ss7 and sip

Signaling SystemNo. 7

By Lee Dryburgh andJeff Hewett

Table of Contents

Copyright

About the Authors

Acknowledgments

Icons Used in This Book

Command SyntaxConventions

Introduction

Part I. Introductions andOverviews

Chapter 1. The Evolution ofSignaling

Chapter 2. Standards

Chapter 3. The Role of SS7

Chapter 4. SS7 NetworkArchitecture and ProtocolsIntroduction

Chapter 5. The PublicSwitched Telephone Network(PSTN)

Part II. Protocols Found in theTraditional SS7/C7 Stack

Chapter 6. Message TransferPart 2 (MTP2)

Chapter 7. Message TransferPart 3 (MTP3)

Chapter 8. ISDN User Part(ISUP)

Chapter 9. SignalingConnection Control Part(SCCP)

Chapter 10. TransactionCapabilities Application Part(TCAP)

Part III. Service-orientedProtocols

Chapter 11. IntelligentNetworks (IN)

Chapter 12. Cellular Networks

Chapter 13. GSM andANSI-41 Mobile ApplicationPart (MAP)

Part IV. SS7/C7 Over IP

Chapter 14. SS7 in theConverged World

Next GenerationArchitecture

SigTran

User Adaptation (UA)

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SS7 and SIP/H.323 InterworkingThe ITU-T originally developed the H.323 [125] for multimedia over Local Area Networks (LANs). It is not a single protocol; rather, itis a vertically-integrated suite of protocols that define the components and signaling. Though it was originally used for video-conferencing, H.323 was enhanced to better support VoIP with the Version 2 release. It is currently the most widely-deployed VoIPsolution today.

One of the main complaints about H.323 is its complexity. With H.323, many messages must be passed to set up even a basicvoice call. SIP [124], is considered a simpler, more flexible alternative to H.323. SIP is a signaling protocol that handles the setup,modification and teardown of multimedia sessions. It was developed in the IETF as a signaling protocol for establishing sessions inan IP network. A session can be a simple two-way telephone call or a multimedia conference. SIP is becoming a popular favorite asthe future of VoIP.

So, how does SigTran play a role in H.323 and SIP? SigTran can provide PSTN connectivity to H.323 and SIP networks. A PSTNGateway application can be used to fulfill this need. The PSTN Gateway sits on the edge of the circuit-switched and packet-switched networks and provides SIP or H.323 interworking to SS7 in the PSTN. Figure 14-29 shows an example of an SIP PSTNGateway application. In this example, the MGC connects to the SGs using SigTran.

Figure 14-29 SIP-PSTN Gateway Application

Figure 14-30 shows a similar example of an H.323 PSTN Gateway application.

Figure 14-30 H.323-PSTN Gateway Application

Another interesting application is the PSTN transit application, in which calls originate and terminate on TDM interfaces and thentransit a voice packet network (such as SIP or H.323). Service providers can use this application to offload their tandem and transitClass 4 and Class 3 switches. This application creates the need for an ISUP transparency. SIP-T [150] (SIP for Telephones)provides a framework for the integration of the PSTN with SIP. Figure 14-31 shows an example of using SIP-T for a PSTN transitapplication.

Figure 14-31 SIP Transit Application

SIP-T meets the SS7 to SIP interworking requirements by providing the following functions:

A standard way of mapping ISUP information into the SIP header for calls that originate in the PSTN. This function ensures thatthe SIP contains sufficient information to route calls (for example, in the case where routing depends on some ISUPinformation).

Use of the SIP INFO [151] Method to transfer mid-call ISUP signaling messages.

A means for MIME [152] encapsulation of the ISUP signaling information in the SIP body provides for ISUP transparency.

When the MGC receives an ISUP message, the appropriate ISUP parameters are translated to the SIP header fields and the ISUPmessage is encapsulated in a MIME attachment, which intermediate SIP entities treat as an opaque object. If the SIP messageterminates the call, it ignores the ISUP attachment because it has no need for it. However, if the call terminates on the PSTN, theencapsulated ISUP message is examined and used to generate the outgoing ISUP message. The version parameter included inthe MIME media type information indicates the encapsulated ISUP message's ISUP variant. If there are different ISUP variants onthe origination and termination side, it is up to the terminating MGC to perform ISUP translation between the variants.

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Layers

MTP Level 3 UA (M3UA)

SCCP User Adaptation(SUA)

MTP Level 2 UserAdaptation (M2UA)

ISDN User Adaptation(IUA)

Transport AdaptationLayer Interface (TALI)

Early Cisco SS7/IPSolution

SS7 and SIP/H.323Interworking

Summary

Part V. Supplementary Topics

Chapter 15. SS7 Security andMonitoring

Chapter 16. SS7 Testing

Part VI. Appendixes

Acronyms

Appendix A. MTP Messages(ANSI/ETSI/ITU)

Appendix B. ISUP Messages(ANSI/UK/ETSI/ITU-T)

Appendix C. SCCP Messages(ANSI/ETSI/ITU-T)

Appendix D. TCAP Messagesand Components

Appendix E. ITU-T Q.931Messages

Appendix F. GSM and ANSIMAP Operations

Appendix G. MTP Timers inITU-T/ETSI/ANSI Applications

Appendix H. ISUP Timers forANSI/ETSI/ITU-T Applications

Appendix I. GSM MobileCountry Codes (MCC) andMobile Network Codes (MNC)

Appendix J. ITU and ANSIProtocol Comparison

Appendix K. SS7 Standards

Appendix L. TektronixSupporting Traffic

Appendix M. Cause Values


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ss7 and sip

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