Skype is a peer-to-peer VoIP application that allows voice calls, text messaging, file transfers and video conferencing. It uses a hybrid peer-to-peer and client-server model where each user acts as a peer on the network. When users cannot directly connect due to firewalls or NAT, Skype uses relaying through supernodes to establish the connection. Relaying involves communicating through an intermediary node when a direct connection is not possible. Skype employs strong encryption and utilizes codecs to digitally encode voice for transmission over the internet.
Session Initiation Protocol (SIP) is an application layer signaling protocol used to establish, modify, and terminate multimedia sessions over the Internet. SIP allows users to initiate and manage telephone calls, video conferences, messaging, and other multimedia sessions. It can be used with other protocols like SDP, RTP, and RTCP to build a complete multimedia architecture. SIP establishes sessions through proxy servers, redirect servers, and registrars, and uses response codes to indicate session status. A basic SIP call flow involves an INVITE request, provisional responses, a final 200 OK response, media transmission via RTP, and termination with a BYE request.
SIP (Session Initiation Protocol) - Study NotesOxfordCambridge
This document provides an overview of Session Initiation Protocol (SIP) including its components and standards. SIP is an application layer control protocol that establishes, modifies and terminates multimedia sessions and calls. It can be used for voice and video calls over Internet Protocol (IP). The key components of SIP include user agents (UAs), proxy servers, redirect servers, registrar servers and location servers. UAs contain user agent clients (UACs) that initiate requests and user agent servers (UASs) that respond to requests. Common UAs are IP phones and gateways.
SIP is a protocol for establishing multimedia sessions over IP networks. It originated from work in the 1990s on protocols like SCIP and SIP drafts. SIP eventually became standardized as RFC 3261 and is now widely used for voice and video calling. Cisco supports SIP in products like Cisco Unified Communications Manager, Cisco Unified Border Element, and Cisco Unified Presence to enable VoIP calling and integration between SIP and other protocols. The future of SIP includes more peer-to-peer implementations and using presence as a foundation for new services.
SIP (Session Initiation Protocol) is a signaling protocol used to create, manage and terminate sessions in an IP based network. This course is for beginners and aims to give a brief introduction to SIP before one ventures into the long RFC documents.
Performance Analysis of VoIP by Communicating Two Systems IOSR Journals
This document discusses performance analysis of Voice over IP (VoIP) communication between two systems. It introduces VoIP technology and explores problems that can occur when sending voice and data packets over different windows. The proposed work evaluates performance of open source VoIP software tools like Linphone for clients, OpenSIPS for the server, and NS-2 for network traffic analysis. Two systems are configured as clients and one as the server to simulate a VoIP call between Client A and B routed through the server.
Kundan Singh provides an overview of Session Initiation Protocol (SIP) for networking experts. He has a PhD in Internet telephony from Columbia University and has worked on SIP implementations and open source projects since 1999. The document summarizes what attendees will learn about SIP including its history, standards, architecture, protocols, call flows, challenges, and hands-on exercises. It also provides background on Kundan's expertise in areas like VoIP, audio coding, and real-time protocols.
The document provides an overview of transporting voice over IP networks. It describes the layers of the OSI model and how the IP protocol suite, including IP, TCP, UDP, RTP, and RTCP, operate at each layer to transport data packets. IP provides best-effort delivery of packets using routing protocols. TCP provides reliable, in-order delivery, while UDP is unreliable but lower latency. RTP carries encoded voice/video payload and provides timestamps/sequence numbers, while RTCP exchanges quality feedback. Signaling protocols negotiate payload formats to allow different encoding schemes.
Session Initiation Protocol (SIP) is an application layer protocol for setting up and managing multimedia communication sessions over IP networks. It allows users to initiate, modify and terminate multimedia sessions that include voice, video and messaging applications. SIP supports mobility through proxy servers that can forward calls to a user's current location. Common security threats to SIP include registration hijacking, message modification and denial of service attacks. Recommended security mechanisms include TLS for hop-by-hop security, S/MIME for end-to-end encryption, and digest authentication.
Session Initiation Protocol (SIP) is an application layer signaling protocol used to establish, modify, and terminate multimedia sessions over the Internet. SIP allows users to initiate and manage telephone calls, video conferences, messaging, and other multimedia sessions. It can be used with other protocols like SDP, RTP, and RTCP to build a complete multimedia architecture. SIP establishes sessions through proxy servers, redirect servers, and registrars, and uses response codes to indicate session status. A basic SIP call flow involves an INVITE request, provisional responses, a final 200 OK response, media transmission via RTP, and termination with a BYE request.
SIP (Session Initiation Protocol) - Study NotesOxfordCambridge
This document provides an overview of Session Initiation Protocol (SIP) including its components and standards. SIP is an application layer control protocol that establishes, modifies and terminates multimedia sessions and calls. It can be used for voice and video calls over Internet Protocol (IP). The key components of SIP include user agents (UAs), proxy servers, redirect servers, registrar servers and location servers. UAs contain user agent clients (UACs) that initiate requests and user agent servers (UASs) that respond to requests. Common UAs are IP phones and gateways.
SIP is a protocol for establishing multimedia sessions over IP networks. It originated from work in the 1990s on protocols like SCIP and SIP drafts. SIP eventually became standardized as RFC 3261 and is now widely used for voice and video calling. Cisco supports SIP in products like Cisco Unified Communications Manager, Cisco Unified Border Element, and Cisco Unified Presence to enable VoIP calling and integration between SIP and other protocols. The future of SIP includes more peer-to-peer implementations and using presence as a foundation for new services.
SIP (Session Initiation Protocol) is a signaling protocol used to create, manage and terminate sessions in an IP based network. This course is for beginners and aims to give a brief introduction to SIP before one ventures into the long RFC documents.
Performance Analysis of VoIP by Communicating Two Systems IOSR Journals
This document discusses performance analysis of Voice over IP (VoIP) communication between two systems. It introduces VoIP technology and explores problems that can occur when sending voice and data packets over different windows. The proposed work evaluates performance of open source VoIP software tools like Linphone for clients, OpenSIPS for the server, and NS-2 for network traffic analysis. Two systems are configured as clients and one as the server to simulate a VoIP call between Client A and B routed through the server.
Kundan Singh provides an overview of Session Initiation Protocol (SIP) for networking experts. He has a PhD in Internet telephony from Columbia University and has worked on SIP implementations and open source projects since 1999. The document summarizes what attendees will learn about SIP including its history, standards, architecture, protocols, call flows, challenges, and hands-on exercises. It also provides background on Kundan's expertise in areas like VoIP, audio coding, and real-time protocols.
The document provides an overview of transporting voice over IP networks. It describes the layers of the OSI model and how the IP protocol suite, including IP, TCP, UDP, RTP, and RTCP, operate at each layer to transport data packets. IP provides best-effort delivery of packets using routing protocols. TCP provides reliable, in-order delivery, while UDP is unreliable but lower latency. RTP carries encoded voice/video payload and provides timestamps/sequence numbers, while RTCP exchanges quality feedback. Signaling protocols negotiate payload formats to allow different encoding schemes.
Session Initiation Protocol (SIP) is an application layer protocol for setting up and managing multimedia communication sessions over IP networks. It allows users to initiate, modify and terminate multimedia sessions that include voice, video and messaging applications. SIP supports mobility through proxy servers that can forward calls to a user's current location. Common security threats to SIP include registration hijacking, message modification and denial of service attacks. Recommended security mechanisms include TLS for hop-by-hop security, S/MIME for end-to-end encryption, and digest authentication.
VoIP uses packet networks to carry voice calls in addition to data. It works by converting analog voice signals to digital data packets which are transmitted over IP networks and reconverted to analog at the receiving end. Key components include IP phones, signaling servers, and protocols like SIP and H.323 which handle call setup and signaling. Quality of service for VoIP depends on factors like packet loss, delay, and jitter which can be managed through queuing and reserving bandwidth for voice traffic.
SIP routing relies on DNS lookups including NAPTR, SRV, and A records to determine the transport protocol, port, and IP address for a SIP request. The request URI and Route headers are used to determine where to send a request, while subsequent in-dialog requests follow the remote target and route set stored in the dialog state. Understanding SIP routing through RFC 3263 is essential for knowing how requests traverse the network and how responses follow the reverse path back.
The Session Initiation Protocol (SIP) is an application-layer signaling protocol used for establishing multimedia sessions over Internet Protocol (IP) networks, such as voice or video calls. SIP can be used to initiate a call, invite participants and manage a call. It defines several methods for call setup, maintenance and termination. Common SIP methods include INVITE for call initiation, ACK to acknowledge call setup, BYE to terminate a call, and REGISTER for registering location. SIP uses SDP for negotiating media capabilities and RTP for transporting media streams.
This document discusses advanced peer-to-peer SIP concepts and NAT traversal techniques. It describes how P2P services like call forwarding, voicemail, and music on hold can be implemented within a P2P overlay network or with help from peer nodes. It also covers challenges of NAT traversal and evaluates solutions like STUN, TURN, ICE, and HIP that aim to establish media connections through NATs and firewalls. Testing methods are important to prove the effectiveness of NAT traversal approaches.
1. The document introduces Session Initiation Protocol (SIP), explaining that it is an application layer signaling protocol for initiating, modifying, and terminating multimedia communication sessions over IP such as voice and video calls.
2. It describes why SIP is used, including for conferencing, distance learning, video conferencing, instant messaging, and voice calls. It also outlines the main components of a SIP network including user agents, proxies, and redirects servers.
3. The document provides an overview of how SIP works by outlining the signaling process for registration, call setup and teardown, redirection, and media routing between user agents.
IPv4 is the current version of the Internet Protocol but has limitations including a limited 32-bit address space that is nearly depleted, lacking built-in network security, and limited quality of service capabilities. IPv6 was developed to address these issues by using a larger 128-bit address space to avoid scarcity, incorporating IPsec to provide security, and improving quality of service and auto-configuration features. While IPv6 adoption is still growing, transitioning networks to be dual-stacked with both IPv4 and IPv6 ensures compatibility and avoids missing traffic from users on IPv6-only networks.
Join us for an introductory webinar on VoIP and learn:
- The fundamental principles of VoIP including RTP and SIP
- What voice metrics to measure and why they matter
- The different methods to monitor and troubleshoot VoIP
Www ccnav5 net_ccna_1_chapter_3_v5_0_exam_answers_2014Đồng Quốc Vương
This document provides answers to exam questions for CCNA 1 Chapter 3 v5.0. It addresses topics related to networking fundamentals including data encapsulation, protocols, network layers, and IP addressing. The questions cover concepts such as the OSI model, TCP/IP protocols, framing, and error control. Proprietary protocols are defined as those developed by private organizations to operate on any vendor hardware.
This document discusses various strategies for transitioning from IPv4 to IPv6. It begins by establishing that IPv4 addresses are running out due to the IANA and RIR pools being depleted. It then outlines three main strategies: doing nothing and remaining IPv4-only; extending the life of IPv4 through NAT or acquiring more addresses; and implementing IPv4/IPv6 coexistence techniques like dual-stack, 6rd, or large-scale NAT. Each strategy is defined and its advantages and disadvantages are discussed. The document provides guidance on which approaches may be suitable depending on an organization's needs and infrastructure capabilities.
This document discusses Voice over Internet Protocol (VoIP) technology which allows voice calls over broadband internet instead of regular phone lines. It details some key advantages of VoIP like lower taxes, portability, inclusion of advanced call features at no extra cost. The document also provides information on networking concepts like IP addressing, routing protocols, network applications and sample Cisco configurations for implementing VoIP in a university placement department and reception desk network.
MAF ICIMS™ Monitoring, Analytics & Reporting for Microsoft Teams and UC - glo...MAF InfoCom
MAF ICIMS™ is a reporting and analytics solution for Unified Communication and Collaboration (UC&C) platforms and other data sources such as Session Border Controllers (SBC’s), Gateways, Trading Platforms, Turrents & Dealer Boards. It allows you to gain valuable business and technical insights through its reports, daily dashboards and historical trend monitors. Its flexible, user defined nature means you tell the software what you want to see instead of the software dictating to you what you will see.
This document provides an overview of a project report on Voice over Internet Protocol (VoIP) submitted by two students, Amardeep Singh and Jaswinder Singh, at Chandigarh Engineering College in partial fulfillment of their B-Tech degree in Electronics and Communication Engineering. The report introduces VoIP technology, discusses software and hardware used in the project including Cisco routers and switches, and provides details on configuring an IP phone network with Cisco Call Manager Express including assigning IP addresses via DHCP and configuring phone directory numbers. Future enhancements discussed include integrating VoIP with wireless networks.
1.What is IP address
2.When & how it was devised
3.IPV4 Features & its functionality
4.Benefits of IPV4 & Devices supporting IPV4
5.Problems of IPV4 & What happened to IPV5
6.What led to IPV6
7.IPV6 Features & Functionality
8.Benefits of IPV6 & supporting devices
9.How transition from IPV4 to IPV6 will happen
10.Problems & challenges that are anticipated & Conclusion
ccna project on topic company infrastructurePrince Gautam
Prince Gautam submitted a presentation on CCNA that introduces CCNA and networking. It defines CCNA, describes the importance of networking for communication and resource sharing. It also summarizes different types of networking including LAN, MAN, WAN and common networking devices like hubs, switches, routers. The presentation further explains concepts like subnetting, supernetting, routing protocols like RIP, EIGRP, OSPF and basic router configuration.
This document discusses Mobile Internet Protocol (Mobile IP) and how it allows mobile devices to stay connected to the internet without changing their IP address as they move between different networks. It covers key topics such as:
- The basics of Mobile IP including definitions of terms like home agent, foreign agent, and care-of-address.
- How Mobile IP works including the process of discovering the care-of-address, registering with foreign agents, and tunneling packets to the mobile node's current location.
- Adaptations made to transport protocols like TCP to improve performance over wireless networks.
The document is a training report submitted by Sagar Shashank summarizing his CCNA training completed at CETPA INFOTECH PVT. LTD. from June 20 to July 31, 2018. It includes an introduction to networking concepts and protocols like IP, ICMP, routing protocols, switching technologies and OSI model. The report contains detailed explanations of topics covered during the training along with declarations and acknowledgements.
Network Interview Questions documents common networking concepts and protocols. It defines networking as interconnecting computers, describes bandwidth as the maximum data transfer rate of a connection, and VLAN as a logical grouping of ports on a switch. It also summarizes protocols like CIDR for IP address allocation, VLSM for subnetting, unicast for one-to-one transmission, multicast for one-to-many, and broadcast for one-to-all transmission. Key networking protocols like CDP, SNMP, OSPF, RIP, BGP, and PPPoE are also outlined.
VoIP uses packet networks to carry voice calls in addition to data. It works by converting analog voice signals to digital data packets which are transmitted over IP networks and reconverted to analog at the receiving end. Key components include IP phones, signaling servers, and protocols like SIP and H.323 which handle call setup and signaling. Quality of service for VoIP depends on factors like packet loss, delay, and jitter which can be managed through queuing and reserving bandwidth for voice traffic.
SIP routing relies on DNS lookups including NAPTR, SRV, and A records to determine the transport protocol, port, and IP address for a SIP request. The request URI and Route headers are used to determine where to send a request, while subsequent in-dialog requests follow the remote target and route set stored in the dialog state. Understanding SIP routing through RFC 3263 is essential for knowing how requests traverse the network and how responses follow the reverse path back.
The Session Initiation Protocol (SIP) is an application-layer signaling protocol used for establishing multimedia sessions over Internet Protocol (IP) networks, such as voice or video calls. SIP can be used to initiate a call, invite participants and manage a call. It defines several methods for call setup, maintenance and termination. Common SIP methods include INVITE for call initiation, ACK to acknowledge call setup, BYE to terminate a call, and REGISTER for registering location. SIP uses SDP for negotiating media capabilities and RTP for transporting media streams.
This document discusses advanced peer-to-peer SIP concepts and NAT traversal techniques. It describes how P2P services like call forwarding, voicemail, and music on hold can be implemented within a P2P overlay network or with help from peer nodes. It also covers challenges of NAT traversal and evaluates solutions like STUN, TURN, ICE, and HIP that aim to establish media connections through NATs and firewalls. Testing methods are important to prove the effectiveness of NAT traversal approaches.
1. The document introduces Session Initiation Protocol (SIP), explaining that it is an application layer signaling protocol for initiating, modifying, and terminating multimedia communication sessions over IP such as voice and video calls.
2. It describes why SIP is used, including for conferencing, distance learning, video conferencing, instant messaging, and voice calls. It also outlines the main components of a SIP network including user agents, proxies, and redirects servers.
3. The document provides an overview of how SIP works by outlining the signaling process for registration, call setup and teardown, redirection, and media routing between user agents.
IPv4 is the current version of the Internet Protocol but has limitations including a limited 32-bit address space that is nearly depleted, lacking built-in network security, and limited quality of service capabilities. IPv6 was developed to address these issues by using a larger 128-bit address space to avoid scarcity, incorporating IPsec to provide security, and improving quality of service and auto-configuration features. While IPv6 adoption is still growing, transitioning networks to be dual-stacked with both IPv4 and IPv6 ensures compatibility and avoids missing traffic from users on IPv6-only networks.
Join us for an introductory webinar on VoIP and learn:
- The fundamental principles of VoIP including RTP and SIP
- What voice metrics to measure and why they matter
- The different methods to monitor and troubleshoot VoIP
Www ccnav5 net_ccna_1_chapter_3_v5_0_exam_answers_2014Đồng Quốc Vương
This document provides answers to exam questions for CCNA 1 Chapter 3 v5.0. It addresses topics related to networking fundamentals including data encapsulation, protocols, network layers, and IP addressing. The questions cover concepts such as the OSI model, TCP/IP protocols, framing, and error control. Proprietary protocols are defined as those developed by private organizations to operate on any vendor hardware.
This document discusses various strategies for transitioning from IPv4 to IPv6. It begins by establishing that IPv4 addresses are running out due to the IANA and RIR pools being depleted. It then outlines three main strategies: doing nothing and remaining IPv4-only; extending the life of IPv4 through NAT or acquiring more addresses; and implementing IPv4/IPv6 coexistence techniques like dual-stack, 6rd, or large-scale NAT. Each strategy is defined and its advantages and disadvantages are discussed. The document provides guidance on which approaches may be suitable depending on an organization's needs and infrastructure capabilities.
This document discusses Voice over Internet Protocol (VoIP) technology which allows voice calls over broadband internet instead of regular phone lines. It details some key advantages of VoIP like lower taxes, portability, inclusion of advanced call features at no extra cost. The document also provides information on networking concepts like IP addressing, routing protocols, network applications and sample Cisco configurations for implementing VoIP in a university placement department and reception desk network.
MAF ICIMS™ Monitoring, Analytics & Reporting for Microsoft Teams and UC - glo...MAF InfoCom
MAF ICIMS™ is a reporting and analytics solution for Unified Communication and Collaboration (UC&C) platforms and other data sources such as Session Border Controllers (SBC’s), Gateways, Trading Platforms, Turrents & Dealer Boards. It allows you to gain valuable business and technical insights through its reports, daily dashboards and historical trend monitors. Its flexible, user defined nature means you tell the software what you want to see instead of the software dictating to you what you will see.
This document provides an overview of a project report on Voice over Internet Protocol (VoIP) submitted by two students, Amardeep Singh and Jaswinder Singh, at Chandigarh Engineering College in partial fulfillment of their B-Tech degree in Electronics and Communication Engineering. The report introduces VoIP technology, discusses software and hardware used in the project including Cisco routers and switches, and provides details on configuring an IP phone network with Cisco Call Manager Express including assigning IP addresses via DHCP and configuring phone directory numbers. Future enhancements discussed include integrating VoIP with wireless networks.
1.What is IP address
2.When & how it was devised
3.IPV4 Features & its functionality
4.Benefits of IPV4 & Devices supporting IPV4
5.Problems of IPV4 & What happened to IPV5
6.What led to IPV6
7.IPV6 Features & Functionality
8.Benefits of IPV6 & supporting devices
9.How transition from IPV4 to IPV6 will happen
10.Problems & challenges that are anticipated & Conclusion
ccna project on topic company infrastructurePrince Gautam
Prince Gautam submitted a presentation on CCNA that introduces CCNA and networking. It defines CCNA, describes the importance of networking for communication and resource sharing. It also summarizes different types of networking including LAN, MAN, WAN and common networking devices like hubs, switches, routers. The presentation further explains concepts like subnetting, supernetting, routing protocols like RIP, EIGRP, OSPF and basic router configuration.
This document discusses Mobile Internet Protocol (Mobile IP) and how it allows mobile devices to stay connected to the internet without changing their IP address as they move between different networks. It covers key topics such as:
- The basics of Mobile IP including definitions of terms like home agent, foreign agent, and care-of-address.
- How Mobile IP works including the process of discovering the care-of-address, registering with foreign agents, and tunneling packets to the mobile node's current location.
- Adaptations made to transport protocols like TCP to improve performance over wireless networks.
The document is a training report submitted by Sagar Shashank summarizing his CCNA training completed at CETPA INFOTECH PVT. LTD. from June 20 to July 31, 2018. It includes an introduction to networking concepts and protocols like IP, ICMP, routing protocols, switching technologies and OSI model. The report contains detailed explanations of topics covered during the training along with declarations and acknowledgements.
Network Interview Questions documents common networking concepts and protocols. It defines networking as interconnecting computers, describes bandwidth as the maximum data transfer rate of a connection, and VLAN as a logical grouping of ports on a switch. It also summarizes protocols like CIDR for IP address allocation, VLSM for subnetting, unicast for one-to-one transmission, multicast for one-to-many, and broadcast for one-to-all transmission. Key networking protocols like CDP, SNMP, OSPF, RIP, BGP, and PPPoE are also outlined.
This document analyzes the Skype peer-to-peer internet telephony protocol. It describes key aspects of the Skype network including that it uses an overlay peer-to-peer network of ordinary hosts and super nodes. The Skype login server authenticates users but does not store any other user information. Skype is able to traverse NATs and firewalls and uses wideband codecs to achieve good call quality over bandwidth as low as 32kbps. The document also outlines Skype's encryption methods and analyzes its functions like call establishment and media transfer through experimentation with the network traffic.
Skype is a software application that allows users to make voice and video calls over the internet. It was developed in 2002 and acquired by eBay in 2005. Key features include free calling between Skype users, low-cost calls to phones worldwide, and video conferencing. While it provides an inexpensive way to communicate globally, Skype has limitations such as requiring the called party to answer like a regular phone call and not allowing conference callers to see each other.
This document contains study of Peer to Peer Distributed system.Three Models of Distributed system.Such as Centralizes,Decentralized,Hybird Model and Pros and cons of these models. Skpye and Bit torrent architecture is also discussed.This tutorial can be very help full for those who are beginners.
Peer-to-Peer (P2P) has become a buzzword and file-sharing applications like Kazaa are very popular and account for a lot of Internet traffic nowadays. The emphasis of my talk will be on the evolution of P2P file-sharing and the technology behind the scenes. I also try to give examples how P2P can be used for other applications like Skype.
From the Un-Distinguished Lecture Series (http://ws.cs.ubc.ca/~udls/). The talk was given Feb. 16, 2007.
Peer-to-peer (P2P) networks are a type of computer network architecture where individuals form a loose group to share resources directly with others in the group without a centralized server. There are two main types of P2P network structures - unstructured and structured. Unstructured networks do not use algorithms to organize the network, while structured networks use algorithms to optimize routing. Popular applications of P2P networking include file sharing, media streaming, grid computing, instant messaging, and voice over internet protocol.
Skype uses a peer-to-peer architecture where each computer has equal capabilities as both a client and server. It allows for voice and video calls over the internet using protocols like SIP and RTP. Skype employs a hybrid P2P model with supernodes to decentralize functions like user search and directory sharing. Calls between Skype users can happen directly or through relays if behind firewalls, while connections to regular phones use SkypeOut gateways.
This document provides an overview of Skype and ICQ instant messaging protocols. It discusses the history and features of instant messaging technologies. It then describes the network architectures, encryption methods, and functions of Skype and the OSCAR protocol used by ICQ, including user login, search, call establishment, and message exchange.
This document provides an overview of VoIP services through a seminar presentation. It discusses how VoIP came about as an alternative to traditional circuit-switched telephony using the PSTN. VoIP allows carrying voice calls over an IP network by digitizing and packetizing voice streams using protocols like SIP and H.323. Some key benefits of VoIP include reduced costs, increased flexibility, and mobility. Popular VoIP service providers include Skype, while security poses ongoing challenges to VoIP adoption.
Analysis & Signature of Skype VoIP Session Traffic.pptxAhmedSohair2
The document analyzes Skype network traffic with the goal of detecting patterns intrinsic to the Skype protocol. It discusses how Skype uses a peer-to-peer network with super nodes to route traffic. The methodology section outlines how the author captured and analyzed Skype signaling traffic under different network configurations. Key findings include that Skype uses incrementing session IDs, encodes message types, and exchanges IP addresses for NAT detection during an initial UDP probe handshake with super nodes.
Here is a draft proposal for migrating the Windows XP machines in the new LSDG research group to Linux:
Proposal to Migrate LSDG Desktops from Windows XP to Linux
Introduction
The new LSDG research group at Linx LLC will be using desktop operating systems. Currently, some machines in the larger Linx LLC organization run Windows XP and Windows 7. As LSDG will be a separate research group, we need to consider the best desktop OS choice for their needs and the longevity of the machines.
Analysis
Windows XP is no longer supported by Microsoft, so continuing to use it poses major security risks. Without updates and patches, XP machines are vulnerable to exploits. Support for Windows 7 will also end
This document discusses multicasting and provides examples of how it can be used. Multicasting allows a server to send data to multiple clients simultaneously without using excessive bandwidth. It describes how multicasting works using UDP and IGMP, and provides examples of chat and picture sharing applications that can benefit from multicasting. The key aspects of multicasting covered are unicast, broadcast, and multicast addressing; the IGMP protocol; multicast routing; and application models for multicasting.
International Journal of Engineering Research and Applications (IJERA) is an open access online peer reviewed international journal that publishes research and review articles in the fields of Computer Science, Neural Networks, Electrical Engineering, Software Engineering, Information Technology, Mechanical Engineering, Chemical Engineering, Plastic Engineering, Food Technology, Textile Engineering, Nano Technology & science, Power Electronics, Electronics & Communication Engineering, Computational mathematics, Image processing, Civil Engineering, Structural Engineering, Environmental Engineering, VLSI Testing & Low Power VLSI Design etc.
This document provides information about the Networks Laboratory course offered at Anjalai Ammal Mahalingam Engineering College. It includes the syllabus, list of experiments, objectives and outcomes of the course. The course aims to teach students socket programming, simulation tools, and hands-on experience with networking protocols. Some key experiments include implementing stop-and-wait and sliding window protocols, socket programming, simulating ARP/RARP, PING and traceroute, and studying routing algorithms. The course is intended to help students use simulation tools, implement protocols, and analyze network performance and routing.
This document provides a summary of Prateek's professional experience in software development for telecom and networking. Over 9.5 years, he has worked on projects involving optical networking, load balancing servers, protocol development, and customer support. His responsibilities have included technical lead roles, individual development work, design, testing, and system integration. He has strong skills in C, C++, Linux, networking protocols, data structures, and development tools like version control systems. His work experience includes roles at NEC Technology, Brocade Communication, Juniper Networks, and Huawei Technology where he contributed to projects involving network security, load balancing, network address translation, and more.
The document discusses Internet Protocol version 4 (IPv4) and version 6 (IPv6). It provides details on IPv4 such as its 32-bit addressing scheme divided into classes A, B, and C. It also describes IPv4 features, benefits, and shortcomings like limited address space. The document then covers IPv6 which uses a 128-bit address format to overcome IPv4 limits and provide additional features.
The document provides information about various networking concepts and protocols. It contains 26 questions and answers about topics such as IGMP, ping, tracert, RSVP, DHCP, domains vs workgroups, NAT, PPP, IP spoofing, IP datagrams, application gateways, circuit gateways, default gateways, LANs, intranets vs the Internet, protocols, FTP, the OSI model layers, network types, topologies, IP, TCP, UDP, IP addressing classes, multicasting, DNS, telnet, and SMTP. It also defines MAC addresses.
National Diploma Unit 08 Communication Technology Assignment 2 Support Material provides information on communication protocols and wireless technology. It explains the principles of signal theory and describes common communication protocols like TCP/IP and Bluetooth. It discusses how digital signals are represented as strings of zeros and ones. The document also covers wireless LAN protocols like 802.11a, 802.11b, and 802.11g and security protocols like WEP and WPA. It provides examples of wireless technologies in use such as mobile phones, WiFi, and infrared communications.
Domain 4: Communication and Network Security - Review
Application Layer TCP/IP Protocols and Concepts, Layer 1 Network Cabling, LAN Technologies and Protocols, LAN Physical NetworkTopologies, WAN Technologies and Protocols, Network Devices and Protocols and Network Attacks
The document outlines the course content for a Small Office Home Office (SOHO) IT Network Setup course. The course covers topics such as network components, configurations, email and file sharing setup. It includes chapters on understanding networks, network components, terminologies, a SOHO network lab, and advanced Google search operators. Network abbreviations and concepts such as IP addressing, static versus dynamic IP, and private versus public IP are also defined.
Small office Home office , network setup in detailsapel7
This document provides an outline for a course on Small Office Home Office (SOHO) IT network setup. It includes chapters on network fundamentals like components, configurations and terminology. It also covers topics like email and file sharing setup. An initial section defines common network abbreviations.
The Kyoto Protocol was devised to reduce greenhouse gas emissions and combat global warming. It outlined goals for developed countries to reduce emissions and increase energy efficiency, minimize emissions increases in developing countries, and promote sustainable practices. The Protocol set specific emissions reduction targets ranging from 8-10% below 1990 levels for developed countries. This document examines challenging aspects of international climate negotiations, the successes and failures of the agreement, factors contributing to these outcomes, and an overall assessment of the agreement.
The document discusses network design using TCP/IP. It covers IP addressing, subnet masks, default gateways, and subnetting. It also discusses network security methods like IP packet filtering, encryption, authentication, and IPSec. Optimizing the subnet design, IP performance, remote subnets, and quality of service can create an effective network infrastructure.
This session covered cyber security and ethical hacking topics such as network hacking, Kali Linux, IPV4 vs IPV6, MAC addresses, wireless hacking techniques like deauthentication attacks, cracking WEP and WPA encryption, and post-connection attacks including ARP spoofing and MITM attacks. The presenter emphasized the importance of securing networks by using strong passwords, disabling WPS, and enabling HTTPS to prevent hacking attempts.
Lucene is an open source search engine library written in Java. It provides full text search functionality and supports indexing, searching, sorting and filtering of documents. Lucene creates an inverted index of terms extracted from documents, which allows for fast searching. The index is divided into segments for improved performance. Documents are indexed by adding them to memory first before being flushed to segments. Updates and deletes are handled by marking documents as deleted rather than removing them. Scoring is based on term frequency and inverse document frequency to determine relevance to a query.
Agile is an adaptive and iterative process for managing product development and delivery that focuses on continuous improvement and delivering value to customers. It involves planning in short iterations, implementing changes incrementally, obtaining frequent feedback to make improvements, and emphasizing collaboration and self-organization among cross-functional teams. Some key Agile methods include Scrum, Kanban, and extreme programming (XP).
MapReduce provides an easy way to process large datasets in a distributed manner. It uses mappers to process input data and generate intermediate key-value pairs, and reducers to combine those intermediate pairs into the final output. Key aspects include job tracking, splitting data into tasks, and storing intermediate output locally rather than on HDFS for efficiency, since it is discarded after reducing.
Hive allows querying and managing large datasets stored in distributed systems like HDFS. It uses a schema-on-read approach where the schema is not enforced until query time, allowing flexible data storage. Hive supports built-in, user-defined, and aggregate functions. It performs equi-joins and supports various join strategies like map-side joins, reduce-side joins, shuffle joins, and bucket joins. Map-side joins perform the join during the map phase by joining records with the same key from sorted and partitioned tables.
JSON has an easier grammar and is more lightweight than XML, leading to better performance. It is also readable, transportation independent, and can represent almost any data. There are parsers available for JSON in every programming language. Jackson and Gson are two libraries that can be used to parse JSON in Java - with Jackson using ObjectMapper to read/write JSON strings to POJOs, and Gson using similar fromJson/toJson methods. The POJOs need default constructors when parsing with these libraries.
Analysis of Emergency Evacuation of Building using PEPASurinder Kaur
This document proposes using process algebra to model and analyze emergency evacuation from a multi-story building. It identifies issues to address like evacuation time, congestion points, individual behaviors, and how the situation affects evacuation. The model represents evacuees, places, rescuers, the situation, and their interactions using PEPA processes. Results show plots of stair occupancy over time for different stair widths, people saved by external rescuers over time, evacuation time for various situations, and evacuation time vs external evacuees.
Network Address Translation (NAT) allows private IP addresses to be used within a local area network (LAN) while providing access to the public internet. NAT maps private IP addresses to public IP addresses, allowing multiple devices to share public IP addresses. The main NAT traversal challenges are that NAT prevents outside systems from initiating connections to inside systems and communication between systems that are both behind NAT routers. Proposed solutions include using third-party servers to reverse connections or techniques like UDP and TCP hole punching that establish connections directly between systems.
XSLT is used to transform XML documents into other XML documents or HTML. It uses XPath to navigate XML documents. Templates are used to define transformation rules that are applied when nodes are matched. Common elements used in XSLT include value-of to extract node values, for-each for loops, apply-templates to apply templates to child nodes, and copy to duplicate nodes in the output. Conditional logic can be added using elements like if, choose, when and otherwise.
The DOM (Document Object Model) is a W3C standard that defines a programming interface for XML and HTML documents. The DOM represents an XML document as nodes and objects that can be manipulated programmatically. The DOM defines the logical structure of documents and the way a document is accessed and manipulated. Key points:
- The DOM allows manipulation of the contents of an XML document through a programming interface.
- The DOM represents an XML document as a tree structure, with nodes and objects that can be accessed and manipulated.
- Common DOM node types include elements, attributes, text nodes, comments and documents. The DOM defines interfaces and properties to represent the node relationships and access node contents.
This document provides an overview of using the DOM (Document Object Model) API in Java to parse, traverse, modify, and generate XML documents. It describes the core DOM interfaces like Document and Node, how to parse an XML file into a DOM document tree using JAXP, and examples of traversing nodes, modifying content, and generating a new XML document from scratch using the DOM.
intelligent sensors and sensor networksSurinder Kaur
The document discusses intelligent sensors and sensor networks. It describes using neural networks for decision making and learning in intelligent sensors. Specifically, it discusses using spiking neural networks for human localization based on sensor data from laser range finders and other sensors. It also examines using neural networks like radial basis function networks and multilayer perceptrons for material classification based on sensor readings. Finally, it proposes a universal sensor interface chip that can provide local intelligence to develop various intelligent sensor applications.
OpenMP and MPI are two common APIs for parallel programming. OpenMP uses a shared memory model where threads have access to shared memory and can synchronize access. It is best for multi-core processors. MPI uses a message passing model where separate processes communicate by exchanging messages. It provides portability and is useful for distributed memory systems. Both have advantages like performance and portability but also disadvantages like difficulty of debugging for MPI. Future work may include improvements to threading support and fault tolerance in MPI.
A review of the growth of the Israel Genealogy Research Association Database Collection for the last 12 months. Our collection is now passed the 3 million mark and still growing. See which archives have contributed the most. See the different types of records we have, and which years have had records added. You can also see what we have for the future.
Main Java[All of the Base Concepts}.docxadhitya5119
This is part 1 of my Java Learning Journey. This Contains Custom methods, classes, constructors, packages, multithreading , try- catch block, finally block and more.
Chapter wise All Notes of First year Basic Civil Engineering.pptxDenish Jangid
Chapter wise All Notes of First year Basic Civil Engineering
Syllabus
Chapter-1
Introduction to objective, scope and outcome the subject
Chapter 2
Introduction: Scope and Specialization of Civil Engineering, Role of civil Engineer in Society, Impact of infrastructural development on economy of country.
Chapter 3
Surveying: Object Principles & Types of Surveying; Site Plans, Plans & Maps; Scales & Unit of different Measurements.
Linear Measurements: Instruments used. Linear Measurement by Tape, Ranging out Survey Lines and overcoming Obstructions; Measurements on sloping ground; Tape corrections, conventional symbols. Angular Measurements: Instruments used; Introduction to Compass Surveying, Bearings and Longitude & Latitude of a Line, Introduction to total station.
Levelling: Instrument used Object of levelling, Methods of levelling in brief, and Contour maps.
Chapter 4
Buildings: Selection of site for Buildings, Layout of Building Plan, Types of buildings, Plinth area, carpet area, floor space index, Introduction to building byelaws, concept of sun light & ventilation. Components of Buildings & their functions, Basic concept of R.C.C., Introduction to types of foundation
Chapter 5
Transportation: Introduction to Transportation Engineering; Traffic and Road Safety: Types and Characteristics of Various Modes of Transportation; Various Road Traffic Signs, Causes of Accidents and Road Safety Measures.
Chapter 6
Environmental Engineering: Environmental Pollution, Environmental Acts and Regulations, Functional Concepts of Ecology, Basics of Species, Biodiversity, Ecosystem, Hydrological Cycle; Chemical Cycles: Carbon, Nitrogen & Phosphorus; Energy Flow in Ecosystems.
Water Pollution: Water Quality standards, Introduction to Treatment & Disposal of Waste Water. Reuse and Saving of Water, Rain Water Harvesting. Solid Waste Management: Classification of Solid Waste, Collection, Transportation and Disposal of Solid. Recycling of Solid Waste: Energy Recovery, Sanitary Landfill, On-Site Sanitation. Air & Noise Pollution: Primary and Secondary air pollutants, Harmful effects of Air Pollution, Control of Air Pollution. . Noise Pollution Harmful Effects of noise pollution, control of noise pollution, Global warming & Climate Change, Ozone depletion, Greenhouse effect
Text Books:
1. Palancharmy, Basic Civil Engineering, McGraw Hill publishers.
2. Satheesh Gopi, Basic Civil Engineering, Pearson Publishers.
3. Ketki Rangwala Dalal, Essentials of Civil Engineering, Charotar Publishing House.
4. BCP, Surveying volume 1
This document provides an overview of wound healing, its functions, stages, mechanisms, factors affecting it, and complications.
A wound is a break in the integrity of the skin or tissues, which may be associated with disruption of the structure and function.
Healing is the body’s response to injury in an attempt to restore normal structure and functions.
Healing can occur in two ways: Regeneration and Repair
There are 4 phases of wound healing: hemostasis, inflammation, proliferation, and remodeling. This document also describes the mechanism of wound healing. Factors that affect healing include infection, uncontrolled diabetes, poor nutrition, age, anemia, the presence of foreign bodies, etc.
Complications of wound healing like infection, hyperpigmentation of scar, contractures, and keloid formation.
A workshop hosted by the South African Journal of Science aimed at postgraduate students and early career researchers with little or no experience in writing and publishing journal articles.
How to Add Chatter in the odoo 17 ERP ModuleCeline George
In Odoo, the chatter is like a chat tool that helps you work together on records. You can leave notes and track things, making it easier to talk with your team and partners. Inside chatter, all communication history, activity, and changes will be displayed.
This presentation was provided by Steph Pollock of The American Psychological Association’s Journals Program, and Damita Snow, of The American Society of Civil Engineers (ASCE), for the initial session of NISO's 2024 Training Series "DEIA in the Scholarly Landscape." Session One: 'Setting Expectations: a DEIA Primer,' was held June 6, 2024.
বাংলাদেশের অর্থনৈতিক সমীক্ষা ২০২৪ [Bangladesh Economic Review 2024 Bangla.pdf] কম্পিউটার , ট্যাব ও স্মার্ট ফোন ভার্সন সহ সম্পূর্ণ বাংলা ই-বুক বা pdf বই " সুচিপত্র ...বুকমার্ক মেনু 🔖 ও হাইপার লিংক মেনু 📝👆 যুক্ত ..
আমাদের সবার জন্য খুব খুব গুরুত্বপূর্ণ একটি বই ..বিসিএস, ব্যাংক, ইউনিভার্সিটি ভর্তি ও যে কোন প্রতিযোগিতা মূলক পরীক্ষার জন্য এর খুব ইম্পরট্যান্ট একটি বিষয় ...তাছাড়া বাংলাদেশের সাম্প্রতিক যে কোন ডাটা বা তথ্য এই বইতে পাবেন ...
তাই একজন নাগরিক হিসাবে এই তথ্য গুলো আপনার জানা প্রয়োজন ...।
বিসিএস ও ব্যাংক এর লিখিত পরীক্ষা ...+এছাড়া মাধ্যমিক ও উচ্চমাধ্যমিকের স্টুডেন্টদের জন্য অনেক কাজে আসবে ...
Leveraging Generative AI to Drive Nonprofit InnovationTechSoup
In this webinar, participants learned how to utilize Generative AI to streamline operations and elevate member engagement. Amazon Web Service experts provided a customer specific use cases and dived into low/no-code tools that are quick and easy to deploy through Amazon Web Service (AWS.)
2. 1 Introduction
Skype is peer-to-peer VoIP application . It is used for voice calls and text
messages. Now file transfer and video conferencing is also alloweed.
It was launched in 2003 by Niklas Zennstrom and JanusFriis. Now
Microsoft owns it.The first generation of VoIP is MSN and IM It is the sec-
ond generation of VoIP technology and its successsor i.e. the third generation
of voIP is represented by Google Talk
The term VoIP stands for Voice over Internet Protocol. and it refers to the
protocol and the technologies employed for voice communication and multi-
media session over the IP. The other commonly used for the same purpose
are Interent Telephony or IP telephony. Using these technologies a user can
make telephone calls via internet rather than the traditional public switched
telephone networks. IT is a complete process that involve various steps. these
ssteps are listed below:
• Signaaling and media channel setup
• Digitization of analog voice signals
• encoding of signals
• Packetization
• Transmission of packets over IP network
2 Services provided by Skype
Skype provides following services:
• Messaging
For text messaging the additional features provided are group chat,
emoticons record of messaging history and even editing of previous mes-
sages.
• Voice chat
It allows telephone calls between a pair of skype users, conferencing
and use of a proprietary audio codec. The telephone call between a
pair of skype user for a group of country charges are same as charges
for calls within the country.
1
3. • Video conferencing
Now it supports video conferencing for all major software platform like
windows, Mac OS X, Linux. It can support video call upto 5 people.
The latest version of skype supports high quality video for windows.
• Audio conferencing
Like video conferencing audio conferencing is also supported by all
major platforms. However the number of user participate in audio
conferencing is much more than that in video conferencing . It can
support upto 25 participants.
3 Technical details
Skype is hybrid peer-to-peer and client-server system. The most interesting
feature of skype is that it is able to work efficiently even in the presence
of firewalls and NAT i.e. it is able to penetrate the NAT and firewall in
peer-to-peer approach.
It uses decentralized approach. It has no central server other than the
logon server. The whole network is maintained in the decentralized manner
using overlay peer-to-peer technology. Since it is usin g peer-to-peer technol-
ogy . Hence each node that is part of the network has to contribute to the
network, its bandwidth and the some percent of CPU cycles. In this way the
whole network load gets distributed among all the participating nodes.
Some important terms used in the Skype technology are described in follow-
ing section:
Skype Client
Each Skype user is termed Skype client. All the skype users are similar
i.e. peers and all are termed Skype client. There is ony one server that
maintains the login details of the user.
Ports
Skype client opens the TCP or UDP connection on the specified port
number. The Port number is given at the time of installation.
Host Cache
Host cache is a table that contain the list of super nodes. Each entry of
super noder has its IP address and port number. Each SC refresh this
2
4. cache at regular interval. Refreshing the table is the most important
part of entire Skype operation. This cache helps client to find the list
of all reachable nodes.
Codecs
The codec is used to convert the analog voice signals to digital signals.
Skype codecs allow frequencies between 50-8000 Hz to pass through.
This range is the characteristic of wideband codec.
Buddy list
Buddy list i.e. the list of all the Skype clients to which a user wants
to communicate to. The information about the buddies is stored in
the Windows registry. The buddy list is digitally signed and encrypted.
The list is local to the SC’s machine and is not stored on the central
server. Hence when a client uses a different machine it need to log on
to Skype network and reconstruct its buddy list.
Encryption
Skype uses 256-bit encryption algrithm AES i.e. Advanced Encryp-
tion Standard to encrypt the digital data before transmitting it on the
internet.
NAT and firerwall
SC uses STUN and TURN to determine the type of NAT and firewall
behind which the SC is working. Like the buddy list it is also stored
locally on the windows registry. Hence it needs to refreshed periodically.
4 Skype Functions
The step by step description of the Skype Functions is as following:
Startup When the Skype Client run the software for the first time,
this step is required.Once it get installed further this step is not needed.
Login Each time the SC wants to use Skype it needs to login. It
requries to enter its login details i.e. username and password. Then
the central server autenticates it with the stored details. The login step
also notifies the other SC about the presence of the currently logined
SC. In this step the NAT and Firewall are also determined.
User search Skype uses the Global Index Technology to search
foe the other users.This search is distributed. It is claimed by the Skype
3
5. and observed during various researches that it is guranteed to find a user
if it exists and has logged in during the llast 72 hours. Skype provides
a search dialog box where user can enter the SC it is searching for and
press the find button. The search process proceed s in differntly on the
different type of networks. The search process is explained briefly :
When the SC is on public IP address The SC sends a TCP
packet to its Super Node(SN).
SN in turn provide SC list of four nodes with their IP address and
port number to contact.
Now SC sends UDP packet to these 4 nodes asking wheter they
are the node SC is looking for. If it find the nodde SC stops the
search. In case it fails to find the node, it informs the SN using
TCP packet.
Now SN provide it list of 8 different nodes to contact to. Again
SC asks these nodes using UDP, if it is successful this time it stops
otherwise it proceed in the same manner until it finds the user or
it is determined that user doesn’t exists.
However the researches shows thst the Skype able to find the user
if it exists within few seconds.
When Sc is behind port-restricted NAT A SC behind a port re-
stricted NAT exchanges data between SN, and some of the nodes
which responds to its UDP request during login process.
SC behind port-restricted NAT, UDP-restricted firewall
SC sends its search request over TCP to its SN.
The SN then performs the search on the behalf of the SC. Then SN
informs SC of the search results. Thus in this case the SC itself
does not contact any of the nodes.
It is worth noting that the search results are cached at the
intermediate nodes.
Call establishment and Tear Down
Call can be made to a user in the Buddy list or to the user not in
the buddy list. The process for call establishment is explained in the
following :
When the callee is in the buddy list When the callee is already
there in the SC’s buddy list the SC just need to call the buddy.
However the call establishment mechanism is different for differ-
ent network setup. The following points desccribe the call setup
process in detail.
4
6. Caller and the callee both are on the public IP
As the caller clicks on the call button, the caller sends a
TCP : SYN message to establish the connection. The callee
acknowledges the caller’s request by sending a TCP : ACK
packet.
In this way TCP connection between the caller and the callee
establishes. After wards the caller and the callee communi-
cates using TCP connection.
Caller behind port-restricted NAT, callee on public IP
In this case the signaling and messages don’t flow directly
between the caller and the callee. The caller sends signaling
informationover TCP to a super node.
Now the super node forwardds the signal to callee over TCP.
In this way the call gets established.
Both are behind the port-restricted NAT, UDP-restricted firewall
The call is established via supernode over TCP connection.
and then they communicate.
When the callee is not in the buddy list
When the callee is not in the buddy list of the caller, the caller first
need search the callee over the networ, using the above mentioned
approaches. When it finds the callee , add it to its buddy list and
then call establisment is done in the same way when the callee is
in the buddy list of the caller.
TEARDOWN : The process of tear down is same as the process
of call establishment.
Media Transfer and Codecs:
The process is explained in the following points:
When both the caller and the callee are on the public IP
The media flows over the UDP betweent he caller and the callee.
Either of the caller or the callee is behind the port-restricted NAT
In this case the audio data is transmitted via a supernode over
UDP.
Both are behind port-restricted NAT and UDP-restricted firewall
5
7. The communication is done via supernode over TCP.
Some important feature supported by the Skype during audio/video
transfer:
• Silence Suppression : Slice suppression means that when nei-
ther the caller nor the callee is sending any data i.e. not speaking,
then the voice packets should not flow between them. However
the Skype does not support silence suppression.
• Skype supports puttingcallonhold.
• The experiements shows that the Codec frequency range is 50 Hz
to 8000 Hz.
Conferencing
5 Security in Skype
Skype uses 256 − bit encryption algorithm AES i.e. Advanced Encryption
Standard. This algorithm is considered to be the strongest encryption algo-
rithm, which is also used by the U.S. govt. organizations.
6 Relaying in Skype
6.1 Relaying
When the two peers in any peer-to-peer network could not communicatte
directly due to any reason, they communicate via another peer or node in
p2p network. Such communication techniques is termed as relaying.
Relaying plays a very important role in Skype p2p networking.
6.2 Relaying is needed in the following cases:
• When callee is behind NAT ,in this case if the caller attempts to
establish the connection in usual way or directly, then the caller initi-
ated communication will be considered as outside to inside connection
attempt by the NAT aat callee’s end. So the NAT will simply refuse
the connection attempt.
6
8. Hence the caller need the assistance from relay server to establish the
connection. The caller is connected to a superpeer, When it wants
to call the callee, it first informs the superpeer that it wants to have
connection with the specified callee.
The superpeer then communicate this request to the non-NATed su-
perpeer to which the callee is connected. The callee’s superpeer then
informs this to the callee.
The callee then sends a fake message addressed to the caller, this will
make a hole in the NAT.
Now the caller’s message can enter the callee’s network. Because these
messages will be considered as response to the callee’s fake message.
• Boththecaller and the callee isbehind NAT, in this case neither
the caller nor the callee can accept a call initiated by the other, making
the call seemingly impossible.
Now the super peers are used for relaying.The caller should have con-
nection with a non-NATed superpeer to be able to call a callee behind
the NAT. On the callee’s end it is required that the callee should also
have a connection with a non-NATed super peer to recieve the calls
from a caller behind NAT.
When the caller wants to call callee behindNAT, it first informs to the
non-NATed superpeer to which it already has the connection cconnec-
tion that it wants to make call to the specific callee by mentioning its
IP and port.
The superpeer inturn will inform to the superpeer to which the callee is
connected that the caller with given IP and port wants to have commu-
nicaation wih the callee. Now the callee’s superpeer informs the callee
that the caller wants to communicate with you.
Now its upto callee, whether it accepts the call or not. If callee accepts
the caller’s request the involved super eers select a third party non-
NATed superpeer , this superpeer is termed the relay node and its role
is to relay the messages between the caller and the callee through the
communication.
The superpeer connected to the caller and callee then instructs the
caller and the callee respectively to establish a connection with the cho-
sen superpeer.
Now the caller and the callee both get connected to a common superpeer-
the relay node.
7
9. When caller wants to send message to the calle, it first transmits the
message to the relay node, which forward the packet to the callee.
When the callee wants to response to the caller, the callee simply sends
this response to the relay node, which forward it to the caller.
In this way the whole communication between the caller and callee
behind NAt communicates.
8