Using an External DHT as a SIP Location Service

Peer-to-peer Internet telephony using the Session Initiation Protocol (P2P-SIP) can exhibit two different architectures: an existing P2P network can be used as a replacement for lookup and updates, or a P2P algorithm can be implemented using SIP messages. In this paper, we explore the first architecture using the OpenDHT service as an externally managed P2P network. We provide design details such as encryption and signing using pseudo-code and examples to provide P2P-SIP for various deployment components such as P2P client, proxy and adaptor, based on our implementation. The design can be used with other distributed hash tables (DHTs) also

Using SIP as P2P Technology

Nowadays peer-to-peer (p2p) technologies are widely adopted and used for building even more sophisticated services: from ubiquitous file-sharing systems to the even more popular Internet telephony. In addition, the Session Initiation Protocol (SIP) has been used for different purposes. Due to its intrinsic generality and flexibility, it could be adopted to build and manage also p2p applications. Moreover, the p2p philosophy could be applied to the existing SIP architecture, to cope with issues such as Denial of Service (DoS). In this paper, we survey the state of the art of the joint use of p2p and SIP. Some hints and examples in using SIP as a core technological component of the p2p world are also presented

Security in Peer-to-Peer SIP VoIP

VoIP (Voice over Internet Protocol) is one of the fastest growing technologies in the world. It is used by people all over the world for communication. But with the growing popularity of internet, security is one of the biggest concerns. It is important that the intruders are not able to sniff the packets that are transmitted over the internet through VoIP. Session Initiation Protocol (SIP) is the most popular and commonly used protocol of VoIP. Now days, companies like Skype are using Peer-to-Peer SIP VoIP for faster and better performance. Through this project I am improving an already existing Peer-to-Peer SIP VoIP called SOSIMPLE P2P VoIP by adding confidentiality in the protocol with the help of public key cryptography

Non-Repudiation in Internet Telephony

We present a concept to achieve non-repudiation for natural language conversations over the Internet. The method rests on chained electronic signatures applied to pieces of packet-based, digital, voice communication. It establishes the integrity and authenticity of the bidirectional data stream and its temporal sequence and thus the security context of a conversation. The concept is close to the protocols for Voice over the Internet (VoIP), provides a high level of inherent security, and extends naturally to multilateral non-repudiation, e.g., for conferences. Signatures over conversations can become true declarations of will in analogy to electronically signed, digital documents. This enables binding verbal contracts, in principle between unacquainted speakers, and in particular without witnesses. A reference implementation of a secure VoIP archive is exhibited.Comment: Accepted full research paper at IFIP sec2007, Sandton, South Africa, 14-16 May 200

Reflections on security options for the real-time transport protocol framework

The Real-time Transport Protocol (RTP) supports a range of video conferencing, telephony, and streaming video ap- plications, but offers few native security features. We discuss the problem of securing RTP, considering the range of applications. We outline why this makes RTP a difficult protocol to secure, and describe the approach we have recently proposed in the IETF to provide security for RTP applications. This approach treats RTP as a framework with a set of extensible security building blocks, and prescribes mandatory-to-implement security at the level of different application classes, rather than at the level of the media transport protocol

Past, present and future of IP telephony

“Copyright © [2008] IEEE. Reprinted from International Conference on Communication Theory, Reliability, and Quality of Service, 2008. CTRQ '08. ISBN:978-0-7695-3190-8. This material is posted here with permission of the IEEE. Internal or personal use of this material is permitted. However, permission to reprint/republish this material for advertising or promotional purposes or for creating new collective works for resale or redistribution must be obtained from the IEEE by writing to [email protected]. By choosing to view this document, you agree to all provisions of the copyright laws protecting it.”Since the late 90's IP telephony, commonly referred to as Voice over IP (VoIP), has been presented as a revolution on communications enabling the possibility to converge historically separated voice and data networks, reducing costs, and integrating voice, data and video on applications. This paper presents a study over the standard VoIP protocols H.323, Session Initiation Protocol (SIP), Media Gateway Control Protocol (MGCP), and H.248/Megaco. Given the fact that H.323 and SIP are more widespread than the others, we focus our study on them. For each of these protocols we describe and discuss its main capabilities, architecture, stack protocol, and characteristics. We also briefly point their technical limitations. Furthermore, we present the Advanced Multimedia System (AMS) project, a new system that aims to operate on Next Generation Networks (NGN) taking the advantage of its features, and it is viewed as the successor to H.323 and SIP