Session Initiation Protocol (SIP) arises as a new standard of establishing and releasing connections for vast variety of multimedia applications. The protocol may be used for voice calls, video calls, video conferencing, gaming and many more. The 3GPP (3rd Generation Partnership Project) suggests SIP as the signalling solution for 3rd generation telephony. Thereby, this purely IP-centric protocol appears as a promising alternative to older signalling systems such as H.323, SS7 or analog signals in PSTN. In contrast to them, SIP does not focus on communication with PSTN network. It is more similar to HTTP than to any of the mentioned protocols. The main standardisation body behind Session Initiation Protocol is The Internet Engineering Task Force (IETF). The most recent paper published on SIP is RFC 3261 [5]. Moreover, there are working groups within IETF that publish suggestions and extensions to the main standard. One of those extensions is “Caller Preferences for the Session Initiation Protocol (SIP)” [1]. This document describes a set of new rules that allow a caller to express preferences about request handling in servers. They give ability to select which Uniform Resource Identifiers (URI) a request gets routed to, and to specify certain request handling directives in proxies and redirect servers. It does so by defining three new request header fields, Accept-Contact, Reject-Contact, and Request-Disposition, which specify the caller preferences. [1]. The aim of this project is to extend the existing software with caller preferences and evaluate the new functionality.

This document is the final report of the master's thesis "Implementation and Evuation of the Service Peer Discovery Protocol", carried out at the Center for Wireess Systems, KTH, Stockholm. This thesis addresses the problem of service discovery in peer-to-peer mobile networks by implementing and evaluating a previously designed protocl (the Service Peer Discovery Protocol). The main feature of peer-to-peer networks is that users connected to them can communicate directly with each other, without the necessity of interaction via a central point. However, in order for two networks users (ir peers) to communicate, they must have a means to locate and address each other, which is in gernal called a discovery protocol. There are many different solutions for discoverying protocols that work efficiently in fixed or slow-moving networks, but full mobility introduces a set of new difficulties for the discovery of peers and their services. The potential changes in location, which can occur very ofter, the changes in IP address that these changes cuase, and roaming between networks of different kinds are good examples of these difficulties. To solve these problems, a new Service Peer Discovery Protocol was designed and a test application built. The next step towards the introduction of this protocol was creating a working implementation, setting up a suitable test environment, performing experiments, and evaluating its performance. This evaluation could lead to improvments in the protoocl. The aim of this thesis is to implement and document the Service Peer Discovery Protocol, to carry out measurements of it, to evaluate the efficiency of the protocol, and to suggest ways in which it could be improved. The Service Peer Discovery Protocol was found to be well targeted to wireless, peer-to-peer networks, althgouh improvements in the protocol could make it more time and traffic-efficient while maintaining the same level of performance. / Detta är den slutliga rapporten för examensarbetet "Implementation och utvädering av Service Peer Discovery Protocol", utfört på Center for Wireless Systems, KTH, Stockholm.  Uppsatsen behandlar problemet med sökning efter tjänster i icke-hierarkiska (peer-to-peer) mobila nätverk genom att implementera och utvädera ett redan konstruerat protokoll (Service Peer Discovery Protocol). Den huvudsakliga fördelen med icke-hierarkiska nätverk är att anslutna anvndare (parter) kan kommunicera direkt med varandra, utan att behöva interagera med en central punkt.  Dock måste metoder för att lokalisera och adressera andra parter vara tillgängliga för att parterna skall kunna kommunicera, metoder som kalla sökprotokoll (discovery protocol). Det finns många olika sökprotokollösningar som fungerar effektivt i fasta eller långsamma mobila nätverk, men med full mobilitet introduceras ett antal nya svårigheter vid s kande efter parter och tjänster. Den potentiella förändringen av position (vilken kan inträffa ofta), byte av IP-address som dessa förändringar medför, och förflyttning mellan olika typer av nätverk, är exempel på sådana svårigheter. För att lösa dessa problem, konstruerades protokollet Service Peer Discovery Protocol och en testapplikation byggdes.  Nästa steg mot en introducering av detta protokoll var en fungerande implementation, en lämplig testmilö, utförandet av tester och en utvädering av prestandan.  Utväderingen syftade till att förbättra protokollet.  Syftet med detta examensar1ete är att implementera och dokumentera protokollet Service Peer Discovery Protocol, att göra mätningar, att utvädera effektiviteten samt att föreslå förbättringar av protokollet. Service Peer Discovery Protocol, fanns vara väl anpassat till icke-hierarkiska trådlösa nätverk.  Dock torde förbättringar av protokollet innebära tidseffektivare och trafikeffektivare beteende utan att kompromissa prestandanivån.

This Master thesis work was carried out at theWireless Center at KTH and it is part of a pilot project. This thesis is conducted for the Institute for Microelectronics and Information Technology (IMIT) at the Royal Institute of Technology (KTH) in Stockholm (Sweden) and for the Department of Telecommunications at Politecnico di Torino in Turin (Italy). This thesis addresses an area with significant potential for offering services to mobile users. In such a scenario users should have minimal interaction with applications which, by taking into account available context information, should be able to make decisions, such as setting up delivery paths between peers without requiring a third party for the negotiation. In wireless reconfigurable networks, the mobile users are on the move and must deal with dynamic changes of network resources. In such a network, mobile users should be able to contact other peers or resources by using the current route. Thus although manual configuration of the network is a possible solution, it is not easily used because of the dynamic properties of the system which would demand too much user interaction. However, existing discovery protocols fall short of accomodating the complexity of reconfigurable and heterogeneous networks. The primary objective of this thesis work was to investigate a new approach at the application level for signaling by taking advantage of SIP’s features. The Session Initiation Protocol (SIP) is used to provide naming and localization of the user, and to provide functionality to invite users to establish sessions and to agree on communication parameters. The Specific Event Notification of the SIP protocol provides a framework for the notification of specific events and I believed that it could be instantiated as solution to the problem for reconfigurable application networks. This thesis proposes a method for providing localization information to SIP User Agents in order to establish sessions for service discovery. Furthermore, this method should consider context meta-data to design strategies effective in heterogeneous networks. A viable solution must support (re)location of users at the application layer when they roam between different wireless networks, such as GPRS and WLAN. An analysis of the implications of the proposed model is presented; in this analysis emphasis has been placed on how this model interacts with existing services.

Les technologies de communication et les services web sont devenus disponibles dans notre vie numérique, les réseaux informatiques continuent de croître et de nouveaux protocoles de communication sont constamment définis et développés. Par la suite, la standardisation et la normalisation des protocoles sont dispensables pour permettre aux différents systèmes de dialoguer. Bien que ces normes peuvent être formellement vérifiés, les développeurs peuvent produire des erreurs conduisant à des implémentations défectueuses. C'est la raison pour laquelle leur mise en œuvre doit être strictement examinée. Cependant, la plupart des approches de tests actuels exigent une stimulation de l’exécution dans le cadre des tests (IUT). Si le système ne peut être consulté ou interrompu, l'IUT ne sera pas en mesure d'être testé. En outre, la plupart des travaux existants sont basées sur des modèles formels et très peu de travaux s'intéressent à la formalisation des exigences de performance. Pour résoudre ces problèmes, nous avons proposé une approche de test basé sur la logique "Horn" afin de tester passivement la conformité et la performance des protocoles. Dans notre approche, les exigences peuvent être formalisées avec précision. Ces exigences formelles sont également testées par des millions de messages collectés à partir des communicants réels. Les résultats satisfaisants des expériences effectuées ont prouvé le bon fonctionnement et l'efficacité de notre approche. Aussi pour satisfaire les besoins croissants de tests distribués en temps réel, nous avons également proposé un cadre de tests distribués et un cadre de tests en ligne et nous avons mis en œuvre notre plateforme dans un environnement réel à petite échelle avec succès / While today’s communications are essential and a huge set of services is available online, computer networks continue to grow and novel communication protocols are continuously being defined and developed. De facto, protocol standards are required to allow different systems to interwork. Though these standards can be formally verified, the developers may produce some errors leading to faulty implementations. That is the reason why their implementations must be strictly tested. However, most current testing approaches require a stimulation of the implementation under tests (IUT). If the system cannot be accessed or interrupted, the IUT will not be able to be tested. Besides, most of the existing works are based on formal models and quite few works study formalizing performance requirements. To solve these issues, we proposed a novel logic-based testing approach to test the protocol conformance and performance passively. In our approach, conformance and performance requirements can be accurately formalized using the Horn-Logic based syntax and semantics. These formalized requirements are also tested through millions of messages collected from real communicating environments. The satisfying results returned from the experiments proved the functionality and efficiency of our approach. Also for satisfying the increasing needs in real-time distributed testing, we also proposed a distributed testing framework and an online testing framework, and performed the frameworks in a real small scale environment. The preliminary results are obtained with success. And also, applying our approach under billions of messages and optimizing the algorithm will be our future works

Background. SIP and RTP are two protocols that are widely used, and they play an important role in VoIP services. VoIP is an integral part of many communication services, e.g., Microsoft Teams, Skype, Discord, and communications over cellular networks (VoLTE and VoWiFi). Since these technologies are so widely used, a high level of security is paramount. Objectives. The aim of this study is threefold: (1) To investigate if it is possible to create a penetration testing methodology for SIP and RTP, where the target group is penetration testers with no previous knowledge of these protocols. (2) To identify previously discovered vulnerabilities and attacks. (3) Due to the lack of domain experts, a methodology of this kind will hopefully help penetration testers without prior knowledge, easing them into a new work area. Further, the aim is to increase awareness of potential vulnerabilities in such systems. Methods. Through a literature review, threat modeling, and exploratory penetration testing on three different testbeds, several vulnerabilities and attacks were identified and validated. From the results, a methodology was compiled. For evaluation purposes, it was evaluated by a third party, who tested it on a testbed and gave feedback. Results. The results from our research show that SIP and RTP are susceptible to a wide array of different attacks even to this day. From our literature study, it was determined that most of these attacks have been known for a long time. Using exploratory penetration testing, we managed to verify most of these attacks on three different systems. Additionally, we discovered a few novel attacks that we did not find in previous research. Conclusions. Our literature study suggests that SIP and RTP based systems are relatively susceptible to multiple attacks. Something we also validated during the exploratory testing phase. We successfully executed multiple existing attacks and some new attacks on three different testbeds. The methodology received mostly positive feedback. The results show that many of the participants appreciated the simplicity and concrete model of the methodology. Due to the low number of participants in the evaluation, an improvement to the study and results would be to increase the population and also have multiple novice penetration testers test several different systems. An increase in the number of testbeds would also further support the results and help generalize the methodology. / Bakgrund. SIP och RTP  är två protokoll som är vitt använda och spelar en väldigt viktig roll i VoIP-tjänster. VoIP utgör en viktig del i många kommunikationstjänster, t.ex. Microsoft Teams, Skype och Discord, men även i kommunikation över mobilnätet (VoLTE och VoWiFi). Eftersom dessa teknologier används i så stor utsträckning, är säkerhet av största vikt. Syfte. Syftet med denna studie är trefaldig: (1) Undersöka om det är möjligt att utforma en penetration testningsmetod för SIP och RTP, för en målgrupp av penetrationstestare utan förkunskaper kring dessa protokoll. (2) Att identifiera sårbarheter och attacker från tidigare studier. (3) På grund av brist på kompentens inom området penetrationstestning och telekommunikation kan en sådan här metod förhoppningsvis hjälpa till att introducera penetrationstestare utan tidigare erfarenhet till det här specifika området. Ytterligare är också målet att att öka medvetheten när det kommer till sårbarheter i sådana system. Metod. Genom en literaturstudie, hotmodellering och utforskande penetrationstestning på tre olika testmiljöer har ett flertal sårbarheter och attacker identifieras och utförts. Från resultatet utformades en metod för penetrationstesning, som sedan evaluerades genom att en tredje part testade metoden och gav återkoppling som rör metodens format och struktur. Resultat. Resultaten från vår studie visar att SIP och RTP är sårbara för en rad olika attacker än idag. Resultaten från vår litteraturstudie visar att många av dessa attacker har varit kända under en lång tid. Vi lyckades verifiera de flesta av dessa attacker genom utforskande penetationstestning på tre olika system. Dessutom lyckades vi identifiera ett antal nya attacker som inte tidigare nämnts i forskning inom området. Slutsatser. Resultaten från vår litteraturstudie visar att system som använder sig av SIP och RTP är relativt sårbara för en mängd olika attacker. Detta bekräftades i den utforskande testningen, där ett flertal kända samt nya attacker utfördes framgångsrikt. Den interna evalueringen i studien visar på att metoden kan appliceras framgångsrikt på ett flertal olika system, med begränsningen att endast tre system testats. Resultaten från den externa evalueringen, där penetrationstestare blev tillfrågade att utvärdera och testa metoden visar att de hade en relativt positiv inställning till metoden. För att ytterligare underbygga detta påstående krävs en större population, både för testningen och utvärderingen. Det krävs också att en större mängd testmiljöer används för att kunna generalisera metoden.

Voice Over IP (VoIP) telephony is becoming widespread, and is often integrated into computer networks. Because of his, it is likely that malicious software will threaten VoIP systems the same way traditional computer systems have been attacked by viruses, worms, and other automated agents. While most users have become familiar with email spam and viruses in email attachments, spam and malicious traffic over telephony currently is a relatively unknown threat. VoIP networks are a challenge to secure against such malware as much of the network intelligence is focused on the edge devices and access environment. A novel security architecture is being developed which improves the security of a large VoIP network with many inexperienced users, such as non-IT office workers or telecommunication service customers. The new architecture establishes interaction between the VoIP backend and the end users, thus providing information about ongoing and unknown attacks to all users. An evaluation of the effectiveness and performance of different implementations of this architecture is done using virtual machines and network simulation software to emulate vulnerable clients and servers through providing apparent attack vectors.

With the convergence of the wireless networks and the Internet and the booming demand for multimedia applications, the next-generation (beyond the third generation, or B3G) wireless systems are expected to be all IP-based and provide real-time and non-real-time mobile services anywhere and anytime. Powerful and efficient mobility support is thus the key enabler to fulfil such an attractive vision by supporting various mobility scenarios. This thesis contributes to this interesting while challenging topic. After a literature review on mobility support architectures and protocols, the thesis starts presenting our contributions with a generic multi-layer mobility support framework, which provides a general approach to meet the challenges of handling comprehensive mobility issues. The cross-layer design methodology is introduced to coordinate the protocol layers for optimised system design. Particularly, a flexible and efficient cross-layer signalling scheme is proposed for interlayer interactions. The proposed generic framework is then narrowed down with several fundamental building blocks identified to be focused on as follows. As widely adopted, we assume that the IP-based access networks are organised into administrative domains, which are inter-connected through a global IP-based wired core network. For a mobile user who roams from one domain to another, macro (inter-domain) mobility management should be in place for global location tracking and effective handoff support for both real-time and non-real-lime applications. Mobile IP (MIP) and the Session Initiation Protocol (SIP) are being adopted as the two dominant standard-based macro-mobility architectures, each of which has mobility entities and messages in its own right. The work explores the joint optimisations and interactions of MIP and SIP when utilising the complementary power of both protocols. Two distinctive integrated MIP-SIP architectures are designed and evaluated, compared with their hybrid alternatives and other approaches. The overall analytical and simulation results shown significant performance improvements in terms of cost-efficiency, among other metrics. Subsequently, for the micro (intra-domain) mobility scenario where a mobile user moves across IP subnets within a domain, a micro mobility management architecture is needed to support fast handoffs and constrain signalling messaging loads incurred by intra-domain movements within the domain. The Hierarchical MIPv6 (HMIPv6) and the Fast Handovers for MIPv6 (FMIPv6) protocols are selected to fulfil the design requirements. The work proposes enhancements to these protocols and combines them in an optimised way. resulting in notably improved performances in contrast to a number of alternative approaches.

ABSTRACT SIMULATION BASED INVESTIGATION OF AN IMPROVEMENT FOR FASTER SIP RE-REGISTRATION TANRIVERDi, Eda M.Sc., Department of Electrical and Electronics Engineering Supervisor: Prof. Dr. Semih BiLGEN July 2004, 78 pages In this thesis, the Session Initiation Protocol (SIP) is studied and an improvement for faster re-registration is proposed. This proposal, namely the &ldquo / registration &ndash / activation&rdquo / , is investigated with a simulation prepared using OPNET. The literature about wireless mobile networks and SIP mobility is reviewed. Conditions for an effective mobile SIP network simulation are designed using message sequence charts. The testbed in [1] formed by Dutta et. al. that has been used to observe SIP handover performance is simulated and validated. The mobile nodes, SIP Proxy v servers, DHCP servers and network topology are simulated on &ldquo / OPNET Modeler Radio&rdquo / . Once the simulation is proven to be valid, the &ldquo / registration &ndash / activation&rdquo / is implemented. Different simulation scenarios are set up and run, with different mobile node speeds and different numbers of mobile nodes. The results show that the re-registration delay is improved by applying the &ldquo / registration &ndash / activation&rdquo / but the percentage of improvement depends on the improvement in the database access delay in the SIP Proxy server.

High-performance networking (HPN) is of significance today in order to enable next-generation applications using wired and wireless networks. Some of the examples of HPN include low-latency industrial sensing, monitoring and automation using Wireless Sensor Networks (WSNs). HPN however requires protocol optimization at many layers of the open system interface (OSI) network model in order to meet the stringent performance constraints of the given applications. Furthermore, these protocols need to be impervious to denial of service (DoS) and distributed DoS (DDoS) attacks. Some of the key performance aspects of HPN are low point-to-point and end-to-end latency, high reliability of transmitted frames and performance predictability under various network load situations. This work focuses on two discrete issues in designing protocols for HPN applications. The first research issue looks at the Medium Access Control (MAC) layer of the OSI network model for designing of MAC protocols that provide low-latency and high reliability for point-to-point communication under a WSN. Existing standards in this area are governed by IEEE 802.15.4 specification which defines protocols for MAC and PHY layers for short-range, low bit-rate, and low-cost wireless networks. However, the IEEE 802.15.4 specification is inefficient in terms of latency and reliability performance and, as a result, is unable to meet the stringent operational requirements as defined by counterpart wired sensor networks. Work presented under current research issue describes new MAC protocols that are able to show low-latency transmission performance under strict timing constants for power limited WSNs. This enhancement of the MAC protocols is named extended GTS (XGTS) contained under extended CFP (ECFP) and is published under the IEEE's 802.15.4e standard. The second research issue focuses on the application layer of the OSI network model to design protocols that enhance the robustness of the text based protocols to various traffic inputs. The purpose of this is to increase the reliability of the given text based application layer protocol under a varied load. Session Initiation Protocol (SIP) is used as a case study and the work aims to build algorithms that ensure that SIP can continue to function under specific traffic conditions, which would otherwise deem the protocol useless due to DoS and DDoS attacks. Proposed algorithms investigate techniques that enhance the robustness of the SIP against parsing attacks without performing a deep parse of the protocol data unit (PDU). The desired effect of this is to reduce the time spent in parsing the SIP messages at a SIP router and as a result increase the number of SIP messages processed per unit time at a SIP router.

Modern technologies are growing and Voice over Internet Protocol (VoIP) technology is able to function in heterogeneous networks. VoIP gained wide popularity because it offers cheap calling rates compared to traditional telephone system and the number of VoIP subscribers has increased significantly in recent years. End users need reliable and acceptable call quality in real time communication with best Quality of Service (QoS). Server complexity is increasing to handle all client requests simultaneously and needs huge processing power. VoIP Servers will increase processing power but the engineering tradeoff needs to be considered e.g. increasing hardware will increase hardware complexity, energy consumption, network management, space requirement and overall system complexity. Modern System-on-Chip (SoC) uses multiple core technology to resolve the complexity of hardware computation. With enterprises needing to reduce overall costs while simultaneously improving call setup time, the amalgamation of VoIP with SoC can play a major role in the business market. The proposed VoIP Server model with multiple processing capabilities embedded in it is tailored for multicore hardware to achieve the required result. The model uses SystemC-2.2.0 and TLM-2.0 as a platform and consists of three main modules. TLM is built on top of SystemC in an overlay architectural fashion. SystemC provides a bridge between software and hardware co-design and increases HW & SW productivity, driven by fast concurrent programming in real time. The proposed multicore VoIP Server model implements a round robin algorithm to distribute transactions between cores and clients via Load Balancer. Primary focus of the multicore model is the processing of call setup time delays on a VoIP Server. Experiments were performed using OpenSIP Server to measure Session Initiation Protocol (SIP) messages and call setup time processing delays. Simulations were performed at the KTH Ferlin system and based on the theoretical measurements from the OpenSIP Server experiments. Results of the proposed multicore VoIP Server model shows improvement in the processing of call setup time delays.