Managing ClientInitiated Connections

The Session Initiation Protocol (SIP) allows proxy servers to initiate TCP connections or to send asynchronous UDP datagrams to User Agents in order to deliver requests. However, in a large number of real deployments, many practical considerations, such as the existence of firewalls and Network Address Translators (NATs) or the use of TLS with server-provided certificates, prevent servers from connecting to User Agents in this way. This specification defines behaviors for User Agents, registrars, and proxy servers that allow requests to be delivered on existing connections established by the User Agent. It also defines keep-alive behaviors needed to keep NAT bindings open and specifies the usage of multiple connections from the User Agent to its registrar. Status of This Memo This document specifies an Internet standards track protocol for the Internet community, and requests discussion and suggestions for improvements. Please refer to the current edition of the "Internet Official Protocol Standards " (STD 1) for the standardization state and status of this protocol. Distribution of this memo is unlimited. Copyright Notice Copyright (c) 2009 IETF Trust and the persons identified as th

Handling of IP-Addresses in the Context of Remote Access

Masteroppgave i informasjons- og kommunikasjonsteknologi 2008 – Universitetet i Agder, GrimstadFor various reasons (e.g., security, lack of IPv4-addresses) the services in the home smart space only use private IP addresses. This is unfortunate in the remote service access since these addresses frequently appear in responses sent from a service in the remote smart space (e.g., your home) to the visited smart space (e.g., your friend’s home).The Internet Engineering Task Force (IETF) provides some solutions and workarounds for the problem caused by NAT. In this project, the challenge to me is to summarize the available options, rank the options according to which one is preferred for the RA-scenario. I will come up with my practical NAT traversal techniques by testing and gathering data on the reliability of NAT traversal techniques since none of the existing ones seems to work well. A demonstration of the key features will be shown in the thesis. NAT traversal techniques apply to TCP and UDP need to be researched in advance. Handling of peers behind all kinds of NAT need to be tested and determined for the communication. The result of the paper will well improve the evaluation of specific issues on NAT and the creating of an UNSAF proposal

Publish/Subscribe Gateway for Real-time Communication

Tässä diplomityössä esitellään yhdyskäytävä, jolla voidaan yhdistää IP-verkot ja informaatiokeskeiset julkaisu/tilaus-verkot toisiinsa sekä mahdollistaa niiden käyttäminen rinnakkain. Internetin arkkitehtuuria on ehdotettu uudistettavaksi siten, että nykyisestä ns. isäntäkeskeisestä mallista siirryttäisiin informaatiokeskeiseen malliin. Eräs projekti, jossa tätä tutkitaan, on PURSUIT, jossa tietoliikenne perustuu julkaisu/tilaus-malliin. Projektissa on otettu huomioon myös tämän uuden arkkitehtuurin käyttöönottaminen Internetissä. Tähän liittyen tässä diplomityössä on suunniteltu yhdyskäytävä, joka muuntaa IP-liikenteen julkaisu/tilaaja-pohjaiseksi ja päinvastoin. Muunnos voidaan tehdä SIP-protokollaa käyttäville puheluille sekä IP-monilähetystä hyödyntäville multimediavirroille. Yhdyskäytävän avulla operaattorit voivat hyödyntää verkossaan informaatiokeskeisen mallin ominaisuuksia sekä siihen liittyviä mekanismeja, kuten tilatonta monilähetystä, ja verkon käyttäjät puolestaan voivat edelleen käyttää IP-yhteyksiä ja -sovelluksia. Työssä kuvataan, yhdyskäytävän toiminnallisuudet, jotka mahdollistavat SIP-istunnon alullepanemisen, parametrien neuvottelun, media-istunnon käynnistämisen sekä istunnon ylläpitämisen ja katkaisemisen julkaisu/tilaus-verkon ylitse. Työssä on myös suunniteltu SIP-rekisteriöintipalvelinsovellus, joka hoitaa käyttäjien rekisteröinnin, puheluiden uudelleenohjaukset sekä käyttäjien liikkuvuuden. Lisäksi kuvataan yhdyskäytävään sisältyvä mekanismi, jolla multimedian virtauttaminen monilähetyksenä on toteutettu. Yhteyskäytävä vastaa tässä tapauksessa monilähetysryhmien luomisesta ja purkamisesta sekä istuntoihin liittymisestä ja poistumisesta. Yhdyskäytävän suunnitelman lisäksi diplomityössä kuvataan prototyypin toteutus sekä arvioimme järjestelmän vastaavuutta työssä määriteltyihin vaatimuksiin. Lisäksi analysoimme järjestelmän suorituskykyä ja liikenteen määrää istuntojen eri vaiheissa, sekä vertaamme näitä tuloksia IP- ja julkaisu/tilaus-verkkojen välillä.This thesis proposes a design of a gateway, which connects IP and publish/subscribe networks together, enabling their co-existence, for example, during an IP to pub/sub migration phase. There is a proposal to revise the architecture of the present Internet, from "Host-Centric Networking" to a new concept called "Information-Centric Networking (ICN)". One of the ongoing projects in this field is the PURSUIT project, which uses the publish/subscribe paradigm as a basic communication model. Since the proposal from the PURSUIT project has gained quite much interest recently, the next step is to consider the process of deploying the new Internet architecture. This thesis focuses on gateway's mechanism to transparently convert IP-based end-to-end traffic to the publish/subscribe based and vice versa, in order to support voice communication using Session Initiation Protocol as well as multimedia streaming over multicast. The main idea of our design is to allow operators to utilize the features of Information-Centric Networking, while home users or companies can still use legacy IP connectivity and applications. In this scenario, the operators will gain benefits from new solutions, e.g., stateless Bloom-filter based multicast forwarding in the pub/sub network. We describe the gateway's functionalities to handle SIP session initialization, parameters negotiation, media session establishment, as well as maintaining and terminating the session over the publish/subscribe network. This includes a design of a pub/sub based SIP registrar for taking care of user registration, call redirection, and mobility. Moreover, we also discuss the mechanism to support multimedia streaming over multicast. Our gateway is responsible for group establishment, session joining and leaving, and eventually group termination. In addition to our design, we describe an implemented prototype, and evaluate the system's functionalities according to the requirements of this thesis. After that, we analyze the performance of the design and implementation, traffic density during different phases of both SIP and multicast sessions, and finally compare the call setup duration between IP and pub/sub networks

An FPGA-Based System for Tracking Digital Information Transmitted via Peer-to-Peer Protocols

This thesis addresses the problem of identifying and tracking digital information that is shared using peer-to-peer file transfer and Voice over IP (VoIP) protocols. The goal of the research is to develop a system for detecting and tracking the illicit dissemination of sensitive government information using file sharing applications within a target network, and tracking terrorist cells or criminal organizations that are covertly communicating using VoIP applications. A digital forensic tool is developed using an FPGA-based embedded software application. The tool is designed to process file transfers using the BitTorrent peer-to-peer protocol and VoIP phone calls made using the Session Initiation Protocol (SIP). The tool searches a network for selected peer-to-peer control messages using payload analysis and compares the unique identifier of the file being shared or phone number being used against a list of known contraband files or phone numbers. If the identifier is found on the list, the control packet is added to a log file for later forensic analysis. Results show that the FPGA tool processes peer-to-peer packets of interest 92% faster than a software-only configuration and is 99.0% accurate at capturing and processing BitTorrent Handshake messages under a network traffic load of at least 89.6 Mbps. When SIP is added to the system, the probability of intercept for BitTorrent Handshake messages remains at 99.0% and the probability of intercept for SIP control packets is 97.6% under a network traffic load of at least 89.6 Mbps, demonstrating that the tool can be expanded to process additional peer-to-peer protocols with minimal impact on overall performance

Service composition based on SIP peer-to-peer networks

Today the telecommunication market is faced with the situation that customers are requesting for new telecommunication services, especially value added services. The concept of Next Generation Networks (NGN) seems to be a solution for this, so this concept finds its way into the telecommunication area. These customer expectations have emerged in the context of NGN and the associated migration of the telecommunication networks from traditional circuit-switched towards packet-switched networks. One fundamental aspect of the NGN concept is to outsource the intelligence of services from the switching plane onto separated Service Delivery Platforms using SIP (Session Initiation Protocol) to provide the required signalling functionality. Caused by this migration process towards NGN SIP has appeared as the major signalling protocol for IP (Internet Protocol) based NGN. This will lead in contrast to ISDN (Integrated Services Digital Network) and IN (Intelligent Network) to significantly lower dependences among the network and services and enables to implement new services much easier and faster. In addition, further concepts from the IT (Information Technology) namely SOA (Service-Oriented Architecture) have largely influenced the telecommunication sector forced by amalgamation of IT and telecommunications. The benefit of applying SOA in telecommunication services is the acceleration of service creation and delivery. Main features of the SOA are that services are reusable, discoverable combinable and independently accessible from any location. Integration of those features offers a broader flexibility and efficiency for varying demands on services. This thesis proposes a novel framework for service provisioning and composition in SIP-based peer-to-peer networks applying the principles of SOA. One key contribution of the framework is the approach to enable the provisioning and composition of services which is performed by applying SIP. Based on this, the framework provides a flexible and fast way to request the creation for composite services. Furthermore the framework enables to request and combine multimodal value-added services, which means that they are no longer limited regarding media types such as audio, video and text. The proposed framework has been validated by a prototype implementation