Session Initiation Protocol

Session initiation protocol (SIP) is a well-known protocol that is used to provide voice over IP (VoIP) signaling and multimedia services in today's evolving Internet. With the popularity of peer-to-peer (P2P) system, it is envisioned to use SIP in P2P networks, to increase the communication flexibility without a centralized organization and control. The REsource LOcation And Discovery (RELOAD) protocol is used to facilitate SIP signaling in the P2P overlay network. RELOAD with interactive connectivity establishment (ICE) functionality in P2P SIP is used to improve the performance of the SIP under varying network conditions. The aim of this paper is to examine the ICE working mechanism in P2P SIP and explain the operational functionality of ICE in a P2P SIP overlay networking environment. Also, this paper discusses different working methods, challenges and problems that are used by the RELOAD protocol for establishing peer-to-peer SIP connections.

The Session Initiation Protocol (SIP) is a signaling protocol for managing various types of real time sessions between parties over an Internet Protocol network. An open issue is the control of overload situations that occur when the incoming flow of requests to a SIP server overcomes the server processing capacity. In particular, call establishment times increase due to overload, which triggers retransmissions and causes a further increase of the total incoming flow of requests. This paper proposes an overload control system for regulating both the queue length and the CPU load of the SIP server. The proposed control system is made of two PI controllers tuned by minimizing a proper cost function using the Extremum Seeking algorithm. A real working implementation of the SIP overload controller has been made in the open source SIP server Kamailio. A performance evaluation and comparison of the proposed controller with the main proposals existing in literature has been carried out. Results show that the proposed control system counteracts overload situations and provides a goodput close to the optimal while maintaining low call establishment delays and retransmission ratios.

Voice Over Internet Protocol (VoIP) adalah salah satu contoh  perkembangan teknologi komunikasi dan informasi saat ini. Teknologi VoIP sangat menguntungkan karena menggunakan jaringan berbasis IP yang sudah memiliki jaringan kuat di dunia sehingga biaya untuk melakukan panggilan jauh lebih efisien daripada menggunakan telepon analog. Dengan memanfaatkan jaringan komputer yang sudah ada memungkinkan teknologi VoIP dapat diimplementasikan di Universitas Sam Ratulangi Namun yang menjadi masalah dalam penggunaan teknologi ini adalah banyaknya persepsi yang menyatakan bahwa kualitas suara pada percakapan VoIP masih tergolong buruk. Hal ini banyak disebabkan oleh penggunaan codec yang tidak sesuai dengan kapasitas jaringan dan masalah pada jaringan IP yang digunakan yang pada akhirnya menyebabkan terjadinya delay, jitter, dan packet loss .Oleh karena itu perlu dilakukan analisis sebelum teknologi ini dijalankan. Dengan alasan di atas penulis melakukan analisis dari segi teknis mulai dari ...

Primeiramente, gostaríamos de agradecer aos nossos pais, avós e demais familiares por terem dado todo o suporte necessário durante esta trajetória. Com o auxílio e palavras de conforto nos momentos difíceis, eles tornaram essa caminhada menos árdua e mais gratificante. Às nossas namoradas, pelo amor, carinho e compreensão sempre que foi necessário abdicar dos momentos juntos e focar na realização deste trabalho. Aos nossos colegas de turma, pelo companheirismo apresentado nas horas de estudo e pelos momentos de descontração nas horas apropriadas. Ao nosso orientador, Prof. Djan Rosário, pela confiança, paciência, dedicação e orientação. À Prof. Sheila Santisi Travessa, pelo suporte e dedicação oferecido em todo o período de desenvolvimento deste trabalho. Aos membros da banca examinadora, por aceitarem o convite. "Temos o destino que merecemos. O nosso destino está de acordo com nossos méritos." (Albert Einstein). RESUMO O VoIP é uma tecnologia cada vez mais difundida e utilizada nos dias atuais. Nas organizações já virou uma tendência e vem aumentando o seu crescimento entre usuários com acesso à Internet. Devido a sua ascensão, inúmeras evoluções são realizadas para melhorar a comunicação, destacando-se a criação do protocolo SIP (Session Initial Protocol) capaz de iniciar, alterar e finalizar sessões multimídia através de mensagens transmitidas em formato texto pela rede. Apesar das vantagens oferecidas por esta tecnologia, ela trouxe algumas preocupações, principalmente relacionadas à integridade das informações disponibilizadas no protocolo SIP. Ataques que se utilizam das vulnerabilidades existentes neste protocolo podem gerar diversos danos, e trazer prejuízos a quem utiliza este tipo de comunicação. Neste sentido, o presente trabalho teve como objetivo estudar e analisar algumas das vulnerabilidades presentes nos sistemas VoIP, focando apenas nos ataques ao protocolo SIP. Esta análise foi realizada através de um cenário montado para simular três vulnerabilidades estudadas no decorrer do desenvolvimento, são elas: roubo de registro, ataque de flood e mensagens maliciosas. Como resultado dessas simulações, ficou visível que o SIP é um protocolo bastante frágil quando está desprotegido, e essas fragilidades podem causar graves danos na estrutura em que ele fornece comunicação. Conclui-se então que, plataformas VoIP que utilizam o SIP para transporte das informações, devem ser implantadas levando em consideração uma maior proteção deste protocolo, evitando assim, que usuários maliciosos venham a desfrutar das informações do mesmo. Palavras-chave: SIP, vulnerabilidades, ataque, segurança e VoIP.

Modern feature-rich telecommunications services offer significant opportunities to human users. To make these services more usable, facilitating personalisation is very important. Such personalisation enhances the users' experience considerably. The Session Initiation Protocol and Distributed Feature Composition architecture allow users to select and compose telecommunications network applications or features. In this paper we view feature composition as a configuration problem. We model feature composition using a ...

In this paper we look at the signaling for session mobility using the Session Initiation Protocol (SIP) and the Session Description Protocol (SDP). Bearing in mind that perceived voice quality is sensitive mostly to delay variation (also called jitter), we work on bounding that value for providing a better voice Quality of Service over Wireless LAN (WLAN). For mobility we use a hybrid SIP-MIP stack. The basic idea is to combine the strengths of MIP for doing fast handovers and the strengths of SIP for powerful session adaptation capabilities. Numerical calculation figures as well as real implementation results are provided. It can be observed that the session update cycle during mobility can be modified in such a way that session delay variation bounds achieved are quite low and there are almost no spikes. We use composite performance metrics that we defined in previous work to analyze the correlation between the data patterns for session signaling delay in the classical case and our proposed case to better highlight the achieved performance delta.

Intelligent precision agriculture incorporates a number of Internet of Things (IoT) devices and drones to supervise agricultural activities and surroundings. The collected data are then forwarded to processing centers to facilitate crucial decisions. This can potentially help optimize the usage of agricultural resources and thwart disasters, enhancing productivity and profitability. To facilitate monitoring and decision, the smart devices in precision agriculture must exchange massive amounts of data across the open wireless communication channels. This inadvertently introduces a number of vulnerabilities, exposing the collected data to numerous security and privacy threats. To address these issues, massive security solutions have been introduced to secure the communication process in precision agriculture. However, most of the current security solutions either fail to offer perfect protection or are inefficient. In this paper, a scheme deploying efficient cryptographic primitives such as hashing, exclusive OR and random number generators is presented. We utilize the Burrows-Abadi-Needham (BAN) logic to demonstrate the verifiable security of the negotiated session keys. In addition, we execute an extensive semantic analysis which reveals the robustness of our scheme against a myriad of threats. Moreover, comparative performance evaluations demonstrate its computation overheads and energy consumption efficiency.

In this paper, we evaluate available steganographic techniques for SIP (Session Initiation Protocol) that can be used for creating covert channels during signaling phase of VoIP (Voice over IP) call. Apart from characterizing existing steganographic methods we provide new insights by introducing new techniques. We also estimate amount of data that can be transferred in signalling messages for typical IP telephony call.

The Session Initiation Protocol (SIP) is the principal signalling protocol in Voice over IP (VoIP) systems, responsible for initialising, terminating, and maintaining sessions amongst call parties. However, the problem with the SIP protocol is that it was not designed to be secure by nature as the HTTP digest authentication used in SIP is insecure, making it vulnerable to a variety of attacks. The current solutions rely on several standardised encryption protocols, such as TLS and IPsec, to protect SIP registration messages. However, the current centralised solutions do not scale well and cause algorithm overload when encoding and decoding SIP messages. In trying to rectify this issue, we propose in this paper a blockchain-based lightweight authentication mechanism, which involves a decentralised identity model to authenticate the SIP client to the SIP server. Our mechanism uses a smart contract on the Ethereum blockchain to ensure trust, accountability and preserves user privacy. We provided a proofof-concept implementation to demonstrate our work. Further analysis of this approach's usability, mainly CPU and memory usage, was conducted comparing to IPsec and TLS. Then we discussed our system's security and presented a security analysis. Our analysis proves that our approach satisfies the SIP protocol security requirements. Keywords-Voice over IP (VoIP), SIP protocol, Blockchain technology.

Real-time communication has witnessed a dramatic increase in recent years in user daily usage. In this domain, Session Initiation Protocol (SIP) is a well-known protocol found to provide trusted services (voice or video) to end users along with efficiency, scalability, and interoperability. Just like other Internet technology, SIP stores its related data in databases with a predefined data structure. In recent, SIP technologies have adopted the real advantages of in-memory databases as cache systems to ensure fast database operations during real-time communication. Meanwhile, in industry, there are several names of in-memory databases that have been implemented with different structures (e.g., query types, data structure, persistency, and key/value size). However, there are limited resources and poor recommendations on how to select a proper in-memory database in SIP communications. This paper provides recommended and efficient in-memory databases which are most fitted to SIP servers by evaluating three types of databases including Memcache, Redis, and Local (OpenSIPS built-in). The evaluation has been conducted based on the experimental performance of the impact of in-memory operations (store and fetch) against the SIP server by applying heavy load traffic through different scenarios. To sum up, evaluation results show that the Local database consumed less memory compared to Memcached and Redis for read and write operations. While persistency was considered, Memcache is the preferable database selection due to its 25.20 KB/s for throughput and 0.763 s of call-response time.

The Session Initiation Protocol (SIP) retransmission mechanism is designed to maintain reliable transmission over lossy or faulty network conditions. However, the retransmission can amplify the traffic overload faced by the SIP servers. In this paper, by modeling the interaction between an overloaded downstream server and its upstream server as a feedback control system, we propose two Proportional-Integral (PI) control algorithms to mitigate the overload by regulating the retransmission rate in the upstream server. We provide the design guidelines for both overload control algorithms to ensure the system stability. Our OPNET simulation results demonstrate that: (1) without the control algorithm applied, the overload at a downstream server may propagate to its upstream servers and cause widespread network failure; (2) in case of short-term overload, both proposed feedback control solutions can mitigate the overload effectively without rejecting calls or reducing resource utilization, thus avoiding the disadvantages of existing overload control solutions for SIP networks.

In this paper, embedded software engineering approach is employed for the development of a SIPbased VoIP server. Signaling control functions of SIP servers are integrated and implemented in an embedded system platform. This embedded SIP server also serves as network access server (NAS) client to a centralized RADIUS server to provide user's authorization, authentication, and accounting information. Other advanced telephony features, such as voice conferences, call waiting, call on-hold, etc…, can be designed in the embedded SIP server as well. In this study, the embedded software architecture and development procedure for the embedded SIP server such as hardware platform, the embedded software function, development procedures, and functional test are described. This embedded SIP server can be beneficial for frequent computer users with larger telephone usage, such as people working in global branch offices of enterprises, schools on the academic communication backbone, or research institutes among Universities on an academic broadband infrastructure.

Voice over Internet Protocol (VoIP) facilitates transmission of voice and video signals instantly over Internet Protocol (IP) [1]. It has a low cost of building infrastructure and low operation charges. Therefore, in terms of service charges and infrastructure implementation, it enables video/audio calls over a longer distance to be cheaper than traditional communication [2]. VoIP communication uses Session Initiation Protocol (SIP) for initiating, maintaining, and ending audio/video calls [3]. However, SIP is vulnerable because it sends plain text messages. Hackers can access and change information present in SIP headers [4]. Also, it lacks the data necessary to verify a caller [5]. As a result, by exploiting SIP

The Quality for Service Protocol draft-aranda-dispatch-q4s-02.txt Status of this Memo This Internet-Draft is submitted to IETF in full conformance with the provisions of BCP 78 and BCP 79. Internet-Drafts are working documents of the Internet Engineering Task Force (IETF), its areas, and its working groups. Note that other groups may also distribute working documents as Internet-Drafts. Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress."

This document describes a set of scenarios in which complications when identifying which policy to apply for a host are encountered. This problem is abstracted as "host identification". Describing these scenarios allows commonalities between scenarios to be identified, which is helpful during the solution design phase. This document does not include any solution-specific discussions. IESG Note This document describes use cases where IP addresses are overloaded with both location and identity properties. Such semantic overloading is seen as a contributor to a variety of issues within the routing system [RFC4984]. Additionally, these use cases may be seen as a way to justify solutions that are not consistent with IETF Best Current Practices on protecting privacy [BCP160] [BCP188]. Status of This Memo This document is not an Internet Standards Track specification; it is published for informational purposes. This is a contribution to the RFC Series, independently of any other RFC stream. The RFC Editor has chosen to publish this document at its discretion and makes no statement about its value for implementation or deployment. Documents approved for publication by the RFC Editor are not a candidate for any level of Internet Standard; see Section 2 of RFC 5741. Information about the current status of this document, any errata, and how to provide feedback on it may be obtained at http://www.rfc-editor.org/info/rfc7620.

This paper introduces a new paging technique to track and wake up a mobile node (MN) attached to an access point (AP) in a wireless LAN network after a session initiation protocol (SIP) INVITE message is initiated by a caller. A tracking agent (TA) keeps track of the mobiles' handoffs between the APs. A paging agent (PA) triggers the TA to page the mobile when a SIP INVITE is received for one of its users. The context transfer feature of our paging protocol allows the paging messages to deliver the station context in order to enable faster session reestablishment. The AP then does onlink paging in a wireless link. SIP extensions are needed to trigger the PA to start paging MNs to notify their dormant status using an extended SIP REGISTER method. Tracking protocol is analyzed to compare soft-and hard-state approaches for state inconsistency ratio, message rate, and the overall cost. The simulation model we developed enables us to evaluate the traffic introduced by the tracking protocol and the cache (state) size. Paging protocol is analyzed for CPU processing times and the transmission delays in the SIP session setup with paging. Simulation of the paging with context transfer is used to show the gains in reauthentication. Index Terms-Context transfer, fluid flow and random walk mobility model, onlink paging, paging agent (PA), session initiation protocol (SIP), tracking agent (TA).

Voice Over Internet Protocol (VoIP) adalah sebuah teknologi yang mampu melewatkan trafik suara, video dan data yang berbentuk paket melalui jaringan IP. Penggunaan jaringan IP memungkinkan penghematan biaya dikarenakan tidak perlu membuat infrastruktur baru untuk komunikasi suara dan penggunaan lebar bandwith yang lebih kecil dibandingkan dengan telepon biasa. Penerapan sistem VoIP khususnya dalam penelitian ini penerapan sistem VoIP pada sekolah dengan memanfaatkan jaringan Local Area Network yang ada pada SMK AL -HIKMAH Curug Kab.Tangerang yang ada menggunakan beberapa hardware dan software tambahan seperti router MikroTik dan Access Point outdoor serta software Mini SIP Server dan Zoiper Softphone. Router MikroTik disini berfungsi sebagai DHCP Server untuk mengurangi memory load pada Access Point outdoor sehingga dapat berjalan dengan maksimal, dan untuk media transmisi pada sistem ini menggunakan Access Point outdoor dan kabel UTP sebagai jalur pengiriman paket suara pada sistem . Software Mini SIP Server menggunakan protokol Session Initation Protocol (SIP) berfungsi sebagai server VoIP yang memproses/memonitoring panggilan yang berada pada sistem VoIP, yang nanti nya sinyal VoIP akan di sebar dengan Access Point outdoor. Untuk endpoint di sini untuk memulai dan mengakhiri panggilan mengunakan softphone Zoiper.

Session Initiation Protocol (SIP) is a new signaling protocol designed to establish multimedia sessions in telecommunication networks. In this paper, we suggest the extension of SIP functionalities to coordinate QoS mechanisms deployed in IP networks, and especially in DiffServ domain. Indeed, the interaction between small and big TCP sessions may have dramatic consequences on small TCP sessions. Hence, we use SIP to achieve QoS management on a session basis, in which the over all activity of the user during the session is considered. The suggested mechanisms deal with two issues: first, session scheduling based on session duration and/or volume, and second bandwidth allocation on a per-flow basis using equivalent bandwidth estimation techniques. The proposed mechanisms are implemented in the SIP proxy server as QoS management algorithms, and they are validated by simulations.

The Session Initiation Protocol (SIP) has been selected to manage sessions within the 3GPP IP-based Multimedia Subsystem (IMS). SIP sessions have to be highly available and reliable. This paper presents a performance evaluation of a novel state-sharing algorithm for providing reliable SIP services. The novelty is a result of sharing SIP (application-protocol) states as opposed to transport-protocol states. The most important performance parameters are evaluated in terms of the system variables such as server failure and repair time distributions, call arrival and call duration distributions, and link parameters, i.e., delay and bandwidth. The considered performance parameters are availability, reliability and dependability. Concerning the state transport mechanism, the results show that TFTP significantly outperforms FTP with respect to the overall performance.

NGN core flexibility, its subject to adoption and the rapid growth in the personal mobile devices led to rapid growth in mobile based services as mobile E-Learning. The personalized E-Learning service is proposed based on J2ME , it is integrated with open source IMS control frame work for user management, session establishment using SIP protocol and Multimedia learning content delivery like voice, video, whiteboard sharing using RTP and RTSP. The proposed E-Learning platform comprises on four layers user agent layer (3G mobile phone, J2ME based SIP learning and teaching agent application software), access layer (3G UMTS, Wi-Fi, GPRS), Data communication layer (SIP signaling, XML data exchange between mobile client and Application server) and E-Control and Management layer (IMS control functions, HSS, Learning content application servers). This personalized E-Learning provides opportunity for learners and teachers to learn, teach and control management everywhere.

SIP (The Session Initiation Protocol) is widely used as the signaling protocol for various services in the omnipresent environment. SIP is textual based protocol so the ability to process SIP messages quickly is critical for the performance of consumer electronic devices, such as SIP phone and the various Gateway. Our work enables homogeneous communication between heterogeneous distributed devices. Using SIP protocol communication bus to control widely varying entities, including ZigBEE, serial based (RS232) X10 and other based devices.

The IP Multimedia Subsystem (IMS) delivers multimedia services using IP integrated voice and packet services. IMS also provide users seamless connectivity. In this paper a location tracking system for IMS is developed track the location of a device. The existing testbed that supports Session Initiated Protocol (SIP) had been developed in the UTM MIMOS Coe, Universiti Teknologi Malaysia. However it is using 2D location tracking mechanism. The 2D location tracking mechanism only supports tracking the mobile unit position in the same level and provides handover or device switching. However if the user need to go to another level, user will have to disconnected the connection. This paper proposed a 3D indoor location tracking for mobile units allow roaming with SIP support. The location tracking will in a way help the user by providing seamless mobility at all services level. This paper present the initial results for the 3D location tracking tested at one level.

One of the most popular and widely applied protocols on multimedia networks is the Session Initiation Protocol (SIP) to create, modify, and terminate the sessions. SIP is the platform of Next Generation Networks (NGN). In this way, SIP should be able to respond to the needs of such a largely-used network. One of the major problems in SIP networks is overload. This challenge creates a sharp drop in quality of service for NGN users. In this regard, many studies have been conducted on the effectiveness of this protocol, especially under overload. A new analytical model is developed that prioritizes the SIP message processing. An analytical approach is proposed based on the Mean Value Analysis (MVA) algorithm in queue theory. Considering some appropriate assumptions customizing MVA as to implement this proposed model and to cope with the limitations of the MVA is highly essential. The output of the analytical model is compared with the standard SIP model obtained from the simulator and the results confirm that prioritizing original messages would enhance the SIP performance at different load conditions. Prioritization of original messages is advantageous, and outperforms the normal SIP. Nevertheless, prioritizing the repeated messages not only has no advantage, but also its performance is less than the normal SIP.

As wireless operators look for new revenue opportunities to justify their investment in third-generation equipment, multimedia services appear to offer the best possibility of satisfying their needs. The IP Multimedia Subsystem (IMS), driven by the 3rd Generation Partnership Project/3rd Generation Partnership Project 2 (3GPP/3GPP2) and based on Session Initiation Protocol (SIP), provides a basis for delivering these next-generation services. In this paper, we present a detailed model of the IMS services that are being considered for implementation in 3GPP. We also analyze architecture options for both existing and new operators and use a network model to investigate the economic viability of IMS deployment. We compare the pros and cons of implementing multimedia services with and without IMS. Finally, we present a case study-based on the network of a thirdgeneration (3G) wireless service provider-that describes the design of the network, considers the impact of IMS services on its capacity, and offers an economic analysis of the network that takes into account traffic forecasts, standards evolution, and product roadmaps.

Video oriented services are taking the lead in terms of revenue in the current Internet environment. In order to tap this revenue, Telcos are resorting to serious measures such as moving to Next Generation Network (NGN) where they can provide QoS for services that demand this feature, such as, IPTV. IPTV is more than just what its name implies and

Protocol testing for various real world applications can be analyzed using several methods, i.e. measurement, simulation and emulation. The first method from several method is live network testing. The second method is network simulation (simulation using Network Simulator) and the last method is emulation method (Emulab Testbed, DETER Testbed, and NIST Net). Emulation Method in network is the method which combines real elements of network application that has been deployed. For example an end host and protocol implementation, with synthetic elements, simulated or abstracted, such as network link, intermediate node, and background traffic. This paper shows the result of our experiment and testing of a Voice over Internet Protocol (VoIP) application using Session Initiation Protocol (SIP). VoIP testing is conducted using an open source network emulation software tool called NIST Net. VoIP whicg use SIP can be deployed in a real network. The network parameters used in our experiments are bandwidth and drop packets. The result shows the bandwidth and jitter characteristic during the experiment. It can be summarized that the quality of VoIP depends on bandwidth, jitter, and drop packet.

Significant research efforts for the convergence of Web and Telecommunication services have been recently spent by research and industry stakeholders. The IETF and W3C are cooperating in specifying how web browsers should evolve to natively support communication services. In this perspective, devising novel mechanisms for signaling message exchange and possible interworking between Web-and SIP-based systems is a hot topic of research. Indeed, discussions are still ongoing on how differences between REpresentational State Transfer (REST) and Session Initiation Protocol (SIP) models should be coped with. This issue is made more difficult by the lack of rigorous modeling of RESTful systems. In this paper we propose a rigorous approach for design and implementation of REST communication services (e.g., a call service) which leverages formal verification techniques, while allowing to meet a specific performance requirement (i.e., maximum call setup delay). First, we formalize the call resource behavior through a Finite State Machine representation by modeling and simulating service expected behavior

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The rapid replacement of PSTN with VOIP networks indicates the definitive phase-out of the PBX/PABX with smartphone-based VOIP technology that uses WLAN connectivity for local communication; however, security remains a key issue, regardless of the communication coverage area. Session initiation protocol (SIP) is one of the most widely adopted VOIP connection establishment protocols but requires added security. On the Internet, different security protocols, such as HTTPS (SSL/TLS), IPSec, and S/MIME, are used to protect SIP communication. These protocols require sophisticated infrastructure and some pose a significant overhead that may deteriorate SIP performance. In this article, we propose the following: i) avoid using Internet bandwidth and complex Internet protocols for local communication within an organization, but harness WLAN connectivity, ii) use multithreaded or multicore computer systems to handle concurrent calls instead of installing hardware-based SIP servers, and iii) run each thread in a separate core. Cryptography is a key tool for securely transmitting confidential data for long-and short-range communication, and the Diffie-Hellman (DH) protocol has consistently been a popular choice for secret key exchanges. Primarily, used for symmetric key sharing, it has been proven effective in generating public/private key pairs, sharing public keys securely over public channels, and subsequently deriving shared secret keys from private/public keys. This key exchange scheme was proposed to safeguard VOIP communication within WLANs, which rely on the SIP for messaging and multimedia communication. For ensuring an efficient implementation of SIP, the system was rigorously analyzed using the M/M/1 and M/M/c queuing models. We analyze the behavior of SIP servers with queuing models with and without end-to-end security and increase users' trust in SIP security by providing a transparent sense of end-to-end security as they create and manage their private and public keys instead of relying on the underlying SIP technology. This research implements instant messaging, voice

Session Initiation Protocol (SIP) can be defined as an application layer protocol that was developed in order to initiate, modify, and stop a multimedia session through internet network. In recent decades, SIP is coming more and more popular as it is adopted by many typical applications such as telephony over IP (ToIP). Due to its popularity, SIP has some security problems that we have to be resolve. In this paper, we concentrate our efforts on SIP malformed messages attack detection. Therefore, we propose our algorithm that can determine whether the received message is normal or abnormal. The simulation results confirm that our algorithm is efficient and can detect the attack in a minimum time.

MANET is one of the wireless networks in which interest is growing due to its flexibility, scalability and non-infrastructure status which is characterized by dynamically distributed clusters of nodes that have the ability to communicate locally and with the Internet. Voice over MANET-IP has become an attractive and popular application. This paper investigates the voice traffic application over the Ad Hoc on Demand Vector reactive routing protocol for different mobility models. The simulation evaluation using OPNET demonstrates the advantage of the Global System for Mobile Communication voice codec over the Internet using the UDP/IP protocol. The voice signaling and the Session Initiation IP (SIP) application layer protocols were tested and compared using different parameters. Different scenarios of mobility and node capacities were considered to test Quality of Service indicators including end-to-end delay, jitter, throughput, load, mean opinion score of voice quality, traffic sent and received, control signals, physical locations of calling nodes and the number of hops. The results showed that using the voice signaling produced better performance in respect of delays when compared to the SIP session. Voice has performed better when compared to the three mobility models with slightly less throughput and longer delays when considering the very fast mobility model. The node capacity showed serious impact on all performance metrics when MANET capacity exceeds 50 nodes in general. Keywords MANET Á AODV Á UDP Á Voice Á QOS Á Mobility Á SIP Please note that the LNCS Editorial assumes that all authors have used the western naming convention, with given names preceding surnames. This determines the structure of the names in the running heads and the author index.

MANET is one of the wireless networks in which interest is growing due to its flexibility, scalability and non-infrastructure status which is characterized by dynamically distributed clusters of nodes that have the ability to communicate locally and with the Internet. Voice over MANET-IP has become an attractive and popular application. This paper investigates the voice traffic application over the Ad Hoc on Demand Vector reactive routing protocol for different mobility models. The simulation evaluation using OPNET demonstrates the advantage of the Global System for Mobile Communication voice codec over the Internet using the UDP/IP protocol. The voice signaling and the Session Initiation IP (SIP) application layer protocols were tested and compared using different parameters. Different scenarios of mobility and node capacities were considered to test Quality of Service indicators including end-to-end delay, jitter, throughput, load, mean opinion score of voice quality, traffic sent and received, control signals, physical locations of calling nodes and the number of hops. The results showed that using the voice signaling produced better performance in respect of delays when compared to the SIP session. Voice has performed better when compared to the three mobility models with slightly less throughput and longer delays when considering the very fast mobility model. The node capacity showed serious impact on all performance metrics when MANET capacity exceeds 50 nodes in general.

MANET is one of the wireless networks in which interest is growing due to its flexibility, scalability and non-infrastructure status which is characterized by dynamically distributed clusters of nodes that have the ability to communicate locally and with the Internet. Voice over MANET-IP has become an attractive and popular application. This paper investigates the voice traffic application over the Ad Hoc on Demand Vector reactive routing protocol for different mobility models. The simulation evaluation using OPNET demonstrates the advantage of the Global System for Mobile Communication voice codec over the Internet using the UDP/IP protocol. The voice signaling and the Session Initiation IP (SIP) application layer protocols were tested and compared using different parameters. Different scenarios of mobility and node capacities were considered to test Quality of Service indicators including end-to-end delay, jitter, throughput, load, mean opinion score of voice quality, traffic sent and received, control signals, physical locations of calling nodes and the number of hops. The results showed that using the voice signaling produced better performance in respect of delays when compared to the SIP session. Voice has performed better when compared to the three mobility models with slightly less throughput and longer delays when considering the very fast mobility model. The node capacity showed serious impact on all performance metrics when MANET capacity exceeds 50 nodes in general.

MANET is one of the wireless networks in which interest is growing due to its flexibility, scalability and non-infrastructure status which is characterized by dynamically distributed clusters of nodes that have the ability to communicate locally and with the Internet. Voice over MANET-IP has become an attractive and popular application. This paper investigates the voice traffic application over the Ad Hoc on Demand Vector reactive routing protocol for different mobility models. The simulation evaluation using OPNET demonstrates the advantage of the Global System for Mobile Communication voice codec over the Internet using the UDP/IP protocol. The voice signaling and the Session Initiation IP (SIP) application layer protocols were tested and compared using different parameters. Different scenarios of mobility and node capacities were considered to test Quality of Service indicators including end-to-end delay, jitter, throughput, load, mean opinion score of voice quality, traffic sent and received, control signals, physical locations of calling nodes and the number of hops. The results showed that using the voice signaling produced better performance in respect of delays when compared to the SIP session. Voice has performed better when compared to the three mobility models with slightly less throughput and longer delays when considering the very fast mobility model. The node capacity showed serious impact on all performance metrics when MANET capacity exceeds 50 nodes in general.

MANET is one of the wireless networks in which interest is growing due to its flexibility, scalability and non-infrastructure status which is characterized by dynamically distributed clusters of nodes that have the ability to communicate locally and with the Internet. Voice over MANET-IP has become an attractive and popular application. This paper investigates the voice traffic application over the Ad Hoc on Demand Vector reactive routing protocol for different mobility models. The simulation evaluation using OPNET demonstrates the advantage of the Global System for Mobile Communication voice codec over the Internet using the UDP/IP protocol. The voice signaling and the Session Initiation IP (SIP) application layer protocols were tested and compared using different parameters. Different scenarios of mobility and node capacities were considered to test Quality of Service indicators including end-to-end delay, jitter, throughput, load, mean opinion score of voice quality, traffic sent and received, control signals, physical locations of calling nodes and the number of hops. The results showed that using the voice signaling produced better performance in respect of delays when compared to the SIP session. Voice has performed better when compared to the three mobility models with slightly less throughput and longer delays when considering the very fast mobility model. The node capacity showed serious impact on all performance metrics when MANET capacity exceeds 50 nodes in general.

MANET is one of the wireless networks in which interest is growing due to its flexibility, scalability and non-infrastructure status which is characterized by dynamically distributed clusters of nodes that have the ability to communicate locally and with the Internet. Voice over MANET-IP has become an attractive and popular application. This paper investigates the voice traffic application over the Ad Hoc on Demand Vector reactive routing protocol for different mobility models. The simulation evaluation using OPNET demonstrates the advantage of the Global System for Mobile Communication voice codec over the Internet using the UDP/IP protocol. The voice signaling and the Session Initiation IP (SIP) application layer protocols were tested and compared using different parameters. Different scenarios of mobility and node capacities were considered to test Quality of Service indicators including end-to-end delay, jitter, throughput, load, mean opinion score of voice quality, traffic sent and received, control signals, physical locations of calling nodes and the number of hops. The results showed that using the voice signaling produced better performance in respect of delays when compared to the SIP session. Voice has performed better when compared to the three mobility models with slightly less throughput and longer delays when considering the very fast mobility model. The node capacity showed serious impact on all performance metrics when MANET capacity exceeds 50 nodes in general.

The quality of human resources is important factor for the passage of an institution. STIKes Perdhaki Charitas is a health educational institution that is always seeking quality education. Assessment of the lecturers and employees is important to measure the performance result. The results of the assessment carried out at this time, currently pay less attention to the weight of each assessment criteria so that assessment becomes less accurate. Then designed a decision support system for evaluation the performance of employees and lecturers using the Simple Additive Weighting method. The method used to design the DSS using Waterfall process model, and the data collection is done by direct observation and interviews. DSS uses Xampp package (Apache, MySql and phpMyAdmin) and PHP programming. The results of this DSS based on month period and year period. The conclusion result from STIKes and DSS almost the same. But the different is the ranking. The same assessment for employees A and B...

With the widespread adoption of SIP-based VoIP, understanding the characteristics of SIP traffic behavior is critical to problem diagnosis and security protection of IP Telephony. In this paper, we propose a general methodology for profiling SIP-based VoIP traffic behavior at multiple levels: SIP server host, server entity and individual user levels. Using SIP traffic traces captured in a production VoIP service, we illustrate the characteristics of SIP-based VoIP traffic behavior in an operational network and demonstrate the effectiveness of our general profiling methodology. In particular, we show how our profiling methodology can help identify performance anomalies through a case study.

The rapid replacement of PSTN with VOIP networks indicates the definitive phase-out of the PBX/PABX with smartphone-based VOIP technology that uses WLAN connectivity for local communication; however, security remains a key issue, regardless of the communication coverage area. Session initiation protocol (SIP) is one of the most widely adopted VOIP connection establishment protocols but requires added security. On the Internet, different security protocols, such as HTTPS (SSL/TLS), IPSec, and S/MIME, are used to protect SIP communication. These protocols require sophisticated infrastructure and some pose a significant overhead that may deteriorate SIP performance. In this article, we propose the following: i) avoid using Internet bandwidth and complex Internet protocols for local communication within an organization, but harness WLAN connectivity, ii) use multithreaded or multicore computer systems to handle concurrent calls instead of installing hardware-based SIP servers, and iii) run each thread in a separate core. Cryptography is a key tool for securely transmitting confidential data for long-and short-range communication, and the Diffie-Hellman (DH) protocol has consistently been a popular choice for secret key exchanges. Primarily, used for symmetric key sharing, it has been proven effective in generating public/private key pairs, sharing public keys securely over public channels, and subsequently deriving shared secret keys from private/public keys. This key exchange scheme was proposed to safeguard VOIP communication within WLANs, which rely on the SIP for messaging and multimedia communication. For ensuring an efficient implementation of SIP, the system was rigorously analyzed using the M/M/1 and M/M/c queuing models. We analyze the behavior of SIP servers with queuing models with and without end-to-end security and increase users' trust in SIP security by providing a transparent sense of end-to-end security as they create and manage their private and public keys instead of relying on the underlying SIP technology. This research implements instant messaging, voice

Voice over IP communication will dominate the world. However, given the growing demand for voice and data communication to make any and all communication reliable and secure, several attacks occur frequently in communication networks, so this work is based on verifying security, analyzing risks, vulnerabilities, such as verifying the attacks and proposing a security measure for voice over IP communication on embedded

A 4G systems can provide a comprehensive IP based solution where voice, data, video streamed multimedia can be provided to users on an “Anytime, Anywhere” The aim of the work is to estimate quality of metric values with different IEEE 802.11 standards (IEEE 802.11. b. /a/g), utilized for SIP (Session initialized protocol) video call setup procedure over wireless link scenario directed to mobile devices. We used two video call modes of operation over heterogeneous WiFi standards. The proposed model application layers in SIP protocol based on multicast mobile agents used in two different modes of video call established. Initiate services through CBR used for Session control overhead packets. Our analytical model provides that the video call set-up performance, jitter and delay in peer to peer networks. Moreover, the call setup performance can be improved significantly using the robust in application link layer such as SIP with a comparison of WIFI heterogeneous network standards propo...

Intellectual mobile terminals (or users) of next generation wireless networks are expected to initiate/Establish voice over IP (VoIP) calls using session set-up protocols like H.323 or SIP (Session Initialized Protocols). To provide quality metrics of video conferences, telemedicine and other voice over broadband telephony (VoBB) applications. In this work, we analyse the performance of the H.323 call setup procedure over the wireless link. We used to call modes of operation over heterogeneous networks. The proposed model application layers in the RTP Control Protocol and Real-Time Transport Control Protocol (RTCP) protocols used in two different modes of call established. Initiate services through VoIP used for H.323 control packets. Our analytical model provides that the VoIP call set-up performance, jitter and delay in peer to peer networks. Moreover, the call setup performance can be improved significantly using the robust in application link layer such as RTP/RCTP with a comparison of heterogeneous network proposed in our paper. The analytical results were validated by our experimental measurements

Presence and Instant Messaging have recently emerged as a new medium of communications over the Internet. Presence is a means for finding, retrieving, and subscribing to changes in the presence information (e.g. "online" or "offline") of other users. Instant messaging is a means for sending small, simple messages that are delivered immediately to online users. Applications of presence and instant messaging currently use independent, non-standard and non-interoperable protocols developed by various vendors. The goal of the Instant Messaging and Presence Protocol (IMPP) Working Group is to define a standard protocol so that independently developed applications of instant messaging and/or presence can interoperate across the Internet. This document defines a minimal set of requirements that IMPP must meet.

In order to improve the quality of the secure communication we need to require secure system hence the review paper has to be defining the requirement of the secure system as well as the cost effective system. Our Embedded system is to design and develop a low cost feature which is based on embedded platform for VOIP media using ARM architecture. VOICE OVER INTERNET PROTOCAL (VOIP) terminal, dedicatedly used for the encryption/de-encryption of the VoIP signal and the RTP (Real-time Control Protocol) voice packet. The encryption flow of the packet is described when the VoIP protocol is SIP (Session Initiation Protocol) and the encryption algorithm is RC4. Keyword: mini2240 development board , Linux OS, RTP,SIP,Qt creator

The frantic nature of technological advances in the area of multimedia communications, compounded with the convergence of telecommunication and computer networks has led to the emergence of new class of applications sometime called ubiquitous computing or ambient intelligence. In this paper we present a project which converges with this new generation of intelligent applications. We have named this project UCIAN for "University Communication at Intelligent Ad Hoc Networks". This project is dedicated to university teachers; it consists of creating an intelligent environment at the university that allows teachers to explore SIP (Session Initiation Protocol) services, in particular Instant Messaging (IM) service and Presence service.

Session Initiation Protocol (SIP) can be defined as an application layer protocol that was developed in order to initiate, modify, and stop a multimedia session through internet network. In recent decades, SIP is coming more and more popular as it is adopted by many typical applications such as telephony over IP (ToIP). Due to its popularity, SIP has some security problems that we have to be resolve. In this paper, we concentrate our efforts on SIP malformed messages attack detection. Therefore, we propose our algorithm that can determine whether the received message is normal or abnormal. The simulation results confirm that our algorithm is efficient and can detect the attack in a minimum time.