Towards Megaco Architecture

J. Peltola, J. Aromaa, A. Mustonen Satakunta Polytechnic November 28th 2002

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

AbstractThe 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 TermsVoice Convergence over IP, Megaco, Network

I. INTRODUCTION
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 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 and how to make calls between these two networks.

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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]
MCU H.323 Terminals

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 accepts registration requests. The Registrar is typically colocated 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]
SIP Redirect Server

IP IP IP IP
SIP Proxy Server SIP Proxy Server and Registrar

IP IP

PSTN PSTN
H.323 Gateway

SIP User Agent

SIP User Agent

IP IP

Fig. 2: Major SIP components

ATM ATM
H.323 Gateway

H.323 Gatekeeper

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 the Media 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 in 1998. These two protocols combined and the resulting

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

B. SIP The Session Initiation Protocol (SIP) is an application-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.

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MGC

IP rS 3o 32 H.

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.

Telephone Signaling (ISUP) Exchange

TP SC

.248 aco/H Meg

SG

Subscriber

RTP

M G

MG

Fig.4: Gateway distribution and control

one or more MGs are slaves. The Megaco protocol provides means for describing and controlling the connections of media streams. The Megaco/H.248
Media Gateway
Context
Termination Termination RTP Stream Stream RTP
Termination SCNBearer Bearer SCN Channel Channel

*
*
*

RT P

PSTN PSTN
M PC
Context
Termination RTP Stream Stream RTP

8 .24 o/H c a eg M

IP IP

VoIP Term inal

Termination Termination SCNBearer Bearer SCN Channel

Null Context Context Null

Termination Termination SCNBearer Bearer SCN Channel Channel

Context

MGC
S ign a lin g C o ntrol M edia C o ntrol

Termination Termination RTP Stream Stream RTP

Termination Termination SCNBearer Bearer SCN Channel Channel

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

SG

MG

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

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]

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]

A. The Connection Model The connection model for the Megaco protocol describes the logical entities within the MG that can be controlled by the MGC. The main abstractions used in the connection model are Terminations and Contexts. [4] A Termination sources and/or sinks one or more streams. In a multimedia conference, a Termination can be multimedia and sources or sinks multiple media streams. [4] A Context is an association between a collection of Terminations. It describes the topology (who hears/sees whom) of associated Terminations and the media mixing if more than one Terminations are involved in the association. There is a special type of Context, the null Context, which contains all Terminations that are not associated to any other Termination (e.g. idle lines in the access gateway). These concepts are shown in Fig. 5. [4]

IV. MEGACO/H.248 PROTOCOL The Megaco/H.248 protocol is a master/slave protocol. In this architecture the MGC is a master and

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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 Add command 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 Add EXPLANATION Command adds a Termination to a Context. The Add command on the first Termination in a Context is used to create a Context. Command modifies the properties, events and signals of a Termination. Command disconnects a Termination from its Context and returns statistics on the Termination's participation in the Context. The Subtract command on the last Termination in a Context deletes the Context. Command atomically moves a Termination to another Context. Command returns the current state of properties, events, signals and statistics of Terminations. Command returns all the possible values for Termination properties, events and signals allowed by the MG. Command allows the MG to inform the MGC of the occurrence of events in the MG. 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. Direction MGC - MG

Modify Subtract

MGC - MG MGC - MG

Move AuditValue

MGC - MG MGC - MG

AuditCapabilities

MGC - MG

Notify

MG - MGC

ServiceChange

MGC - MG MG - MGC

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TRANSACTIONx
ContextID1

Command1

Command1

Command3

ContextID2

Command1

ContextID3

Command1

Command2

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. 1) MGC, MG and analogous subscriber The first example describes a situation where analogous subscriber wants to call another analogous subscriber. 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

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

MGC

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 version number 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]
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]

Megaco/H.248

Megaco/H.248

MG1

RTP

MG2

Fig. 8: MG equipped with analogous interface

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 the Megaco/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

protocols used in this scenario are described in Fig 8. 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 Analogous Termination. 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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The end of the call is not described in this flow chart (Fig. 9). The call termination starts when the MGC is informed about off hook. Then the MGC sends the Subtract command to both MGs. With this command the Context between the RTP Termination and Analogous Termination is terminated. The RTP Termination stops to exist and Analogous Termination is placed to Null Context to detect new events.

MG 1
ServiceChange Reply Modify Reply

MGC
ServiceChange Reply Modify Reply

MG 2

Off hook

Notify Reply Modify

Dialing

Notify Reply Add Reply Modify Reply Notify Reply Modify Reply Add Reply

Answer

In this example the call is originated from the PSTN. The MGC receives the IAM (Initial Address Message) from the phone exchange. The transfer layer of the ISUP message (ISDN User Part) is changed in the SG to transfer information also in the packet network. The MGC replies to the IAM with the ACM (Address Complete Message). The MGC is aware of the destination of the incoming call. The MGC adds Terminations to the MG1: 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 packet network. These two Terminations are now in one Context. Terminations are also added to MG2. Both MGs send Reply to the Add command. All commands and notifications in this example are replied. A Context is created where one Termination represents the Analogous bearer and another the RTP stream. The MG2 is ordered to send ringing information to the user. The MG2 detects off hook and sends this information to the MGC. The MGC knows that the call has been answered and reports this to the PSTN side with an ANM (Answer Message). The MGC modifies the RTP Terminations in both MGs with address information. After this the two-way RTP stream will start between the MG1 and the MG2. Flow chart of this example is shown in Fig. 11.

Modify Reply RTP-stream

PSTN
IAM ACM

MG1

MGC

MG2

Add

Fig. 9: Flow chart when the MGC instructs the MG to setup a call

Reply Add Alerting

2) Interworking with the PSTN This example describes how the PSTN and a network based on the Megaco protocol are working together. Signaling information (SS7), associated with a call coming from the PSTN side, is transferred through the SG and terminated to the MGC. Speech information between the two networks is transferred through the MGs. The phone exchange controls the PSTN calls and the MGC controls calls in the IP network. Main components and protocols of this example are shown in Fig. 10.
SG
SS7
Megaco/H.248

Reply Answer

Notify ANM Reply Modify

Reply

Modify

Reply RTP Stream

MGC
Megaco/H.248 RTP

Fig. 11: Example of the flow chart when combining two networks

Telephone Exchange

PCM

MG1

MG2

Fig. 10: Interworking with the PSTN

3) Interaction betweenthe MGC and the VoIP terminal It is possible that the MGC has to control VoIP terminals such as SIP or H.323 terminals. To do this the MGC needs to handle SIP and H.323 protocols. In the SIP environment the MGC acts as a user agent and in the H.323 environment the MGC sends RAS messages

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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.

PSTN
IAM

MG
Add

MGC SIP User Agent

Reply Invite

Telephone Exchange

SS7

SG
8 .24 o/H c a eg M

MGC
Ringing Modify

Alerting

P SI

Fig. 12: Megaco and SIP

ANM

M PC

MG

RTP

SIP User Agent

ACM Reply Answer 200 OK

Telephone Exchange

SS7

SG
8 .24 /H co a eg M

MGC

H.323

GK
RTP Stream

ACK

Hang up REL Bye

H.323

Fig. 13: Megaco and H.323

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 both Terminations. 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 and sends 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.

M PC

RTP

MG

H.323 Terminal

Subtrack

Reply 200 OK RLC

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

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. 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 networks 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 call signaling 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. 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 Modify message 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. 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.
PSTN
IAM Add Reply ACM

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. 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. A. Laboratory environment without the Megaco/H.248 call control The telecommunications laboratory at Satakunta Polytechnic 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). 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 the PSTN, SIP or H.323 networks. The most suitable

MG

MGC

GK

H.323 Terminal

ARQ ACF Setup ARQ ACF Alerting

ANM

Connect Terminal Capability Set Master Slave Determination Open Logical Channels Modify
Reply RTP Stream

Close Logical Channels

REL

Release Complete DRQ DCF Subtract

Reply

DRQ DCF

RLC

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

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architecture for this purpose is Ericssons 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.
PSTN and PLMN in Satakunta Polytechnic
PLMN subscribers Ericsson VG
CI SCO

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.

IP network

PSTN and PLMN in Satakunta Polytechnic

PC with NetMeeting PC with NetMeeting
PLMN subscribers Tigris MG

IP network

YSTEM S

Cisco 3600

S ERIES

PLMN
E e lt r ost at i c

Luc ent Lu cen t L uce nt Luc ent Lu cen t Lu cen t L uce nt Luc ent Lu cen t L uce nt Luc ent Lu cen t L uce nt Luc ent Luc ent Lu cen t Luc ent Lu cen t Lu cen t L uce nt Luc ent Lu cen t Lu cen t

Lu cen t

PCM Ericsson SS7 SG

C IS C O

01 TON E C LOCK

M EM 02 ORY E XPN I NTFC

PRNET 02 03I OCR I NTFC CON T II III

TO NE TONE PANET 03I 04I CKE TPA CKE 04I T 05I 06I 05I CO NT I I I DCET / ONT GEN I I I DET/ G EN II TO NE TONE CLOCK CLOCK

07 06I III

08 07I III

09I 08I III

09I 10I II

10 1I 1 III

11I 12I II

12I 13I II

14I 13I III

15I 14I III III

16 III

17

18

P OW E R S UP P LY

POW ER

UNI T

Cisco Gatekeep
C I S CO

C I SC O

YSTEM S

Cisco 3600

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Fig. 16: The telecommunications laboratory at Satakunta Polytechnic

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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 offers Tigris 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

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 which offers 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 gateways and the Megaco call control protocol.

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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. This architecture 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. REFERENCES
[1] [2] [3] [4] [5] [6] [7] [8] [9] [10] [11] VocalTec the first and the best in IP Telephony, Available: http://www.vocaltec.com/html/telephony/introduction.htm ITU-T H.323, Visual Telephone Systems and Equipment for Local Area Networks which provide a Guaranteed quality of Service, 11/1996 H. Schulzrinne, E. Schooler, J. Rosenberg, SIP: Session Initiation Protocol, M. Handley, RFC 2543, March 1999 ITU-T H.248, Gateway Control Protocol, 06/2000 ITU-T H.323 versio 4, Packet-Based Multimedia Communications Systems, 11/2000 J. Rosenberg, H. Schulzrinne, G. Camarillo, A. Johnston, J. Peterson, R. Sparks, M. Handley, E. Schooler, SIP: Session Initiation Protocol , RFC 3261, June 2002 Cisco Systems, Understanding Packet Voice Protocols, Available: http://www.sipcenter.com/files/Cisco_UPVP_wp.pdf IEEE Communications Magazine: Megaco/H.248: A New Standard for Media Gateway Control, Tom Taylor, October 2000 ETSI Standard TR 101 300 V2.1.1, Telecommunications and Internet Protocol Harmonization Over Networks (TIPHON); Description of technical issues, 10/1999 Alberto Conte, Laurent-Philippe Anquetil and Thomas Levy, Experiencing Megaco Protocol for Controlling Nondecomposable VoIP Gateways, IEEE, 2000 M. Brahmanapally, P. Viswanadham, K. Gundamaraju, Megaco/H.248 Call flow examples , draft-ietf-megacocallflows-00.txt, IETF, March 2002 Available: http://www.ietf.org/proceedings/02jul/I-D/draft-ietf-megacocallflows-00.txt 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 Parlay Group web page, Available: http://www.parlay.org/ ETSI Standard ES 201 915-1 V1.3.1, ETSI Open Service(OSA); Application Programming Interface (API), 2002 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 Masters 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.

[12] [13] [14] [15]

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