Inter-asterisk exchange (IAX) deployment scenarios in SIP-enabled networks / / Mohamed Boucadair |
Autore | Boucadair Mohamed |
Pubbl/distr/stampa | Chichester, U.K. : , : Wiley, , 2009 |
Descrizione fisica | 1 online resource (275 p.) |
Disciplina |
004.69/5
004.695 |
Collana | Wiley series on communications networking & distributed systems |
Soggetto topico |
Internet telephony
Computer network protocols |
ISBN |
1-282-01079-4
9786612010798 0-470-74218-6 0-470-74219-4 |
Formato | Materiale a stampa ![]() |
Livello bibliografico | Monografia |
Lingua di pubblicazione | eng |
Nota di contenuto |
Foreword -- Terminology and Definitions -- Acronyms and Abbreviations -- Acknowledgement -- 1 Introduction -- 1.1 General Introduction -- 1.2 On Voice over IP and Telephony over IP -- 1.3 Context -- 1.4 Enhancement Strategies to Solve SIP Issues -- 1.5 IAX: Towards Lightweight Telephony Architectures -- 1.6 IAX and Standardisation -- 1.7 Rationale -- 1.8 What This Book is Not -- 1.9 Structure of the Book -- 2 The IAX Protocol at a Glance -- 2.1 What Does IAX Stand For? -- 2.2 Is IAX Specific to the Asterisk Platform? -- 2.3 What is the Difference between IAX2 and IAX? -- 2.4 Why another New VoIP Protocol? -- 2.5 How Does IAX Solve VoIP Pains? -- 2.6 How is Calls Multiplexing Achieved? -- 2.7 And What About Demultiplexing? -- 2.8 What Port Number Does IAX Use? -- 2.9 What Transport Protocol Does IAX Use? -- 2.10 Is IAX a Reliable Protocol? -- 2.11 How Does IAX Ensure Reliability? -- 2.12 Is there an IAX Registration Procedure? -- 2.13 Does IAX Registration Differ from SIP Registration? -- 2.14 How Are Media Streams Transported in IAX? -- 2.15 Is CODEC Negotiation Supported by IAX? -- 2.16 Is On-Fly CODEC Change Possible During a Call? -- 2.17 IAX: a Path-Coupled or Decoupled Protocol? -- 2.18 Can IAX be Aware of the Status of the Network Load? -- 2.19 What About Security? -- 2.20 Could IAX Devices Be Managed? -- 2.21 Is Firmware Version Updating Supported by IAX? -- 2.22 Can IAX Be Extended to Offer New Features? -- 2.23 How is an IAX Resource Identified? -- 2.24 What Does an IAX URI Look Like? -- 2.25 Is it Possible to Set a Call Involving Several IAX Servers? -- 2.26 Is it Possible to Discover the Location of an IAX Resource? -- 2.27 What Is DUNDi? -- 2.28 What Is TRIP? -- 2.29 What Is ENUM? -- References -- Further Reading -- Part One: IAX Protocol Specifications -- 3 IAX Uniform Resource Identifier -- 3.1 Introduction -- 3.2 Format of IAX Uniform Resource Identifiers -- 3.3 Examples of IAX Uniform Resource Identifiers -- 3.4 Comparing IAX Uniform Resource Identifiers.
3.5 IAX Uniform Resource Identifiers and ENUM -- References -- Further Reading -- 4 IAX Frames -- 4.1 Introduction -- 4.2 Full Frames -- 4.3 Mini Frames -- 4.4 Meta Frames -- 4.5 Encrypted Frames -- 4.6 Conclusion -- References -- Further Reading -- 5 IAX Information Elements -- 5.1 Introduction -- 5.2 List of IAX Information Elements -- 5.3 Example of IAX Information Element Traces -- References -- Further Reading -- 6 IAX Messages -- 6.1 Introduction -- 6.2 Taxonomy of IAX Messages -- 6.3 IAX Requests/Responses -- 6.4 IAX Functional Categories -- 6.5 IAX Media Frames -- 6.6 IAX Reliable/Unreliable Messages -- References -- Further Reading -- 7 IAX Connectivity Considerations -- 7.1 Introduction -- 7.2 IAX Transport Protocol -- 7.3 IAX Port Number -- 7.4 IAX Call Multiplexing and Demultiplexing -- 7.5 IAX Reliability Mechanism -- 7.6 Authentication and Encryption -- 7.7 Conclusion -- References -- Further Reading -- 8 IAX Operations -- 8.1 Introduction -- 8.2 Provisioning and Firmware Download -- 8.3 Registration -- 8.4 Call Setup -- 8.5 Call Tear-Down -- 8.6 Call Monitoring -- 8.7 Call Optimisation -- 8.8 Conclusion -- References -- Further Reading -- Part Two: Discussion and Analysis -- 9 IAX and Advanced Services -- 9.1 Introduction -- 9.2 CODEC Negotiation -- 9.3 Video Sessions -- 9.4 Negotiation of Several Media Types in the Same IAX Session -- 9.5 Presence Services -- 9.6 Instant Messaging -- 9.7 Topology Hiding -- 9.8 Mobility -- 9.9 Miscellaneous -- 9.10 Conclusion -- References -- Further Reading -- 10 Multi-IAX Servers Environment -- 10.1 Introduction -- 10.2 Focus -- 10.3 Discovery of IAX Resources -- 10.4 Setting End-to-End Calls -- 10.5 Load Balancing -- 10.6 Path-Coupled and Path-Decoupled Discussion -- 10.7 Forking -- 10.8 Route Symmetry -- 10.9 Conclusion -- References -- 11 IAX and NAT Traversal -- 11.1 Introduction -- 11.2 Structure -- 11.3 NAT Types -- 11.4 IAX and NAT Traversal Discussion -- 11.5 Operational Considerations -- 11.6 Conclusion -- References. Further Reading -- 12 IAX and Peer-to-Peer Deployment Scenarios -- 12.1 Introduction -- 12.2 Scope -- 12.3 A P2P Solution for Corporate Customers -- 12.4 Conclusion -- References -- Further Reading -- 13 IAX and IPv6 -- 13.1 Introduction -- 13.2 Context and Assumptions -- 13.3 Service Migration to IPv6 -- 13.4 Structure -- 13.5 The IP Address Exhaustion Problem -- 13.6 IPv6: a Long-Term Solution -- 13.7 Why IPv6 May Be Problematic for Telephony Signalling Protocols: the SIP Example -- 13.8 IAX: an IP Version-Agnostic Protocol? -- 13.9 Deployment of IAX Services in a 'Pure' IPv6 Environment -- 13.10 Heterogeneous Environment -- 13.11 Conclusion -- References -- Further Reading -- 14 IAX: Towards a Lightweight SBC? -- 14.1 Introduction -- 14.2 IP Telephony Administrative Domain -- 14.3 Deployment Scenarios -- 14.4 Deployment Contexts -- 14.5 Service Limitations Caused by SBCs -- 14.6 Functional Decomposition -- 14.7 Taxonomy of SBC Functions in an SIP Environment -- 14.8 Validity of these Functions in an IAX Architecture -- 14.9 Conclusion -- References -- Further Reading -- Part Three: Deployment Scenarios in SIP-Based Environments -- 15 Scenarios for the Deployment of IAX-Based Conversational Services -- 15.1 SIP Complications -- 15.2 Structure -- 15.3 Beyond the 'SIP-Centric' Era -- 15.4 Methodology -- 15.5 Overall Context -- 15.6 Architectural Requirements -- 15.7 Brief Comparison -- 15.8 Taxonomy -- 15.9 Introducing IAX into Operational Networks -- 15.10 Conclusion. / -- References -- 16 IAX in the Access Segment of SIP-Based Service Architectures -- 16.1 Introduction -- 16.2 A 'High-Level' Description of the Interworking Function -- 16.3 Examples of Call Flows -- 16.4 Bandwidth Optimisation: An Extension to SIP -- 16.5 Conclusion -- References -- 17 Validation Scenario -- 17.1 Overview -- 17.2 Configuring Asterisk Servers -- 17.3 Configuring the SIP Express Router (SER) -- 17.4 User Agent Configuration -- 17.5 Conclusion -- Further Reading -- Index. |
Record Nr. | UNINA-9910830232403321 |
Boucadair Mohamed
![]() |
||
Chichester, U.K. : , : Wiley, , 2009 | ||
![]() | ||
Lo trovi qui: Univ. Federico II | ||
|
Inter-asterisk exchange (IAX) deployment scenarios in SIP-enabled networks / / Mohamed Boucadair |
Autore | Boucadair Mohamed |
Pubbl/distr/stampa | Chichester, U.K., : Wiley, 2009 |
Descrizione fisica | 1 online resource (275 p.) |
Disciplina |
004.69/5
004.695 |
Collana | Wiley series in communications networking & distributed systems |
Soggetto topico |
Internet telephony
Computer network protocols |
ISBN |
1-282-01079-4
9786612010798 0-470-74218-6 0-470-74219-4 |
Formato | Materiale a stampa ![]() |
Livello bibliografico | Monografia |
Lingua di pubblicazione | eng |
Nota di contenuto |
Foreword -- Terminology and Definitions -- Acronyms and Abbreviations -- Acknowledgement -- 1 Introduction -- 1.1 General Introduction -- 1.2 On Voice over IP and Telephony over IP -- 1.3 Context -- 1.4 Enhancement Strategies to Solve SIP Issues -- 1.5 IAX: Towards Lightweight Telephony Architectures -- 1.6 IAX and Standardisation -- 1.7 Rationale -- 1.8 What This Book is Not -- 1.9 Structure of the Book -- 2 The IAX Protocol at a Glance -- 2.1 What Does IAX Stand For? -- 2.2 Is IAX Specific to the Asterisk Platform? -- 2.3 What is the Difference between IAX2 and IAX? -- 2.4 Why another New VoIP Protocol? -- 2.5 How Does IAX Solve VoIP Pains? -- 2.6 How is Calls Multiplexing Achieved? -- 2.7 And What About Demultiplexing? -- 2.8 What Port Number Does IAX Use? -- 2.9 What Transport Protocol Does IAX Use? -- 2.10 Is IAX a Reliable Protocol? -- 2.11 How Does IAX Ensure Reliability? -- 2.12 Is there an IAX Registration Procedure? -- 2.13 Does IAX Registration Differ from SIP Registration? -- 2.14 How Are Media Streams Transported in IAX? -- 2.15 Is CODEC Negotiation Supported by IAX? -- 2.16 Is On-Fly CODEC Change Possible During a Call? -- 2.17 IAX: a Path-Coupled or Decoupled Protocol? -- 2.18 Can IAX be Aware of the Status of the Network Load? -- 2.19 What About Security? -- 2.20 Could IAX Devices Be Managed? -- 2.21 Is Firmware Version Updating Supported by IAX? -- 2.22 Can IAX Be Extended to Offer New Features? -- 2.23 How is an IAX Resource Identified? -- 2.24 What Does an IAX URI Look Like? -- 2.25 Is it Possible to Set a Call Involving Several IAX Servers? -- 2.26 Is it Possible to Discover the Location of an IAX Resource? -- 2.27 What Is DUNDi? -- 2.28 What Is TRIP? -- 2.29 What Is ENUM? -- References -- Further Reading -- Part One: IAX Protocol Specifications -- 3 IAX Uniform Resource Identifier -- 3.1 Introduction -- 3.2 Format of IAX Uniform Resource Identifiers -- 3.3 Examples of IAX Uniform Resource Identifiers -- 3.4 Comparing IAX Uniform Resource Identifiers.
3.5 IAX Uniform Resource Identifiers and ENUM -- References -- Further Reading -- 4 IAX Frames -- 4.1 Introduction -- 4.2 Full Frames -- 4.3 Mini Frames -- 4.4 Meta Frames -- 4.5 Encrypted Frames -- 4.6 Conclusion -- References -- Further Reading -- 5 IAX Information Elements -- 5.1 Introduction -- 5.2 List of IAX Information Elements -- 5.3 Example of IAX Information Element Traces -- References -- Further Reading -- 6 IAX Messages -- 6.1 Introduction -- 6.2 Taxonomy of IAX Messages -- 6.3 IAX Requests/Responses -- 6.4 IAX Functional Categories -- 6.5 IAX Media Frames -- 6.6 IAX Reliable/Unreliable Messages -- References -- Further Reading -- 7 IAX Connectivity Considerations -- 7.1 Introduction -- 7.2 IAX Transport Protocol -- 7.3 IAX Port Number -- 7.4 IAX Call Multiplexing and Demultiplexing -- 7.5 IAX Reliability Mechanism -- 7.6 Authentication and Encryption -- 7.7 Conclusion -- References -- Further Reading -- 8 IAX Operations -- 8.1 Introduction -- 8.2 Provisioning and Firmware Download -- 8.3 Registration -- 8.4 Call Setup -- 8.5 Call Tear-Down -- 8.6 Call Monitoring -- 8.7 Call Optimisation -- 8.8 Conclusion -- References -- Further Reading -- Part Two: Discussion and Analysis -- 9 IAX and Advanced Services -- 9.1 Introduction -- 9.2 CODEC Negotiation -- 9.3 Video Sessions -- 9.4 Negotiation of Several Media Types in the Same IAX Session -- 9.5 Presence Services -- 9.6 Instant Messaging -- 9.7 Topology Hiding -- 9.8 Mobility -- 9.9 Miscellaneous -- 9.10 Conclusion -- References -- Further Reading -- 10 Multi-IAX Servers Environment -- 10.1 Introduction -- 10.2 Focus -- 10.3 Discovery of IAX Resources -- 10.4 Setting End-to-End Calls -- 10.5 Load Balancing -- 10.6 Path-Coupled and Path-Decoupled Discussion -- 10.7 Forking -- 10.8 Route Symmetry -- 10.9 Conclusion -- References -- 11 IAX and NAT Traversal -- 11.1 Introduction -- 11.2 Structure -- 11.3 NAT Types -- 11.4 IAX and NAT Traversal Discussion -- 11.5 Operational Considerations -- 11.6 Conclusion -- References. Further Reading -- 12 IAX and Peer-to-Peer Deployment Scenarios -- 12.1 Introduction -- 12.2 Scope -- 12.3 A P2P Solution for Corporate Customers -- 12.4 Conclusion -- References -- Further Reading -- 13 IAX and IPv6 -- 13.1 Introduction -- 13.2 Context and Assumptions -- 13.3 Service Migration to IPv6 -- 13.4 Structure -- 13.5 The IP Address Exhaustion Problem -- 13.6 IPv6: a Long-Term Solution -- 13.7 Why IPv6 May Be Problematic for Telephony Signalling Protocols: the SIP Example -- 13.8 IAX: an IP Version-Agnostic Protocol? -- 13.9 Deployment of IAX Services in a 'Pure' IPv6 Environment -- 13.10 Heterogeneous Environment -- 13.11 Conclusion -- References -- Further Reading -- 14 IAX: Towards a Lightweight SBC? -- 14.1 Introduction -- 14.2 IP Telephony Administrative Domain -- 14.3 Deployment Scenarios -- 14.4 Deployment Contexts -- 14.5 Service Limitations Caused by SBCs -- 14.6 Functional Decomposition -- 14.7 Taxonomy of SBC Functions in an SIP Environment -- 14.8 Validity of these Functions in an IAX Architecture -- 14.9 Conclusion -- References -- Further Reading -- Part Three: Deployment Scenarios in SIP-Based Environments -- 15 Scenarios for the Deployment of IAX-Based Conversational Services -- 15.1 SIP Complications -- 15.2 Structure -- 15.3 Beyond the 'SIP-Centric' Era -- 15.4 Methodology -- 15.5 Overall Context -- 15.6 Architectural Requirements -- 15.7 Brief Comparison -- 15.8 Taxonomy -- 15.9 Introducing IAX into Operational Networks -- 15.10 Conclusion. / -- References -- 16 IAX in the Access Segment of SIP-Based Service Architectures -- 16.1 Introduction -- 16.2 A 'High-Level' Description of the Interworking Function -- 16.3 Examples of Call Flows -- 16.4 Bandwidth Optimisation: An Extension to SIP -- 16.5 Conclusion -- References -- 17 Validation Scenario -- 17.1 Overview -- 17.2 Configuring Asterisk Servers -- 17.3 Configuring the SIP Express Router (SER) -- 17.4 User Agent Configuration -- 17.5 Conclusion -- Further Reading -- Index. |
Record Nr. | UNINA-9910876816403321 |
Boucadair Mohamed
![]() |
||
Chichester, U.K., : Wiley, 2009 | ||
![]() | ||
Lo trovi qui: Univ. Federico II | ||
|
Securing VoIP : keeping your VoIP network safe / / Regis J. (Bud) Bates ; acquiring editor, Steve Elliot ; designer, Greg Harris |
Autore | Bates Regis J., Bud |
Edizione | [1st edition] |
Pubbl/distr/stampa | Waltham, Massachusetts : , : Syngress, , 2015 |
Descrizione fisica | 1 online resource (222 p.) |
Disciplina | 004.695 |
Soggetto topico | Internet telephony - Security measures |
Soggetto genere / forma | Electronic books. |
ISBN | 0-12-417122-2 |
Formato | Materiale a stampa ![]() |
Livello bibliografico | Monografia |
Lingua di pubblicazione | eng |
Nota di contenuto |
Cover; Title Page; Copyright Page; Contents; Technical editor biography; About the author; Expertise; Publications; Books; Articles; Acknowledgments; Chapter 1 - Introduction; Securing Voice over Internet Protocol (VoIP): keeping your network safe; History of telephony; History of the Internet Protocol; What goes around comes around; VoIP network and potential problems; The benefits of VoIP; Some initial thoughts on VoIP; What are the reasons for the VoIP hacking attempts; The need for VoIP security; Need for security and causes; Technology; Policy; Terms and attacks; Other vulnerabilities
What is at riskCan a call be eavesdropped?; There is no Holy Grail out there; Summary; Chapter 2 - Policies; What is the problem?; The call control channel - hijacking; Softphone issues; Denial-of-service attacks; Security concerns; Security policy needs; Vulnerability detection and auditing; Is the system vulnerable?; Chapter 3 - VoIP virtual private networks (VPNs); Virtual private networks (VPNs) and encryption; What is a VPN?; The possible VPN solutions; What a VPN can offer; What everyone expects from securing VoIP; What is the impact?; Creating the VPN; IPSec used for site-to-site VPNs Disadvantages of IPSec VPN site-to-site tunnelsSummary; Chapter 4 - Cryptography solutions; Cryptography solutions; What is cryptography and encryption?; Early ciphers used; Digital signatures; Leads to a public key infrastructure; X.509 certificates; Digital certificate servers; Installing certificates on the devices; Summary; Chapter 5 - Authentication; Authentication defined; Details of 802.1x authentication; Use a VoIP-enabled firewall; Use 802.1X authentication for IP phones; Attacking VoIP Authentication; Encrypt the traffic; Authentication on wireless networks; Summary Chapter 6 - Other protocolsOther protocols; Overview of Real-Time Transport Protocol and Real-Time Transport Control Protocol (RTCP); RTCP; Function of Secure RTP; Enter SRTP and SRTCP; SRTP framework; Secure RTP using ZRTP; Mikey; Modes of MIKEY; Preshared key transfer; Public key transfer; Public key with Diffie-Hellman exchange; Transport protocols; Signaling: Session Initiation Protocol; Attacks on SIP; Denial of service; BYE; Authentication; Secure SIP; Transport and network layer security; Summary; Chapter 7 - The business case for securing VoIP; Before we start Overview of the RFC 2196Internal issues; Toll fraud - a big threat; Summary; Chapter 8 - Approaches to VoIP security; Before we start; Build it in layers; Some best practices for infrastructure security; Integrating network security; Additional thoughts and items; Registration spoofing; Summary; Chapter 9 - Final thoughts; Before we start; What we have already covered; Vendor issues; Controlling the risks; PBX best practices; Summary; Index |
Record Nr. | UNINA-9910463801603321 |
Bates Regis J., Bud
![]() |
||
Waltham, Massachusetts : , : Syngress, , 2015 | ||
![]() | ||
Lo trovi qui: Univ. Federico II | ||
|
Securing VoIP : keeping your VoIP network safe / / Regis J. (Bud) Bates ; acquiring editor, Steve Elliot ; designer, Greg Harris |
Autore | Bates Regis J., Bud |
Edizione | [1st edition] |
Pubbl/distr/stampa | Waltham, Massachusetts : , : Syngress, , 2015 |
Descrizione fisica | 1 online resource (222 p.) |
Disciplina | 004.695 |
Soggetto topico | Internet telephony - Security measures |
ISBN | 0-12-417122-2 |
Formato | Materiale a stampa ![]() |
Livello bibliografico | Monografia |
Lingua di pubblicazione | eng |
Nota di contenuto |
Cover; Title Page; Copyright Page; Contents; Technical editor biography; About the author; Expertise; Publications; Books; Articles; Acknowledgments; Chapter 1 - Introduction; Securing Voice over Internet Protocol (VoIP): keeping your network safe; History of telephony; History of the Internet Protocol; What goes around comes around; VoIP network and potential problems; The benefits of VoIP; Some initial thoughts on VoIP; What are the reasons for the VoIP hacking attempts; The need for VoIP security; Need for security and causes; Technology; Policy; Terms and attacks; Other vulnerabilities
What is at riskCan a call be eavesdropped?; There is no Holy Grail out there; Summary; Chapter 2 - Policies; What is the problem?; The call control channel - hijacking; Softphone issues; Denial-of-service attacks; Security concerns; Security policy needs; Vulnerability detection and auditing; Is the system vulnerable?; Chapter 3 - VoIP virtual private networks (VPNs); Virtual private networks (VPNs) and encryption; What is a VPN?; The possible VPN solutions; What a VPN can offer; What everyone expects from securing VoIP; What is the impact?; Creating the VPN; IPSec used for site-to-site VPNs Disadvantages of IPSec VPN site-to-site tunnelsSummary; Chapter 4 - Cryptography solutions; Cryptography solutions; What is cryptography and encryption?; Early ciphers used; Digital signatures; Leads to a public key infrastructure; X.509 certificates; Digital certificate servers; Installing certificates on the devices; Summary; Chapter 5 - Authentication; Authentication defined; Details of 802.1x authentication; Use a VoIP-enabled firewall; Use 802.1X authentication for IP phones; Attacking VoIP Authentication; Encrypt the traffic; Authentication on wireless networks; Summary Chapter 6 - Other protocolsOther protocols; Overview of Real-Time Transport Protocol and Real-Time Transport Control Protocol (RTCP); RTCP; Function of Secure RTP; Enter SRTP and SRTCP; SRTP framework; Secure RTP using ZRTP; Mikey; Modes of MIKEY; Preshared key transfer; Public key transfer; Public key with Diffie-Hellman exchange; Transport protocols; Signaling: Session Initiation Protocol; Attacks on SIP; Denial of service; BYE; Authentication; Secure SIP; Transport and network layer security; Summary; Chapter 7 - The business case for securing VoIP; Before we start Overview of the RFC 2196Internal issues; Toll fraud - a big threat; Summary; Chapter 8 - Approaches to VoIP security; Before we start; Build it in layers; Some best practices for infrastructure security; Integrating network security; Additional thoughts and items; Registration spoofing; Summary; Chapter 9 - Final thoughts; Before we start; What we have already covered; Vendor issues; Controlling the risks; PBX best practices; Summary; Index |
Record Nr. | UNINA-9910787921403321 |
Bates Regis J., Bud
![]() |
||
Waltham, Massachusetts : , : Syngress, , 2015 | ||
![]() | ||
Lo trovi qui: Univ. Federico II | ||
|
Securing VoIP : keeping your VoIP network safe / / Regis J. (Bud) Bates ; acquiring editor, Steve Elliot ; designer, Greg Harris |
Autore | Bates Regis J., Bud |
Edizione | [1st edition] |
Pubbl/distr/stampa | Waltham, Massachusetts : , : Syngress, , 2015 |
Descrizione fisica | 1 online resource (222 p.) |
Disciplina | 004.695 |
Soggetto topico | Internet telephony - Security measures |
ISBN | 0-12-417122-2 |
Formato | Materiale a stampa ![]() |
Livello bibliografico | Monografia |
Lingua di pubblicazione | eng |
Nota di contenuto |
Cover; Title Page; Copyright Page; Contents; Technical editor biography; About the author; Expertise; Publications; Books; Articles; Acknowledgments; Chapter 1 - Introduction; Securing Voice over Internet Protocol (VoIP): keeping your network safe; History of telephony; History of the Internet Protocol; What goes around comes around; VoIP network and potential problems; The benefits of VoIP; Some initial thoughts on VoIP; What are the reasons for the VoIP hacking attempts; The need for VoIP security; Need for security and causes; Technology; Policy; Terms and attacks; Other vulnerabilities
What is at riskCan a call be eavesdropped?; There is no Holy Grail out there; Summary; Chapter 2 - Policies; What is the problem?; The call control channel - hijacking; Softphone issues; Denial-of-service attacks; Security concerns; Security policy needs; Vulnerability detection and auditing; Is the system vulnerable?; Chapter 3 - VoIP virtual private networks (VPNs); Virtual private networks (VPNs) and encryption; What is a VPN?; The possible VPN solutions; What a VPN can offer; What everyone expects from securing VoIP; What is the impact?; Creating the VPN; IPSec used for site-to-site VPNs Disadvantages of IPSec VPN site-to-site tunnelsSummary; Chapter 4 - Cryptography solutions; Cryptography solutions; What is cryptography and encryption?; Early ciphers used; Digital signatures; Leads to a public key infrastructure; X.509 certificates; Digital certificate servers; Installing certificates on the devices; Summary; Chapter 5 - Authentication; Authentication defined; Details of 802.1x authentication; Use a VoIP-enabled firewall; Use 802.1X authentication for IP phones; Attacking VoIP Authentication; Encrypt the traffic; Authentication on wireless networks; Summary Chapter 6 - Other protocolsOther protocols; Overview of Real-Time Transport Protocol and Real-Time Transport Control Protocol (RTCP); RTCP; Function of Secure RTP; Enter SRTP and SRTCP; SRTP framework; Secure RTP using ZRTP; Mikey; Modes of MIKEY; Preshared key transfer; Public key transfer; Public key with Diffie-Hellman exchange; Transport protocols; Signaling: Session Initiation Protocol; Attacks on SIP; Denial of service; BYE; Authentication; Secure SIP; Transport and network layer security; Summary; Chapter 7 - The business case for securing VoIP; Before we start Overview of the RFC 2196Internal issues; Toll fraud - a big threat; Summary; Chapter 8 - Approaches to VoIP security; Before we start; Build it in layers; Some best practices for infrastructure security; Integrating network security; Additional thoughts and items; Registration spoofing; Summary; Chapter 9 - Final thoughts; Before we start; What we have already covered; Vendor issues; Controlling the risks; PBX best practices; Summary; Index |
Record Nr. | UNINA-9910807049603321 |
Bates Regis J., Bud
![]() |
||
Waltham, Massachusetts : , : Syngress, , 2015 | ||
![]() | ||
Lo trovi qui: Univ. Federico II | ||
|
VoIP and PBX Security and Forensics : A Practical Approach / / by Iosif I. Androulidakis |
Autore | Androulidakis Iosif I |
Edizione | [2nd ed. 2016.] |
Pubbl/distr/stampa | Cham : , : Springer International Publishing : , : Imprint : Springer, , 2016 |
Descrizione fisica | 1 online resource (112 p.) |
Disciplina | 004.695 |
Collana | SpringerBriefs in Electrical and Computer Engineering |
Soggetto topico |
Electrical engineering
Computer security System safety Forensic science Communications Engineering, Networks Systems and Data Security Security Science and Technology Forensic Science |
ISBN | 3-319-29721-X |
Formato | Materiale a stampa ![]() |
Livello bibliografico | Monografia |
Lingua di pubblicazione | eng |
Nota di contenuto | Introduction -- Confidentiality, Integrity, and Availability Threats in PBXs -- PBX Technical Details -- PBX Security -- PBX Forensics -- Conclusion. |
Record Nr. | UNINA-9910254251003321 |
Androulidakis Iosif I
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Cham : , : Springer International Publishing : , : Imprint : Springer, , 2016 | ||
![]() | ||
Lo trovi qui: Univ. Federico II | ||
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VoIP and unified communications : internet telephony and the future voice network / / William A. Flanagan |
Autore | Flanagan William A. |
Edizione | [1st edition] |
Pubbl/distr/stampa | Hoboken, New Jersey : , : Wiley, , c2012 |
Descrizione fisica | 1 online resource (316 p.) |
Disciplina |
004.695
621.385 |
Soggetto topico |
Internet telephony
Multimedia communications |
ISBN |
1-280-59175-7
9786613621580 1-118-16603-5 1-118-16604-3 1-118-16601-9 |
Classificazione | TEC041000 |
Formato | Materiale a stampa ![]() |
Livello bibliografico | Monografia |
Lingua di pubblicazione | eng |
Nota di contenuto |
-- Preface xiii -- Acknowledgments xv -- 1 IP Technology Disrupts Voice Telephony 1 -- 1.1 Introduction to the Public Switched Telephone Network 1 -- 1.2 The Digital PSTN 2 -- 1.3 The Packet Revolution in Telephony 8 -- 1.3.1 Summary of Packet Switching 9 -- 1.3.2 Link Capacity: TDM versus Packets 11 -- 1.3.3 VoIP and “The Cloud” 13 -- IN SHORT: Reading Network Drawings 14 -- 2 Traditional Telephones Still Set Expectations 17 -- 2.1 Availability: How the Bell System Ensured Service 18 -- 2.2 Call Completion 19 -- 2.3 Sound Quality: Encoding for Recognizable Voices 20 -- 2.4 Low Latency 23 -- 2.5 Call Setup Delays 24 -- 2.6 Impairments Controlled: Echo, Singing, Distortion, Noise 25 -- 3 From Circuits to Packets 27 -- 3.1 Data and Signaling Preceded Voice 27 -- 3.1.1 X.25 Packet Data Service 27 -- 3.1.2 SS7: PSTN Signaling on Packets 28 -- 3.1.3 ISDN 29 -- 3.2 Putting Voice into Packets 30 -- 3.2.1 Voice Encoding 31 -- 3.2.2 Dicing and Splicing Voice Streams 32 -- 3.2.3 The Latency Budget 33 -- 4 Packet Transmission and Switching 37 -- 4.1 The Physical Layer: Transmission 39 -- IN SHORT: The Endian Wars 40 -- 4.2 Data Link Protocols 41 -- 4.3 IP, the Network Protocol 43 -- 4.4 Layer 4 Transport Protocols 47 -- 4.4.1 Transmission Control Protocol 47 -- 4.4.2 User Datagram Protocol 50 -- 4.4.3 Stream Control Transmission Protocol 51 -- 4.5 Higher Layer Processes 54 -- 4.5.1 RTP 54 -- 4.5.2 RTCP 57 -- 4.5.3 Multiplexing RTP and RTCP on One UDP Port 58 -- 4.5.4 RTP Mixers and Translators 59 -- 4.5.5 Layered Encoding 60 -- 4.5.6 Profiles for Audio and Video Conferences 60 -- 4.5.7 Security via Encryption 61 -- IN SHORT: Public Key Infrastructure (PKI) 62 -- 4.6 Saving Bandwidth 64 -- 4.6.1 Voice Compression 64 -- 4.6.2 Header Compression 66 -- 4.6.3 Silence Suppression, VAD 67 -- 4.6.4 Sub-Packet Multiplexing 69 -- 4.6.5 Protocol and Codec Selection 70 -- 4.7 Differences: Circuit versus Packet Switched 71 -- 4.7.1 Power to the Desktop Phone 71 -- 4.7.2 Phone as Computer and Computer as Phone 72.
4.7.3 Length of a Phone Line 72 -- 4.7.4 Scaling to Large Size 75 -- 4.7.5 Software Ownership and Licenses 75 -- 5 VoIP Signaling and Call Processing 77 -- 5.1 What Packet Voice and UC Systems Share 78 -- 5.2 Session Initiation Protocol (SIP) 80 -- 5.2.1 SIP Architecture 81 -- 5.2.2 SIP Messages 88 -- 5.2.3 SIP Header Fields and Behaviors 94 -- 5.3 Session Description Protocol 101 -- IN SHORT: ABNF 104 -- 5.4 Media Gateway Control Protocol 107 -- 5.4.1 MGW Functions 107 -- 5.4.2 MGW Connection Model 110 -- 5.4.3 Megaco Procedures 112 -- 5.4.4 Megaco Details 115 -- 5.4.5 Signaling Conversion 119 -- 5.4.6 Voice Transcoding 119 -- 5.5 H.323 120 -- 5.5.1 H.323 Architecture 121 -- 5.5.2 Gatekeeper 123 -- 5.5.3 Gateway 126 -- 5.5.4 Terminal 126 -- 5.5.5 Multipoint Control Unit 127 -- 5.5.6 Call Procedures 128 -- 5.6 Directory Services 134 -- 5.6.1 Domain Name Service (DNS) 134 -- 5.6.2 ENUM 135 -- 6 VoIP and Unified Communications Define the Future 139 -- 6.1 Voice as Before, with Additions 139 -- 6.2 Legacy Services to Keep and Improve with VoIP 140 -- 6.2.1 Flexible Call Routing and 800 Numbers 141 -- 6.2.2 Call on Hold 141 -- 6.2.3 Call Transfer 142 -- 6.2.4 Call Forwarding 142 -- 6.2.5 Audio Conferencing 142 -- 6.2.6 Video Conferencing 143 -- 6.2.7 Local Number Portability 144 -- 6.2.8 Direct Inward Dialing, Dialed Number Indication 144 -- 6.2.9 CallMessage Waiting 145 -- 6.2.10 Call Recording 146 -- 6.2.11 Emergency Calling (E911) 146 -- 6.2.12 Tracking IP Phone Locations for E911 150 -- 6.3 Facsimile Transmission 153 -- 6.3.1 Facsimile on the PSTN 153 -- 6.3.2 Real-Time Fax over IP: Fax Relay or T.38 155 -- 6.3.3 Store-and-Forward Fax Handling 160 -- 6.3.4 IP Faxing over the PSTN 161 -- 6.4 Phone Features Added with VoIPUC 162 -- 6.4.1 Presence 163 -- 6.4.2 Forking 163 -- 6.4.3 Voicemail1/4eMail 163 -- 6.4.4 SMS Integration 164 -- 6.4.5 Instant Messaging 165 -- 6.4.6 Webinar Broadcasts 168 -- 6.4.7 Telepresence 168 -- 6.4.8 More UC Features to Consider 168 -- 7 How VoIP and UC Impact the Network 171. 7.1 Space, Power, and Cooling 171 -- 7.2 Priority for Voice, Video, Fax Packets 172 -- 7.3 Packets per Second 174 -- 7.4 Bandwidth 174 -- 7.5 Security Issues 175 -- 7.5.1 Eavesdropping and vLAN Hopping 176 -- 7.5.2 Access Controls for Users and Connections 176 -- 7.5.3 Modems 177 -- 7.5.4 DNS Cache Poisoning 177 -- IN SHORT: Earliest Instance of DNS Cache Poisoning 179 -- 7.5.5 Toll Fraud 179 -- 7.5.6 Pay-per-Call Scams 179 -- 7.5.7 Vishing 180 -- 7.5.8 SIP ScanningSPIT 180 -- 7.5.9 Opening the Firewall to Incoming Voice 181 -- 7.6 First Migration Steps While Keeping Legacy Equipment 181 -- 7.6.1 Circuit-Switched PBX 182 -- 7.6.2 Digital Phones 182 -- 7.6.3 Analog Phones and FX Service 183 -- 7.6.4 Facsimile Machines 184 -- 7.6.5 Modems 185 -- 8 Interconnections to Global Services 187 -- 8.1 Media Gateways 188 -- 8.2 SIP Trunking 192 -- 8.3 Operating VoIP Across Network Address Translation 196 -- 8.3.1 Failures of SIP, SDP (Signaling) 199 -- 8.3.2 Failures of RTP (Media) 199 -- 8.3.3 Solutions 200 -- 8.3.4 STUN: Session Traversal Utilities for NAT 201 -- 8.3.5 TURN: Traversal Using Relays around NAT 204 -- 8.3.6 ICE: Interactive Connectivity Establishment 206 -- 8.4 Session Border Controller 207 -- 8.4.1 Enterprise SBC 209 -- 8.4.2 Carrier SBC 210 -- 8.5 Supporting Multiple-Carrier Connections 212 -- 8.6 Mobility and Wireless Access 213 -- 8.6.1 VoIP on Wireless LANsWi-Fi 213 -- 8.6.2 Integration of Wi-Fi and Cellular Services 214 -- 8.6.3 Packet Voice on Mobile Broadband: WiMAX, LTE 214 -- 8.6.4 Radio over VoIP 215 -- IN SHORT: E&M Voice Signaling 216 -- 9 Network Management for VoIP and UC 217 -- 9.1 Starting Right 218 -- 9.1.1 Acceptance Testing 219 -- 9.1.2 Configuration Management and Governance 220 -- 9.1.3 Privilege Setting 220 -- 9.2 Continuous Monitoring and Management 221 -- 9.2.1 NMS Software 222 -- 9.2.2 Simple Network Management Protocol 223 -- 9.2.3 Web Interface 224 -- 9.2.4 Server Logging 224 -- 9.2.5 Software Maintenance 225 -- 9.2.6 Quality of ServiceExperience Monitoring 225. 9.2.7 Validate Adjustments and Optimization 226 -- 9.3 Troubleshooting and Repair 226 -- 9.3.1 Methods 226 -- 9.3.2 Software Tools 228 -- 9.3.3 Test Instruments 229 -- 10 Cost Analysis and Payback Calculation 231 -- 11 Examples of Hardware and Software 237 -- 11.1 IP Phones 237 -- 11.2 Gateways 240 -- 11.3 Session Border Controllers 242 -- 11.4 Call-Switching Servers 244 -- 11.4.1 IP PBX 246 -- 11.4.2 Conference BridgesControllers 248 -- 11.4.3 Call Recorder 250 -- 11.5 Hosted VoIPUC Service 251 -- 11.6 Management SystemsWorkstations 252 -- 12 Appendixes 253 -- 12.1 Acronyms and Definitions 253 -- 12.2 Reference Documents 268 -- 12.2.1 RFCs 268 -- 12.2.2 ITU Recommendations 272 -- 12.2.3 Other Sources 272 -- 12.3 Message and Error Codes 274 -- Index 277. |
Record Nr. | UNINA-9910141343203321 |
Flanagan William A.
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Hoboken, New Jersey : , : Wiley, , c2012 | ||
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Lo trovi qui: Univ. Federico II | ||
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VoIP and unified communications : internet telephony and the future voice network / / William A. Flanagan |
Autore | Flanagan William A |
Edizione | [1st edition] |
Pubbl/distr/stampa | Hoboken, N.J., : Wiley, c2012 |
Descrizione fisica | 1 online resource (316 p.) |
Disciplina |
004.695
621.385 |
Soggetto topico |
Internet telephony
Multimedia communications |
ISBN |
1-280-59175-7
9786613621580 1-118-16603-5 1-118-16604-3 1-118-16601-9 |
Classificazione | TEC041000 |
Formato | Materiale a stampa ![]() |
Livello bibliografico | Monografia |
Lingua di pubblicazione | eng |
Nota di contenuto |
-- Preface xiii -- Acknowledgments xv -- 1 IP Technology Disrupts Voice Telephony 1 -- 1.1 Introduction to the Public Switched Telephone Network 1 -- 1.2 The Digital PSTN 2 -- 1.3 The Packet Revolution in Telephony 8 -- 1.3.1 Summary of Packet Switching 9 -- 1.3.2 Link Capacity: TDM versus Packets 11 -- 1.3.3 VoIP and “The Cloud” 13 -- IN SHORT: Reading Network Drawings 14 -- 2 Traditional Telephones Still Set Expectations 17 -- 2.1 Availability: How the Bell System Ensured Service 18 -- 2.2 Call Completion 19 -- 2.3 Sound Quality: Encoding for Recognizable Voices 20 -- 2.4 Low Latency 23 -- 2.5 Call Setup Delays 24 -- 2.6 Impairments Controlled: Echo, Singing, Distortion, Noise 25 -- 3 From Circuits to Packets 27 -- 3.1 Data and Signaling Preceded Voice 27 -- 3.1.1 X.25 Packet Data Service 27 -- 3.1.2 SS7: PSTN Signaling on Packets 28 -- 3.1.3 ISDN 29 -- 3.2 Putting Voice into Packets 30 -- 3.2.1 Voice Encoding 31 -- 3.2.2 Dicing and Splicing Voice Streams 32 -- 3.2.3 The Latency Budget 33 -- 4 Packet Transmission and Switching 37 -- 4.1 The Physical Layer: Transmission 39 -- IN SHORT: The Endian Wars 40 -- 4.2 Data Link Protocols 41 -- 4.3 IP, the Network Protocol 43 -- 4.4 Layer 4 Transport Protocols 47 -- 4.4.1 Transmission Control Protocol 47 -- 4.4.2 User Datagram Protocol 50 -- 4.4.3 Stream Control Transmission Protocol 51 -- 4.5 Higher Layer Processes 54 -- 4.5.1 RTP 54 -- 4.5.2 RTCP 57 -- 4.5.3 Multiplexing RTP and RTCP on One UDP Port 58 -- 4.5.4 RTP Mixers and Translators 59 -- 4.5.5 Layered Encoding 60 -- 4.5.6 Profiles for Audio and Video Conferences 60 -- 4.5.7 Security via Encryption 61 -- IN SHORT: Public Key Infrastructure (PKI) 62 -- 4.6 Saving Bandwidth 64 -- 4.6.1 Voice Compression 64 -- 4.6.2 Header Compression 66 -- 4.6.3 Silence Suppression, VAD 67 -- 4.6.4 Sub-Packet Multiplexing 69 -- 4.6.5 Protocol and Codec Selection 70 -- 4.7 Differences: Circuit versus Packet Switched 71 -- 4.7.1 Power to the Desktop Phone 71 -- 4.7.2 Phone as Computer and Computer as Phone 72.
4.7.3 Length of a Phone Line 72 -- 4.7.4 Scaling to Large Size 75 -- 4.7.5 Software Ownership and Licenses 75 -- 5 VoIP Signaling and Call Processing 77 -- 5.1 What Packet Voice and UC Systems Share 78 -- 5.2 Session Initiation Protocol (SIP) 80 -- 5.2.1 SIP Architecture 81 -- 5.2.2 SIP Messages 88 -- 5.2.3 SIP Header Fields and Behaviors 94 -- 5.3 Session Description Protocol 101 -- IN SHORT: ABNF 104 -- 5.4 Media Gateway Control Protocol 107 -- 5.4.1 MGW Functions 107 -- 5.4.2 MGW Connection Model 110 -- 5.4.3 Megaco Procedures 112 -- 5.4.4 Megaco Details 115 -- 5.4.5 Signaling Conversion 119 -- 5.4.6 Voice Transcoding 119 -- 5.5 H.323 120 -- 5.5.1 H.323 Architecture 121 -- 5.5.2 Gatekeeper 123 -- 5.5.3 Gateway 126 -- 5.5.4 Terminal 126 -- 5.5.5 Multipoint Control Unit 127 -- 5.5.6 Call Procedures 128 -- 5.6 Directory Services 134 -- 5.6.1 Domain Name Service (DNS) 134 -- 5.6.2 ENUM 135 -- 6 VoIP and Unified Communications Define the Future 139 -- 6.1 Voice as Before, with Additions 139 -- 6.2 Legacy Services to Keep and Improve with VoIP 140 -- 6.2.1 Flexible Call Routing and 800 Numbers 141 -- 6.2.2 Call on Hold 141 -- 6.2.3 Call Transfer 142 -- 6.2.4 Call Forwarding 142 -- 6.2.5 Audio Conferencing 142 -- 6.2.6 Video Conferencing 143 -- 6.2.7 Local Number Portability 144 -- 6.2.8 Direct Inward Dialing, Dialed Number Indication 144 -- 6.2.9 CallMessage Waiting 145 -- 6.2.10 Call Recording 146 -- 6.2.11 Emergency Calling (E911) 146 -- 6.2.12 Tracking IP Phone Locations for E911 150 -- 6.3 Facsimile Transmission 153 -- 6.3.1 Facsimile on the PSTN 153 -- 6.3.2 Real-Time Fax over IP: Fax Relay or T.38 155 -- 6.3.3 Store-and-Forward Fax Handling 160 -- 6.3.4 IP Faxing over the PSTN 161 -- 6.4 Phone Features Added with VoIPUC 162 -- 6.4.1 Presence 163 -- 6.4.2 Forking 163 -- 6.4.3 Voicemail1/4eMail 163 -- 6.4.4 SMS Integration 164 -- 6.4.5 Instant Messaging 165 -- 6.4.6 Webinar Broadcasts 168 -- 6.4.7 Telepresence 168 -- 6.4.8 More UC Features to Consider 168 -- 7 How VoIP and UC Impact the Network 171. 7.1 Space, Power, and Cooling 171 -- 7.2 Priority for Voice, Video, Fax Packets 172 -- 7.3 Packets per Second 174 -- 7.4 Bandwidth 174 -- 7.5 Security Issues 175 -- 7.5.1 Eavesdropping and vLAN Hopping 176 -- 7.5.2 Access Controls for Users and Connections 176 -- 7.5.3 Modems 177 -- 7.5.4 DNS Cache Poisoning 177 -- IN SHORT: Earliest Instance of DNS Cache Poisoning 179 -- 7.5.5 Toll Fraud 179 -- 7.5.6 Pay-per-Call Scams 179 -- 7.5.7 Vishing 180 -- 7.5.8 SIP ScanningSPIT 180 -- 7.5.9 Opening the Firewall to Incoming Voice 181 -- 7.6 First Migration Steps While Keeping Legacy Equipment 181 -- 7.6.1 Circuit-Switched PBX 182 -- 7.6.2 Digital Phones 182 -- 7.6.3 Analog Phones and FX Service 183 -- 7.6.4 Facsimile Machines 184 -- 7.6.5 Modems 185 -- 8 Interconnections to Global Services 187 -- 8.1 Media Gateways 188 -- 8.2 SIP Trunking 192 -- 8.3 Operating VoIP Across Network Address Translation 196 -- 8.3.1 Failures of SIP, SDP (Signaling) 199 -- 8.3.2 Failures of RTP (Media) 199 -- 8.3.3 Solutions 200 -- 8.3.4 STUN: Session Traversal Utilities for NAT 201 -- 8.3.5 TURN: Traversal Using Relays around NAT 204 -- 8.3.6 ICE: Interactive Connectivity Establishment 206 -- 8.4 Session Border Controller 207 -- 8.4.1 Enterprise SBC 209 -- 8.4.2 Carrier SBC 210 -- 8.5 Supporting Multiple-Carrier Connections 212 -- 8.6 Mobility and Wireless Access 213 -- 8.6.1 VoIP on Wireless LANsWi-Fi 213 -- 8.6.2 Integration of Wi-Fi and Cellular Services 214 -- 8.6.3 Packet Voice on Mobile Broadband: WiMAX, LTE 214 -- 8.6.4 Radio over VoIP 215 -- IN SHORT: E&M Voice Signaling 216 -- 9 Network Management for VoIP and UC 217 -- 9.1 Starting Right 218 -- 9.1.1 Acceptance Testing 219 -- 9.1.2 Configuration Management and Governance 220 -- 9.1.3 Privilege Setting 220 -- 9.2 Continuous Monitoring and Management 221 -- 9.2.1 NMS Software 222 -- 9.2.2 Simple Network Management Protocol 223 -- 9.2.3 Web Interface 224 -- 9.2.4 Server Logging 224 -- 9.2.5 Software Maintenance 225 -- 9.2.6 Quality of ServiceExperience Monitoring 225. 9.2.7 Validate Adjustments and Optimization 226 -- 9.3 Troubleshooting and Repair 226 -- 9.3.1 Methods 226 -- 9.3.2 Software Tools 228 -- 9.3.3 Test Instruments 229 -- 10 Cost Analysis and Payback Calculation 231 -- 11 Examples of Hardware and Software 237 -- 11.1 IP Phones 237 -- 11.2 Gateways 240 -- 11.3 Session Border Controllers 242 -- 11.4 Call-Switching Servers 244 -- 11.4.1 IP PBX 246 -- 11.4.2 Conference BridgesControllers 248 -- 11.4.3 Call Recorder 250 -- 11.5 Hosted VoIPUC Service 251 -- 11.6 Management SystemsWorkstations 252 -- 12 Appendixes 253 -- 12.1 Acronyms and Definitions 253 -- 12.2 Reference Documents 268 -- 12.2.1 RFCs 268 -- 12.2.2 ITU Recommendations 272 -- 12.2.3 Other Sources 272 -- 12.3 Message and Error Codes 274 -- Index 277. |
Altri titoli varianti | Understanding VoIP and unified communications and the future voice network |
Record Nr. | UNINA-9910825080303321 |
Flanagan William A
![]() |
||
Hoboken, N.J., : Wiley, c2012 | ||
![]() | ||
Lo trovi qui: Univ. Federico II | ||
|