permanent or ephemeral.

Permanent Terminations represent physical entities

which have a semi-permanent existence for example

Termination representing a TDM (Time Division

Multiplexing) channel. The Terminations representing

ephemeral information flows would usually exist only

for the duration of their use (e.g. RTP flows).

Ephemeral Terminations are created by the Addcommand and destroyed by the Subtract command. A

 physical Termination is added to or subtracted from a

Context; it is taken from or to the Null Context. [4]

Termination is identified by a unique ID called

TerminationID. The Termination is described by

Properties, Signals, Events and Statistics. Terminations

may have signals applied to them. Signals are MG

generated media streams such as tones and

announcements as well as line signals such as

hookswitch. Properties can be common or specific to

media streams. Properties are grouped into a set of

Descriptors that are included in commands.

Terminations may be programmed to detect Events

which can trigger notification messages to the MGC, or

action by the MG can also trigger notification to the

MGC. Statistics may be accumulated on a Termination

and reported to the MGC. [4], [10]

 B.  Command

The protocol provides commands for manipulating

Contexts and Terminations which are the logical entities

of the connection model. For example, with the Add

command it is possible to add Terminations to a

Context, the Modify command makes it possible to

modify Terminations, the Subtract command subtracts

Terminations from a Context, and the AuditValue and

the AuditCapabilities makes it possible to audit

 properties of Contexts or Terminations. The commands

 provide for complete control of the properties of

Contexts and Terminations. This includes specifying

which events a Termination is to report, which signals

are to be applied to a Termination and specifying the

topology of a Context. Most commands are for the

specific use of the MGC as a command initiator in

controlling MGs as command responders. The

exceptions are the Notify and ServiceChange

commands. Commands, their explanations and

directions are presented in Table I. [4] 

C.  Transactions

The Commands between the Media Gateway

Controller and the Media Gateway are grouped into

Transactions. The Transactions are identified by a

unique TransactionID. Transactions consist of one or

more Actions. An Action consists of a series of

Commands that are limited to operating within a single

Context. Actions are recognized by ContextID. The

relationship between Transactions, Actions and

Commands is shown in Fig. 6. [4]

Every transaction is initiated by a TransactionRequest

and must be closed by a TransactionReply. There is a

way to prevent the sender from assuming that the

TransactionRequest was lost and the Transaction will

take some time to complete. A TransactionPending

indicates that the Transaction is actively being

 processed, but has not been completed. [4]

TABLE I

MEGACO COMMANDS 

Command EXPLANATION  Direction

Add Command adds a

Termination to a Context.

The Add command on the

first Termination in a

Context is used to create a

Context.

MGC - MG

Modify Command modifies the

 properties, events and

signals of a Termination.

MGC - MG 

Subtract Command disconnects a

Termination from its

Context and returns

statistics on the

Termination's participationin the Context. The

Subtract command on the

last Termination in a

Context deletes the

Context.

MGC - MG

Move Command atomically

moves a Termination to

another Context. 

MGC - MG

AuditValue Command returns the

current state of properties,

events, signals and

statistics of Terminations.

MGC - MG

AuditCapabilities Command returns all the

 possible values for

Termination properties,

events and signals allowed by the MG.

MGC - MG

 Notify Command allows the MG

to inform the MGC of the

occurrence of events in the

MG.

MG - MGC

ServiceChange The ServiceChange

command allows the MG

to notify the MGC that a

Termination or group of

Terminations is about to be

taken out of service or has

 just been returned to

service.

MGC - MG

MG - MGC

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MEGACO/1 [123.123.123.4]:55555

Transaction = 50006 {

Context = 5000 {

Modify = A5555 {

Events = 1235 {al/on},

Signals { }; to turn off ringing

}

}

}

Fig. 7: Example of Message [4] 

 D.   Messages

Several Transactions can be grouped into one

Message. The Message has a header, which includes the

identity of the sender. Every Message contains a versionnumber which identifies the version of the protocol

used in this Message. The versions consist of one or two

digits, beginning with version number 1. Example of

Message is shown in Fig. 7. [4]

 E.   Packages

The mechanism for extending the Megaco base

 protocol has been done by means of Packages. The

reason why Packages are needed is that there are

different types of gateways with different types of

Terminations. To utilize these different properties

Packages define additional Properties, Events, Signals

and Statistics that may occur in Terminations. In

Megaco a given property, event, signal or statistic

should be defined in only one package. As a result theMegaco/H.248 packages are small and tightly focused

in content. It is possible to add play tones suitable for

different countries. [5]

 F.   Examples of Megaco Architecture

Information about call flows was gathered from the

IETF Megaco call flow draft which illustrates the usage

of the Megaco protocol [11]. 

Megaco/H.248 is an important protocol when

combining the PSTN and the IP networks. There are

several possibilities to combine these two networks. The

customer or subscriber in the IP network could use IP

 phones or PC based soft phones which use either SIP or

H.323 for call signaling. The PSTN subscriber may be a

subscriber in the traditional analogous phone exchange

or behind the PBX (Phone Branch Exchange). The

Following paragraphs will introduce some cases when

combining calls in the PSTN and the IP networks.Command1

ContextID1

ContextID3

ContextID2

Command1

TRANSACTIONx

Command1 Command3

Command2Command1

Command1ContextID1

ContextID3

ContextID2

Command1

TRANSACTIONx

Command1 Command3

Command2Command1

 Fig. 6: Megaco Transactions, Actions and Commands [4] 

1)   MGC, MG and analogous subscriber

The first example describes a situation where

analogous subscriber wants to call another analogoussubscriber. Both subscribers are directly connected to

the MG. The MG could be for example a Residential

Gateway controlled by the MGC with the Megaco

 protocol. The call is transferred through the IP network

so the RTP streams are used. The components and

 protocols used in this scenario are described in Fig 8.

MG1

MGC

MG2

Megaco/H.248Megaco/H.248

RTP

 Fig. 8: MG equipped with analogous interface

First the MG1 and the MG2 register with the MGC.

They use the ServiceChange command to do this. With

this command MGs also audit their capabilities to the

MGC. After this event the MGC is aware of analogous

interfaces and subscribers behind the interfaces. MGC

sends Reply to both ServiceChange commands. In this

example all commands are replied but this is not

mentioned in the text. The MGC sends Modify

commands which set both MGs to the listening mode.They are now able to detect events, like off hook, which

should be immediately reported to the MGC. All events

needed for call setup are described in Fig. 9.

The MG1 detects off hook and reports this to the

MGC with the Notify command. The MGC is now

aware of off hook and instructs the MG1 to play a dial

tone to this interface. The user dials the number and this

information is delivered to the MGC with the Notify

message. The MGC analyses this number and knows

that the called party is a user connected to the MG2.

The MGC uses the Add command to create RTP

Terminations to both MGs. Now there is a Context

which includes an RTP Termination and AnalogousTermination. The MGC modifies the RTP Termination

and gives necessary information to send an RTP stream

to another MG. Analogous Termination in the MG1

uses a ringing signal to inform the user that the called

 party has been informed about the incoming call. Also

the MG2 is sending information about the incoming call

to the user which is connected to the MG2.

The MG2 detects off hook and informs the MGC

about this event. The MGC knows that the call has been

answered. The MGC modifies the MGs to stop playing

ringing tones and to start transfer media in the form of

the RTP stream across the IP network.

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to Gatekeeper to establish the call. In these cases the

intelligence is situated in VoIP terminals and in the

MGC. The VoIP terminals and the MGC are able to

handle call signaling and call control functions. SIP

interaction with the MGC is presented in Fig.12. Fig. 13

 presents the interaction with the H.323 environment.

The first flow chart in Fig. 14 presents the call flow

when the MGC uses SIP to create a call between the

PSTN and the SIP terminal. The call is originated from

the PSTN and the MGC is informed about the call with

the IAM. The MGC adds Terminations to the MG: one

Termination to the PSTN side to listen to the PCM time

slot and another Termination to the IP network side to

handle the media transfer over the IP network. The

MGs have now a context which includes bothTerminations. The MGC knows that the called party is a

SIP terminal because it received the IAM including the

destination phone number. The MGC uses the SIP

Invite message to inform the called party about the

incoming call. The SIP terminal responses with the

Ringing message. The MGC modifies the RTP

Termination and tells the IP address of the SIP terminal.

The MGC sends an ACM message to the PSTN to

inform that the numbers were received correctly. The

SIP terminal informs with an OK message that the call

has been answered. The MGC reports this to the PSTN

with an ANM message. The MGC accepts the call andsends an ACK message to the SIP terminal. After this

the RTP stream between the SIP terminal and the MG

starts. The MG takes care of media transfer to the PSTN

side.

Releasing the call will take place when the SIP user

hangs up. The MGC is notified by a Bye message. The

MGC sends a REL message to the PSTN and subtracts

all Terminations in the MG. This is done by the

Subtract command. The MGC replies with OK to the

SIP terminal and the PSTN replies with a RLC (Release

Complete) message to the REL message sent by the

MGC. This call termination is also shown in Fig. 14.

Interaction with the H.323 environment is described

in Fig. 15. The MGC uses the H.323 Gatekeeper (GK)for number translations when the call is directed to the

H.323 terminal. The MGC uses RAS messages to

communicate with the GK and H.225 and H.245

 protocols to communicate with the H.323 terminal.

PSTN MG MGC  SIP User  IAM 

ACM - 

Add 

Reply 

Invite 

Ringing 

 Alerting  

 Answer  

200 OK  

ACK  

ANM 

Modify 

Reply 

200 OK  

 RTP   Stream  

Bye REL 

Subtrack  

Reply 

RLC 

 Hang up

 

Agent 

Fig. 14: Example of the interaction with the SIP terminal 

P   C   M   

Telephone

Exchange

Telephone

Exchange

MG

SGSS7

  M e g  a c o  /

  H.  2 4  8

MGC

RTP

H.323GK

H.323

Terminal

H

 . 3  2   3  

Fig. 13: Megaco and H.323 

P   C   M   

Telephone

Exchange

Telephone

Exchange

MG

SGSS7

  M e g  a c o  /  H.  2 4  8

MGC

RTP SIP User

Agent

S   I   P   

 Fig. 12: Megaco and SIP 

In this example a call is initiated from the PSTN. The

MGC receives the IAM and does necessary mapping of

the number to identify that the called number belongs to

the H.323 network. The MGC replies with the ACM to

the IAM. The MGC adds Terminations to the MG, one

for the PSTN side and one for the IP network’s side.

The MGC sends an ACM message to the PSTN as a

reply to the IAM. The MGC acts as a H.323 terminal

towards the H.323 network and generates the callsignaling towards the called party. The MGC initially

generates an admission request (ARQ) towards the

Gatekeeper. The Gatekeeper acknowledges the

admission request message by generating the Admission

confirmation (ACF) message assuming that the GK

used the directed-routed call model. The Gatekeeper

 provides the transport address information of the H.323

terminal. The MGC then initiates the H.225 signaling

 by generating the Setup message towards the H.323

terminal. The H.323 terminal initiates the ARQ towards

the Gatekeeper and receives the ACF as a reply. After

this the H.323 terminal generates the Alerting message

towards the MGC. The MGC, after having received the

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Alerting message from the H.323 Endpoint, generates

an ACM message to the PSTN.

These examples help understand the function of the

Megaco protocol in a discomposed architecture. It is

also important that the MGC can handle calls which use

the SIP or H.323 protocols. The MGC acts as a phone

exchange in the IP network but the switching part is

distributed to several MGs.

The H.323 terminal sends the Connect message when

the called party off hooks. The Terminal Capability set

and the master/slave determination occurs between the

MGC and the H.323 terminal. The Open Logical

channels messages indicate the session and media

related parameters. The MGC generates a Modifymessage towards the MG to modify the RTP

Termination with address information. After this the

RTP stream starts between the MG and the H.323

terminal.

V.  MEGACO SOLUTION AT SATAKUNTA POLYTECHNIC 

Satakunta Polytechnic has a very versatile

telecommunications laboratory. There are several

 possibilities to combine the PSTN and IP networks.

This convergence is done by using intelligent gateways.

There is a need for centralized intelligence which can

handle calls coming from SIP and H.323 networks and

going to the PSTN and vice versa.

The call termination takes place when the H.323

terminal goes on hook. The MGC initiates the tearing

down of the call by closing the logical channels that

were earlier created for exchanging the H.245

information (Close Logical Channels). After receiving

the Release Complete message the MGC generates a

REL message to the PSTN. The MGC generates a

Disengage Request (DRQ) towards the Gatekeeper. The

Gatekeeper acknowledges the Disengage Request

message by generating the Disengage Confirmation

(DCF) message. After this the MG subtracts the two

Terminations from the Context. The context itself is

deleted when the last Termination is subtracted.

 A.   Laboratory environment without the Megaco/H.248

call control

The telecommunications laboratory at SatakuntaPolytechnic is described in Fig. 16. There is a fixed

network telephone exchange (Nokia DX220) in the

laboratory. This represents the PSTN. This exchange is

connected to a public land mobile network (PLMN).

The PLMN includes a mobile switching centre (MSC,

 Nokia DX 200) with an integrated visitor location

register (VLR) and a home location register (HLR), a

 base station controller (BSC, Siemens) and two base

transceiver stations (BTS, Siemens).

PSTN MG MGC GK

IAM

ACM

Add

Reply

ANM

 

Reply

 RTP Stream

REL

Reply

RLC

H.323 Terminal

ARQ

ACF

Setup

ARQ

ACF

Connect

Terminal Capability Set

Master Slave Determination

Open Logical Channels

Modify

Alerting

Close Logical Channels

Release Complete

DRQ

DCF

DRQ

DCF

Subtract

Fig. 15: Example of the interaction with the H.323 environment 

The IP network contains components from several

manufacturers. The H.323 environment is based on

Cisco System Gatekeeper and Gateway. Microsoft NetMeeting is an H.323 terminal in the H.323

environment. Connection to the PSTN goes through

Cisco 3640 Gateway. The Gateway is equipped with an

E1 interface. This enables 30 simultaneous calls to the

PSTN. The SIP environment is based on the Columbia

University SIP server. The connection to the PSTN is

arranged through Cisco 2620 Gateway using the ISDN

2B+D connection. Cisco 7960 VoIP phones represent

the SIP terminals.

The Ericsson IP Telephony (IPT) system includes

Signaling Gateway, three Voice Gateways (VG) and

Sitekeeper. The SG provides signalling mediation between the IP network and the PSTN. The VG

 provides media mixing between the PCM signal and

 packet network. The Sitekeeper provides call control

functions to the IPT system. The Sitekeeper can also act

as a Gatekeeper for H.323 terminals. The IPT system

 provides trunk replacement e.g. it is possible to call

from the PSTN to the PLMN and the media is

transferred in the IP network.

As seen above, there are several different

architectures in the telecommunications laboratory at

Satakunta Polytechnic. To combine these architectures

one intelligent component is needed to handle the

calling and signaling information which comes from thePSTN, SIP or H.323 networks. The most suitable

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architecture for this purpose is Ericsson’s IPT which

has decomposed VGs for media mapping, the SG for

signaling and the Sitekeeper for control. In this

environment the controlling is done without any

standardized call control protocol. The goal is to update

this IPT environment to support the Megaco/H.248

 protocol which is standardized and well known.

 B.   Megaco/H.248 call control in the

telecommunications laboratory

Very important part when updating laboratory

environment towards the Megaco architecture is to

utilize present network components. The Ericsson IPT

system offers all necessary components to realize a

decomposed gateway architecture.

In the IPT system the VG performs media mapping.

Supported media type is audio but in the future also

video and data should be available. Ericsson offersTigris Media Gateway to do these functionalities. Tigris

MG performs audio, video and data mixing. With these

functionalities Tigris MG is suitable MG for laboratory

network. Old VGs have to be replaced because they do

not support video or data traffic.

Ericsson SG performs signalling mediation between

the IP network and the PSTN. It can transfer signaling

information from the PSTN to the MGC or vice versa.

This component is suitable for a Megaco based network

with software update.

The most important equipment in this new network is

the MGC. The MGC handles call control and address

translations. Also other intelligent operations such as billing, provisioning and call state handling are realized

in the MGC. The Sitekeeper can handle H.323

terminals, signaling information that is coming from the

PSTN and it can control VGs. Sitekeeper does not

support SIP or Megaco protocols. This is why software

updates are needed. With software updates Sitekeeper

can be transformed to the MGC. Software updates also

make it possible to use SIP and H.323 terminals for

communication with the MGC.

With the components described above and some

software updates the telecommunications laboratory at

Satakunta Polytechnic will support the Megaco protocol. The PSTN and IP network will work together

and users can use SIP and H.323 terminals.

Interworking with the PSTN and the IP network is

carried out with decomposed components: MG, SG, and

MGC. Important part of this architecture is the

Megaco/H.248 call control protocol. This new network

can be called the next generation network. The design

of this new network where the Megaco protocol is used

to control MGs, is described in Fig. 17.

PCM

ISDN PRI

PCM

ISDN BRI

L u ce n t L uc en tL  u c en t L u c en tL u ce n t L uc e nt L u c en t L uc e nt L u c en tL u ce n t  L u ce nt L u c e ntL uc e n t Lu c en t L  u ce n t L u c e nt L u c e n tL uc e nt L uc e n t L uc en t L u c en tL u ce n tLu c e nt L u ce nt

PACKET

CONT

NET

CONT

PROCR

INTFC

MEMO R Y TONEDET /GENTONECLOCK

POWER SUPPLY

EXP N

I NTFC

02 03

III II

04

III I I

05

IIII I

06

I IIII

07

II III

08

II I II

09

III I I

10

IIII I

11

I IIII

12

IIIII

13

I I III

14

III I I

16 1 7

PO WERU NI T

18

IIII III IIIC TONE

CLOCK

P ACKET

CONT

NET

CONT

TONEDET /GENTONECLO CK

02 0 3

II III

04

III II

05

III I I

06

IIII I

07

I IIII

08

III II

09

II I II

10

III I I

11

III I I

12

I IIII

13

IIIII

14

I I III

15

III I I

01

El etro s tati c

PSTN

PC with NetMeeting

PC with NetMeeting

PCML uc e n t L uc en t L u c en tL u ce n tL uc e nt L uc en tL  u c en t L u c e n tL uc e nt L uc en t L uc en t L u c en tL u ce n t L uc e nt L uc en t L u c en tL u ce n tL u c e n t L uc e n t L uc en t L u c en tL u c e n tLu c e nt L u c e n t

PA CKE TC ONT

NETCO NT

PROCRI NTFC

MEMORY TONEDET/G EN

TONECLOCK

POWERSUPPLY

EXPNI NTFC

02 0 3IIII I

04IIII I

05I I III

06III II

07IIII I

08IIII I

09I I III

10II III

11III II

12III I I

13IIII I

14II III

16 1 7

POW ERUNI T

18

III IIIII IIC TON ECLOCK

PACKETCO NT

NETCONT

TONEDET /GEN

TONECLOCK

02 0 3III II

04IIII I

05IIII I

06I I III

07II III

08IIII I

09IIII I

10I I III

11II III

12III II

13III I I

14IIII I

15

II III

01

Eletro s tati c

PLMN

PCM speech

C I SC O  S

Y STE MS

Ericsson SS7 SG

PSTN subscribers

C ISC O   SY ST EMS

Cisco 3640 H.323 Gateway

C I S CO   S Y S TE MS

Cisco Gatekeep

C I S CO  S

 Y ST E MS

SIP Proxy Server 

CI SCO  S

YST EMSCisco3600  SERIES

Ericsson VG

CISCOS

YSTEMSCisco3600 SERIES

Ericsson VG

CISCOS

YS T EMSCisco 3600  SERIES

Ericsson VG

PSTN and PLMN inIP network

Sitekeeper 

atakunta Polytechnic

PLMN subscribers

CI SCO   SYST EMS

Cisco 2620 SIP Gateway

Fig. 16: The telecommunications laboratory at Satakunta Polytechnic

PCM

ISDN PRI

PCM

ISDN BRI

L u ce n t L uc e nt L uc en t L uc e n t Lu c e n t L u ce n t L u ce nt L u ce nt L uc e nt Lu c en t L uc en tL u c en t L u c e n tL u ce n t L uc e n tL u ce nt L uc e ntL uc e n t Lu c en tL  u ce n t L u c en tL u ce n tL u c e nt L u c en t

PACKETCONT

NETCONT

PROCRINTFC

MEMO R Y TONEDET /GENTONECLO CK

POWER SUPPL Y

EXP NINTFC

02 03III II

04II I II

05III I I

06IIII I

07I III I

08IIIII

09I I III

10II III

11III II

12III II

13III I I

14IIII I

16 1 7

POW ERU NIT

18I IIIIII IIIC

TONECLOCK

PACKETCONT

NETCONT

TO NEDET / GENTO NECLOCK

02 03II III

04III II

05II I II

06III I I

07III II

08I IIII

09IIIII

10I I III

11I IIII

12III II

13III II

14III I I

15IIII I

01

Ele tr ost a t i c

PSTN

PC with NetMeeting

PC with NetMeeting

PCML uc e nt L uc en tL  u c en t L u c en tL u ce n t L uc e nt L u c en tL uc e n t Lu c e n t L u ce n t L u ce nt L u c e ntL uc e n t Lu c en tL  u ce n t L u c e nt L u ce n tL uc e nt Lu c en tL  u ce n t L u c e nt L u ce n tL u ce nt Lu c en t

PAC KETCONT

NETCONT

PROCRINTFC

MEMORY TON EDET/GEN

TON ECLOCK

POWER SUPPL Y

EXP NI NTFC

02 0 3IIII I

04III I I

05IIII I

06II III

07II III

0 8II I II

09III I I

10IIII I

11I IIII

12IIIII

13I I III

14III II

16 1 7

POWERUNI T

18IIII IIII II

C TON ECLOCK

P ACKETCONT

NETCONT

TONEDET /GENTONECLO CK

02 0 3III II

04IIII I

05III I I

06IIII I

07I IIII

08III II

0 9II I II

10III I I

11IIII I

12I IIII

13IIIII

14I I III

15III II

01

Ele tr ost a t i c

PLMN

PCM speech

CISC O S

 Y S TEMS

Ericsson SS7 SG

PSTN subscribers

C I SCO S

YST EMS

Cisco 3640 H.323 Gateway

Cisco Gateke

CISCO  S YST EMS

SIP Proxy Server 

Megaco/H.248

CI SCO SYSTEMS Cisco 3600  SERIES

Tigris MG

Megaco/H.248CISCO

 SYSTEMS

Cisco3600  SERIES

Tigris MGMegaco/H.248

CISCO S YSTEMSCisco 3600  SERIES

Tigris MG

PSTN and PLMN inIP networkSatakunta Polytechnic

PLMN subscribers

CI SCO S YSTEMS

Cisco 2620 SIP Gateway

CISCOS

 YSTEMS Cisco 3600  SERIES

Ericsson MGC

Fig. 17: Design of the next generation network at Satakunta

Polytechnic 

VI.  FUTURE PLANS

In the future the aim is to realize a network based on

the Megaco call control protocol. To achieve this goal it

is very important that manufacturers will cooperate with

us. This environment has many different properties

offering a valuable pilot network opportunity for

manufacturers and service providers. This network will

 be part of telecommunications research network whichoffers its properties to customers to develop their

 products and services. The concept of softswitch will be

more significant when third parties may offer their

services in this network. Softswitch uses open APIs,

such as PARLAY [13], OSA [14] or JAIN [15], to add

new services. These new service APIs will also be part

of the future of this network.

The use of this network for educational purposes is

very significant. In the future this network will be part

of students’ laboratory exercises and will help them

understand the meaning and benefits of network

convergence.

VII.  CONCLUSION 

This paper presents the architecture where the

Megaco/H.248 protocol is needed. The Megaco/H.248

 protocol has been presented and all involving terms

have been clarified. The operation of the Megaco

 protocol has been presented with examples. These

examples are useful when the Megaco protocol is taken

into use in the telecommunications laboratory.

Scalability and centralized intelligence are important

features in next generation networks. These features are

 possible to realize because of the decomposed gatewaysand the Megaco call control protocol.

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52

SIP and H.323 are very useful call signaling

 protocols in the Megaco architecture. The MGC is able

to connect SIP and H.323 terminals and not only to

control Media Gateways. This gives new opportunities

to develop services for the next generation networks

 based on H.323, SIP and Megaco/H.248. Satakunta

Polytechnic may exploit the Megaco architecture to

develop in the field of telecommunication. Thisarchitecture provides new solutions and service

opportunities and old network architectures, SIP

network and H.323 network, can be combined to work

as one network. Management and call control in the IP

network are centralized which makes this solution

superior to the old network. Because Satakunta

Polytechnic is an educational institute it is important

that new protocols can be studied and taken into

 practice.

R EFERENCES 

[1]  VocalTec – the first – and the best – in IP Telephony, Available:

http://www.vocaltec.com/html/telephony/introduction.htm 

[2]  ITU-T H.323, “Visual Telephone Systems and Equipment for

Local Area Networks which provide a Guaranteed quality of

Service,” 11/1996

[3]  H. Schulzrinne, E. Schooler, J. Rosenberg, “SIP: Session

Initiation Protocol, M. Handley,” RFC 2543, March 1999

[4]  ITU-T H.248, “Gateway Control Protocol,” 06/2000

[5]  ITU-T H.323 versio 4, “Packet-Based Multimedia

Communications Systems, “11/2000

[6]  J. Rosenberg, H. Schulzrinne, G. Camarillo, A. Johnston, J.

Peterson, R. Sparks, M. Handley, E. Schooler, “SIP: Session

Initiation Protocol ,” RFC 3261, June 2002

[7]  Cisco Systems, “Understanding Packet Voice Protocols,”

Available: http://www.sipcenter.com/files/Cisco_UPVP_wp.pdf  

[8]  IEEE Communications Magazine: Megaco/H.248: A NewStandard for Media Gateway Control, Tom Taylor, October 2000

[9]  ETSI Standard TR 101 300 V2.1.1, “Telecommunications and

Internet Protocol Harmonization Over Networks (TIPHON);

Description of technical issues,” 10/1999

[10]  Alberto Conte, Laurent-Philippe Anquetil and Thomas Levy,

“Experiencing Megaco Protocol for Controlling Non-

decomposable VoIP Gateways,” IEEE, 2000

[11]  M. Brahmanapally, P. Viswanadham, K. Gundamaraju,

“Megaco/H.248 Call flow examples ,” draft-ietf-megaco-

callflows-00.txt, IETF, March 2002 Available:

http://www.ietf.org/proceedings/02jul/I-D/draft-ietf-megaco-

callflows-00.txt 

[12]  R. Stewart, Q. Xie, K. Morneault, C. Sharp, H. Schwarzbauer, T.

Taylor, I. Rytina, M. Kalla, L. Zhang, V. Paxson, “Stream

Control Transmission Protocol,” RFC 2960, October 2000

[13]  Parlay Group web page, Available: http://www.parlay.org/ 

[14]  ETSI Standard ES 201 915-1 V1.3.1, ETSI Open Service(OSA);

Application Programming Interface (API), 2002

[15]  The JAIN APIs overview, Available:

http://java.sun.com/products/jain/overview.html 

Jani Peltola  received his BEng degree in telecommunications from

Satakunta Polytechnic. After graduation he started to work as a

Research & Development Engineer at Satakunta Polytechnic.

Simultaneously he started to study for Master’s Degree and received

his M.Sc. degree in telecommunications from Tampere University of

Technology, Pori Finland, in 2002. He was born in Kiikoinen, Finland

in 1976.

Since 2000 he has been working at Satakunta Polytechnic. His

current research interests are in the fields of media transportation

 between packet network and switched telephone network.