A method for forensic artifact collection, analysis and incident response in environments running Session Initiation Protocol (SIP) and Session Description protocol

In this paper, we perform an analysis of SIP, a popular voice over IP (VoIP) protocol and propose a framework for capturing and analysing volatile VoIP data in order to determine forensic readiness requirements for effectively identifying an attacker. The analysis was performed on real attack data and the findings were encouraging. It seems that if appropriate forensic readiness processes and controls are in place, a wealth of evidence can be obtained. The type of the end user equipment of the internal users, the private IP, the software that is used can help build a reliable baseline information database. On the other hand the private IP addresses of the potential attacker even during the presence of NAT services, as well as and the attack tools employed by the malicious parties are logged for further analysis

Study, Analysis and Modeling of IP Multimedia systems on next generation networks providing mobile and fixed multimedia services

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Security aspects in voice over IP systems

Security has become a major concern with the rapid growth of interest in the internet. This project deals with the security aspects of VoIP systems. Various supporting protocols and technologies are considered to provide solutions to the security problems. This project stresses on the underlying VoIP protocols like Session Initiation Protocol (SIP), Secure Real-time Transport Procotol (SRTP), H.323 and Media Gateway Control Protocol (MGCP). The project further discusses the Network Address Translation (NAT) devices and firewalls that perform NAT. A firewall provides a point of defense between two networks. This project considers issues regarding the firewalls and the problems faced in using firewalls for VoIP; it further discusses the solutions about how firewalls can be used in a more secured way and how they provide security

Signaling for Internet Telephony

Internet telephony must offer the standard telephony services.However, the transition to Internet-based telephony services also provides an opportunity to create new services more rapidly and with lower complexity than in the existing public switched telephone network(PSTN). The Session Initiation Protocol (SIP) is a signaling protocol that creates, modifies and terminates associations between Internet end systems, including conferences and point-to-point calls. SIP supports unicast, mesh and multicast conferences, as well as combinations of these modes. SIP implements services such as call forwarding and transfer, placing calls on hold, camp-on and call queueing by a small set of call handling primitives. SIP implementations can re-use parts of other Internet service protocols such as HTTP and the Real-Time Stream Protocol (RTSP). In this paper, we describe SIP, and show how its basic primitives can be used to construct a wide range of telephony services

Security Enhancements in Voice Over Ip Networks

Voice delivery over IP networks including VoIP (Voice over IP) and VoLTE (Voice over LTE) are emerging as the alternatives to the conventional public telephony networks. With the growing number of subscribers and the global integration of 4/5G by operations, VoIP/VoLTE as the only option for voice delivery becomes an attractive target to be abused and exploited by malicious attackers. This dissertation aims to address some of the security challenges in VoIP/VoLTE. When we examine the past events to identify trends and changes in attacking strategies, we find that spam calls, caller-ID spoofing, and DoS attacks are the most imminent threats to VoIP deployments. Compared to email spam, voice spam will be much more obnoxious and time consuming nuisance for human subscribers to filter out. Since the threat of voice spam could become as serious as email spam, we first focus on spam detection and propose a content-based approach to protect telephone subscribers\u27 voice mailboxes from voice spam. Caller-ID has long been used to enable the callee parties know who is calling, verify his identity for authentication and his physical location for emergency services. VoIP and other packet switched networks such as all-IP Long Term Evolution (LTE) network provide flexibility that helps subscribers to use arbitrary caller-ID. Moreover, interconnecting between IP telephony and other Circuit-Switched (CS) legacy telephone networks has also weakened the security of caller-ID systems. We observe that the determination of true identity of a calling device helps us in preventing many VoIP attacks, such as caller-ID spoofing, spamming and call flooding attacks. This motivates us to take a very different approach to the VoIP problems and attempt to answer a fundamental question: is it possible to know the type of a device a subscriber uses to originate a call? By exploiting the impreciseness of the codec sampling rate in the caller\u27s RTP streams, we propose a fuzzy rule-based system to remotely identify calling devices. Finally, we propose a caller-ID based public key infrastructure for VoIP and VoLTE that provides signature generation at the calling party side as well as signature verification at the callee party side. The proposed signature can be used as caller-ID trust to prevent caller-ID spoofing and unsolicited calls. Our approach is based on the identity-based cryptography, and it also leverages the Domain Name System (DNS) and proxy servers in the VoIP architecture, as well as the Home Subscriber Server (HSS) and Call Session Control Function (CSCF) in the IP Multimedia Subsystem (IMS) architecture. Using OPNET, we then develop a comprehensive simulation testbed for the evaluation of our proposed infrastructure. Our simulation results show that the average call setup delays induced by our infrastructure are hardly noticeable by telephony subscribers and the extra signaling overhead is negligible. Therefore, our proposed infrastructure can be adopted to widely verify caller-ID in telephony networks

An Architecture for Global Distributed SIP Network Using IPv4 Anycast

Tato diplomová práce se zabývá metodami pro výběr nejbližší RTP proxy k VoIP klientům s použitím IP anycastu. RTP proxy servery jsou umístěny v síti Internetu a přeposílají RTP data pro VoIP klienty za síťovými překladači adres(NAT). Bez zeměpisně rozmístěných RTP proxy serverů a metod pro nalezení nejbližšího RTP proxy serveru by došlo ke zbytečnému poklesu kvality přenosu médialních dat a velkému zpoždení. Tento dokument navrhuje 4 metody a jejich porovnání s podrobnějšími rozbory metod s využitím DNS resolvování a přímo SIP protokolu. Tento dokument také obsahuje měření chování IP anycastu v porovnání mezi metrikami směrování a metrikami časovými. Nakonec dokumentu je také uvedena implemetace na SIP Express Router platformě.This thesis is about using IP anycast-based methods for locating RTP proxy servers close to VoIP clients. The RTP proxy servers are hosts on the public Internet that relay RTP media between VoIP clients in a way that accomplishes traversal over Network Address Translators (NATs). Without geographically-dispersed RTP proxy servers and methods to find one in client's proximity, voice latency may be unbearably long and dramatically reduce perceived voice quality. This document proposes four methods their comparison with further design of DNS-based and SIP-based methods. It includes IP anycast measurements that provides an overview of IP anycast behaviour in terms of routing metrics and latency metrics. It also includes implementation on SIP Express Router platform.

Security for the signaling plane of the SIP protocol

VOIP protocols are gaining greater acceptance amongst both users and service providers. This thesis will aim to examine aspects related to the security of signaling plane of the SIP protocol, one of the most widely used VOIP protocols. Firstly, I will analyze the critical issues related to SIP, then move on to discuss both current and possible future solutions, and finally an assessment of the impact on the performance of HTTP digest authentication, IPsec and TLS, the three main methods use

Using decoys to block SPIT in the IMS

Includes bibliographical references (leaves 106-111)In recent years, studies have shown that 80-85% of e-mails sent were spam. Another form of spam that has just surfaced is VoIP (Voice over Internet Telephony) spam. Currently, VoIP has seen an increasing numbers of users due to the cheap rates. With the introduction of the IMS (IP Multimedia Subsystem), the number of VoIP users are expected to increase dramatically. This calls for a cause of concern, as the tools and methods that have been used for blocking email spam may not be suitable for real-time voice calls. In addition, VoIP phones will have URI type addresses, so the same methods that were used to generate automated e-mail spam messages can be employed for unsolicited voice calls. Spammers will always be present to take advantage of and adapt to trends in communication technology. Therefore, it is important that IMS have structures in place to alleviate the problems of spam. Recent solutions proposed to block SPIT (Spam over Internet Telephony) have the following shortcomings: restricting the users to trusted senders, causing delays in voice call set-up, reducing the efficiency of the system by increasing burden on proxies which have to do some form of bayesian or statistical filtering, and requiring dramatic changes in the protocols being used. The proposed decoying system for the IMS fits well with the existing protocol structure, and customers are oblivious of its operation