Session Initiation Protocol Protocol Research Articles

Desktop aplikacija za razmenu kriptozaštićenog govora

Introduction/purpose: All data exchanged over the Internet as well as other computer networks should be considered exposed to various types of security threats. In light of this, the transmission of voice over applications that do not use any type of crypto-protection allows anyone to discern the content of communication. Since voice transmission requires as little delay as possible, various protocols are used to enable crypto protected real-time communication. This paper presents one solution in a desktop application variant. Methods: The essence in voice exchange systems as well as in other systems where real-time communication is necessary is the establishment of a crypto-protected session which is a virtual secure channel for communication to which only the communicating parties have access. Voice sessions in the application are established with the SIP (Session Initiation Protocol) protocol. The sessions are further protected using the ZRTP (Zimmerman Real-time Transport Protocol) protocol. FusionPBX was used as the SIP server (registrar) for testing purpouses. The application is developed in C++ language using the Qt framework. Results: The final version of the application demostrates that ZRTP and SIP protocols are well suited for establishing crypto protected voice communications with low delay. Conclusion: This solution provides cryptographic functions for data secrecy and the management of cryptographic keys. Improving the solution with digital signatures and certificates will result in additional cryptographic functions: data integrity and personal identification. With this improvement, this solution will be able to withstand modern security threats with low delay.

Security Enhancement of SIP Protocol in VoIP Communication

In Voice over IP (VoIP) systems, calls are started, maintained, and ended using the Session Initiation Protocol (SIP). One of the SIP flaws is the lack of enough data to validate users, and its contents can be changed to fake the Caller Identification (CID). Fake CID can be used by hackers to trick receivers and obtain valuable materials. Existing solutions have a number of faults, including the use of more network resources, the use of insufficient data to identify fake CID, and high call setup delay for caller and callee during the validation process. This work presents an algorithm to enhance SIP protocol security to mitigate the flaws of the previous solutions that are used to address CID spoofing. By using the Media Access Control (MAC) address, the algorithm can validate the CID and warn the callee for the fake CID. The algorithm was developed and tested by using Mininet, Python based open source OpenFlow (POX) controller, SIPp simulator, Linphone softphone, and freePBX. All faked CID were detected and receivers were alerted on the linphone screen. Session setup delay obtained are within the 15.1 ms to 27.3 ms, which are within the acceptable Quality of Service (QoS) ranges.

Prototype wireless network for internet of things based on DECT standard

This paper presents a software prototype of a wireless network for the Internet of Things (IoT) based on the DECT (Digital Enhanced Cordless Telecommunication) standard. It proposes an architecture for encapsulating commands from the most common IoT protocol, MQTT (Message Queuing Telemetry Transport), into SIP (Session Initiation Protocol) packets. A module is created to embed MQTT-SN (MQTT for Sensor Networks) packets into SIP packets. The module is developed in Go language using the built-in "net" library. Delivery of MQTT-SN packets to IoT devices is carried out using the SIP protocol. Source codes and instructions for installing the gateway can be found at https://github.com/iSinyavsky/mqtt-sn-sip-gateway.

Load balancing with SIP protocol using distributed web-server system

Web server system describes an innovative algorithm for balancing load by allocating the task to the clusters (Pai et al., 1998) of SIP server (Aron and Druschel, 1999; Bernardos et al., 2016). The requisition to a cluster of SIP servers is developed by using session initiation protocol (SIP). This algorithm supports three main techniques for the locally distributed WS system such as CJSQ, TJSQ and TLWL for the allocation of works to least values in the servers. This technique is incorporated with the knowledge of the SIP server (Aron and Druschel, 1999; Bernardos et al., 2016; Foundry Networks, 2007) to recognise the variability in call length and for creating a dynamic estimate of back-end server load for peculiar SIP transactions. The load balancer helps to boost both the throughput and response time. The SIP is a protocol developed by knowing the growing importance with uses for VOIP (Pai et al., 1998), audio conferencing and online messaging. We present a detailed analysis of occupancy to show how the SIP algorithm significantly diminishes reduced response time.

A NETWORK PERFORMANCE COMPARISON OF WEBRTC AND SIP AUDIO AND VIDEO COMMUNICA TIONS

Multimedia services are one of the internet needs with high data traffic network count since 2019. Two of the multimedia services, Web Real-Time Communications (WebRTC) and Session Initiation Protocol (SIP), have been widely used in applications for conducting video conferencing. The main objective of this research is to analyze network performance by an application with a client-server for audio and video communications developed with WebRTC and SIP protocols. The SIP system uses the FreePBX server, and the softphone application uses Bria. Whereas WebRTC uses JavaScript with servers on Ubuntu using Node.js. The analysis application uses a star topology and runs on a local network using Wi-Fi. After testing, the results show that the throughput, jitter, and packet loss of WebRTC are better than SIP. This result is caused by several factors including the type of codec, the type of platform, and the way of signaling and routing of each protocol.

Design of P2P-SIP System Network Structure Based on JXTA

Session Initiation Protocol (SIP) is one of the most important signaling connection protocols in the field of VoIP (Voice over Internet Protocol). However, the existing SIP network has some problems such as manual configuration, difficult maintenance and poor scalability. With the development of P2P (Peer-to-Peer) technology, the combination of SIP and P2P has become a trend. JXTA is an open peer-to-peer platform, which provides basic application services for building P2P network. The topic chooses JXTA development platform to build P2P network, and then constructs SIP soft terminal on it. Based on the in-depth study of JXTA structured network and SIP protocol, a P2P SIP system is designed and implemented. The system can maintain normal SIP communication without maintaining the central server. In this paper, the detailed design and implementation process of P2P-SIP system are given, including node join and leave, user join, user search and session establishment. This paper focuses on the implementation process of user search, that is, the collaborative process of discovery service, resolution service, aggregation service and endpoint routing service. Finally, the feasibility of the system is verified by functional test.

SIP Network Secure Communication Model Based on Improved SIP Protocol

The hot topics of session initiation protocol (SIP) authentication are mainly hypertext transfer protocol (HTTP) digest authentication mechanism and public-key cryptography standards (PKI) authentication mechanism. Among them, the PKI authentication mechanism encounters problems such as long encryption and decryption time and difficult actual deployment. Aiming at the shortcomings of the existing identity authentication mechanism of SIP, start with the chaotic sequence generated by chaotic equations, the dynamic password generation is optimised during the authentication process. A SIP security authentication mechanism based on chaotic equations is proposed. Firstly, improved security authentication scheme is applied in the video intercom system, then the chaotic signal generator on the microcontroller is designed, and finally the improved security authentication scheme is applied into the system. The results show that the system can realise all the functions at design time based on the improved SIP certification.

An Improved SIP Authenticated Key Agreement Based on Dongqing et al.

The IP multimedia subsystem represents an architectural framework to support multimedia-based services using internet protocol over wired and wireless media. These IP-based multimedia services rely on session initiation protocol (SIP) for creating, maintaining and terminating the communicative sessions, which underscores the efficiency and security of SIP protocol. Many SIP based authentication schemes have been put forward in the last decade, however with many limitations. Recently, Lu et al. and Chaudhary et al. presented SIP based authentication protocols. Then, Dongqing et al. discovered limitations in Lu et al. and Chaudhary et al. schemes, and presented an improved SIP authentication protocol. Nonetheless, we ascertain that Dongqing et al.’s protocol is prone to privileged insider attack, denial of service attack, and session specific ephemeral secret-leakage attack. Besides, this protocol assumes a strictly time synchronized system, which limits the practical effectiveness of the protocol for a real environment. We also propose an improved SIP authentication protocol that covers the limitations of Dongqing et al. protocol. Our scheme is formally proved as secure using BAN logic analysis. The performance analysis illustrates the comparison for related schemes with proposed scheme, which depicts the efficiency and robustness of the scheme over previous schemes.

Detection of Intruders and Flooding in VoIP using IDS, Jacobson Fast and Hellinger Distance Algorithms

VoIP services are becoming increasingly a big competition to existing telephony services (PSTN). Hence, the need arises to protect VoIP services from all kinds of attacks that target network bandwidth, server capacity or server architectural constrains. SIP Protocol is used for VoIP connection establishment. It works based on either TCP or UDP Protocols. This protocol structure is almost as same as HTTP Protocol, i.e. for every request there will be some response, even though the request is invalid. HTTP Protocol is prone to flooding attacks, like SYN-Flood attack. Because of Session Initiation Protocol (SIP) is same as HTTP, SIP is also prone to Flooding attacks. The proposed Intrusion Detection System (IDS) is used to detect the intruders in telephony system. Genetic algorithm is used to recognize the authorized user. VoIP Flood Detection System (VFDS) is aimed to detect TCP Flooding attacks and SIP Flooding attacks on SIP devices using Jacobian Fast and Hellinger distance algorithms. The Jacobian Fast Algorithm fixes the threshold limit and Hellinger distance calculation is a statistical anomaly based algorithm uses to detect deviation in traffic

RANCANG BANGUN INFRASTRUKTUR VOIP PADA MULTIPLE NETWORK JARINGAN SOHO

Voice Over Internet Protocol (VOIP) is voice communication that uses internet channels. The main focus on VOIP is the suppression of communication costs which are generally implemented on large-scale networks. VoIP in its implementation uses a LAN network system and is supported by VoIP protocols such as standardization of SIP (Session Initiation Protocol) and IAX2 (Internet Asterisk eXchange 2). Along with the development era, VOIP infrastructure implementation began to shift to SOHO (Small Office House Office) network infrastructure, where in terms of network complexity, SOHO is a type of small-scale network. Several small companies that have just implemented computer networks put SOHO networks as an option in their companies. The SOHO infrastructure that continues to develop until now has a fairly high and diverse level of communication access on network protocols (multiple network protocol Access), approaching a large network scale. So that the VOIP design can be a form of solution in handling the problem of communication cost efficiency. The VOIP built in the study uses the Linux Trixbox CE operating system, uses the standardization of the SIP protocol (Session Initiation Protocol) as the protocol used on servers and clients with support for PSTN telephone client equipment using Panasonic KXTS505MXW and Toriphone TP - 2375CID telephones, using Analog Telephone Adapter type LinkSys PAP2T – NA . The results of the research conducted, resulted in a VOIP infrastructure design on a network that has multiple communication access with implementation on the SOHO network type. Keyword : VOIP, LAN, SOHO

Distributed intrusion detection scheme for next generation networks

IP Multimedia Subsystem (IMS) is a next generation network that provides the hypermedia services as data, voice and video to users. Due to high level requirements for IMS services, new kinds of network attacks are endlessly emerging. Thus, it is of paramount importance to protect the networks from attacks. Consequently, the Intrusion Detection Systems (IDS) are quickly becoming a popular requirement in building a network security infrastructure. Securing service and signalization is a vital feature in the IMS network. Signaling is generally based on Session Initiation Protocol (SIP) which offers numerous challenges regarding security which causes issues in IMS network. This work presents a study of SIP protocol and discovers the critical security vulnerabilities in the course of registration phase. We focused on DDoS attacks on IMS server using SIP particularly with REGISTER message and proposed a scheme based on multi agent systems for intrusion detection which takes the advantage of the distributed paradigm to implement an efficient distributed system, as well as the integration of existing techniques, i.e., the well-known IDS SNORT.

Efficient Detection of Attacks in SIP Based VoIP Networks Using Linear l1-SVM Classifier

The Session Initiation Protocol (SIP) is one of the most common protocols that are used for signaling function in Voice over IP (VoIP) networks. The SIP protocol is very popular because of its flexibility, simplicity, and easy implementation, so it is a target of many attacks. In this paper, we propose a new system to detect the Denial of Service (DoS) attacks (i.e. malformed message and invite flooding) and Spam over Internet Telephony (SPIT) attack in the SIP based VoIP networks using a linear Support Vector Machine with l1 regularization (i.e. l1-SVM) classifier. In our approach, we project the SIP messages into a very high dimensional space using string based n-gram features. Hence, a linear classifier is trained on the top of these features. Our experimental results show that the proposed system detects malformed message, invite flooding, and SPIT attacks with a high accuracy. In addition, the proposed system outperformed other systems significantly in the detection speed.

Survey on Detection Techniques for Denial-of-Service Attacks in SIP-Based Voice Over IP Networks

VoIP provides real-time voice communications over packet-switched networks. VoIP is widely used as it offers more flexibility when compared to the traditional Public Switched Telephone Network (PSTN). Session Initiation Protocol (SIP) is a protocol used for initiating a session between the caller and the callee in the VoIP system. SIP is a text-oriented protocol and is implemented in a susceptible environment. These features make SIP protocol vulnerable to various security risks such as Denial of Service, Registration hijacking and Eavesdropping. Detection of these attacks plays a vital role. The Survey focuses on different techniques used for detection of DoS attacks and differentiating between the proposed mechanisms for identifying abnormalities and then notifying the end-users about the possible intrusions.

Behavior study of SIP on IP multimedia subsystem architecture MPLS as transport layer

IP Multimedia Subsystem “IMS” is the modular architecture that provides users with reliable access to multimedia services (voice, video conferencing, interactive services, etc.). However, quality of service and security are one of the main concerns of operators when it comes to multimedia services. The Multi-Protocol Label Switching (MPLS) protocol ensures these two basic needs. IMS is primarily based on the Session Initiation Protocol (SIP) for (i) routing, (ii) establishing, (iii) modifying, and (iv) closing a multimedia session. The SIP protocol represents the most influential protocol in the IMS architecture. The objective of this paper is to evaluate the performance of the SIP protocols in an IMS architecture with the MPLS protocol as the transport layer protocol. The evaluation will focus on: (I) the impact of the number of connections on the duration of SIP session establishment and (Ii) the impact of the number of users on the performances of IMS services (VOIP and Videoconferencing).

Protecting from Cloud-based SIP flooding attacks by leveraging temporal and structural fingerprints

The session initiation protocol (SIP) is among the most popular voice over IP (VoIP) signaling protocols. Like other Internet protocols, deployment in live scenarios showed its vulnerability to flooding attacks. These attacks are very similar to those against TCP protocol but have emerged at the application level of the Internet architecture. In this paper, we present a new approach to protect SIP devices from flooding attacks. Our proposed approach is mainly composed of two algorithms: 1) a detection algorithm that takes into consideration the temporal characteristics of SIP protocol as well as the fingerprints of its messages and 2) a mitigation algorithm that filters SIP messages based on a fingerprint whitelist database. We evaluate our approach through an extensive set of experimental tests using widely distributed virtual machines in the cloud and compare to similar approaches found in the literature. The experiments emulate a large flooding attack launched from mutually distant geographic data centers. The results report short detection time, low sensibility to false alarms and high effectiveness in reducing the computational resources.

An Improved SIP Authentication Scheme Based on Server-Oriented Biometric Verification

The Session Initiation Protocol (SIP) provides a way to control the wired and wireless Voice over Internet Protocol-based communication over an insecure channel. The SIP protocol is not secure due to relying on an intrinsically open text-based communication, which further stresses the strengthening of SIP authentication protocols. Many solutions have been put forward in the last few years to design the secure and efficient SIP authentication protocols for multimedia. Recently, Zhang et al. proposed a SIP authentication protocol with an enhanced feature that enables the server authenticate the users on the basis of biometric verification. However, after a careful observation, one can witness few limitations regarding privileged insider attack, session specific temporary attack, De-synchronization attack; denial-of-service attack, inefficient password modification and lack forward secrecy compromise. We have proposed a secure scheme countering the identified flaws of Zhang et al. and other contemporary schemes. We also demonstrate the security strength of proposed scheme by employing the formal security analysis under BAN logic.

Design, implementation and performance evaluation of a proactive overload control mechanism for networks of SIP servers

The extent and diversity of systems, provided by IP networks, have made various technologies approach integrating different types of access networks and convert to the next generation network (NGN). The session initiation protocol (SIP) with respect to facilities such as being in text form, end-to-end connection, independence from the type of transmitted data, and support various forms of transmission, is an appropriate choice for signalling protocol in order to make connection between two IP network users. These advantages have made SIP be considered as a signalling protocol in IP multimedia subsystem (IMS), a proposed signalling platform for NGNs. Despite having all these advantages, SIP protocol lacks appropriate mechanism for addressing overload causing serious problems for SIP servers. SIP overload occurs when a SIP server does not have enough resources to process messages. The fact is that the performance of SIP servers is largely degraded during overload periods because of the retransmission mechanism of SIP. In this paper, we propose an advanced mechanism, which is an improved method of the windows based overload control in RFC 6357. In the windows based overload control method, the window is used to limit the amount of message generated by SIP proxy server. A distributed adaptive window-based overload control algorithm, which does not use explicit feedback from the downstream server, is proposed. The number of confirmation messages is used as a measure of the downstream server load. Thus, the proposed algorithm does not impose any additional complexity or processing on the downstream server, which is overloaded, making it a robust approach. Our proposed algorithm is developed and implemented based on an open source proxy. The results of evaluation show that proposed method could maintain the throughput close to the theoretical throughput, practically and fairly. As we know, this is the only SIP overload control mechanism, which is implemented on a real platform without using explicit feedback.

Review of SIP based DoS attacks

The Voice over Internet Protocol (VoIP). The VoIP is relatively new and is gaining more and more popularity as it offers a wide range of features and is much more cost effective as compared to the traditional PSTN. But the VoIP brings with it certain security threats which need to be resolved in order to make it a more reliable source of communication. Session Initiation Protocol (SIP) today is considered the standard protocol for multimedia signaling, and the result is a very generic protocol. SIP is specified by the IETF in RFC 3261. From a structural and functional perspective, SIP is application layer signaling text-based protocol used for creating, modifying, and terminating multimedia communications sessions among Internet endpoints. Unfortunately, SIP-based application services can suffer from various security threats as Denial of Service (DoS). attacks on a SIP based VoIP infrastructure that can severely compromise its reliability. In contrast, little work is done to analyze the robustness and reliability of SIP severs under DoS attacks. In this survey, we are discussing the DoS flooding attack on SIP server. Firstly, we present a brief overview about the SIP protocol. Then, security attacks related to SIP protocol. After that, detection techniques of SIP flooding attack and various exploited resources due to attack were discussed and finally the paper reviews previous work done on SIP based DoS attacks

Novel session initiation protocol-based distributed denial-of-service attacks and effective defense strategies

Voice-over-IP (VoIP) and its underlying session initiation protocol (SIP) techniques have become popular in recent years. VoIP/SIP techniques are used widely in unified communication systems and next generation networks, and there is no doubt they will play increasingly important roles in the future of communication techniques. However, unlike transmission control protocol (TCP)-based applications, the user datagram protocol (UDP)-based VoIP/SIP applications are not as mature and they have some security vulnerabilities. Therefore, it is crucial to study VoIP/SIP-related security issues. In this study, we investigated the existing vulnerabilities in the SIP protocol and identify new vulnerabilities in the SIP retransmission mechanisms, which could be exploited by denial-of-service (DoS)/distributed denial-of-service (DDoS) attacks. We prepared a VoIP/SIP security laboratory environment and a DDoS attack simulator. We developed two advanced attacks by exploiting the vulnerabilities identified in the SIP retransmission mechanism and we implemented these attacks in our laboratory environment using the DoS/DDoS attack simulator. Our intelligent attacks could bypass black-lists as well as IP-based rate limiting, packet count-based rate limiting, session/transaction-based rate limiting, and automatic message generation detection systems in the existing state-of-the-art security perimeters, such as firewalls, intrusion detection systems, intrusion prevention systems, and anomaly detection systems. Furthermore, we developed a novel defense mechanism to effectively combat the proposed attacks and we implemented it successfully in our VoIP/SIP security laboratory environment. We showed that our defense mechanism reduced the CPU load of a SIP server under attack from 87% down to 13.6%.

Privacy-preserving, user-centric VoIP CAPTCHA challenges

Purpose – This work aims to argue that it is possible to address discrimination issues that naturally arise in contemporary audio CAPTCHA challenges and potentially enhance the effectiveness of audio CAPTCHA systems by adapting the challenges to the user characteristics. Design/methodology/approach – A prototype has been designed, called PrivCAPTCHA, to offer privacy-preserving, user-centric CAPTCHA challenges. Anonymous credential proofs are integrated into the Session Initiation Protocol (SIP) protocol and the approach is evaluated in a real-world Voice over Internet Protocol (VoIP) environment. Findings – The results of this work indicate that it is possible to create VoIP CAPTCHA services offering privacy-preserving, user-centric challenges while maintaining sufficient efficiency. Research limitations/implications – The proposed approach was evaluated through an experimental implementation to demonstrate its feasibility. Additional features, such as appropriate user interfaces and efficiency optimisations, would be useful for a commercial product. Security measures to protect the system from attacks against the SIP protocol would be useful to counteract the effects of the introduced overhead. Future research could investigate the use of this approach on non-audio CAPTCHA services. Practical implications – PrivCAPTCHA is expected to achieve fairer, non-discriminating CAPTCHA services while protecting the user’s privacy. Adoption success relies upon the general need for employment of privacy-preserving practices in electronic interactions. Social implications – This approach is expected to enhance the quality of life of users, who will now receive CAPTCHA challenges closer to their characteristics. This applies especially to users with disabilities. Additionally, as a privacy-preserving service, this approach is expected to increase trust during the use of services that use it. Originality/value – To the best of authors’ knowledge, this is the first comprehensive proposal for privacy-preserving CAPTCHA challenge adaptation. The proposed system aims at providing an improved CAPTCHA service that is more appropriate for and trusted by human users.