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This document defines the Session Description Protocol, SDP. SDP is intended for describing multimedia sessions for the purposes of session announcement, session invitation, and other forms of multimedia session initiation.

Although the Session Initiation Protocol (SIP) includes a suite of security services that has been expanded by numerous specifications over the years, there is no single place that explains how to use SIP to establish confidential media sessions. Additionally, existing mechanisms have some feature gaps that need to be identified and resolved in order for them to address the pervasive monitoring threat model. This specification describes best practices for negotiating confidential media with SIP, including both comprehensive protection solutions which bind the media to SIP-layer identities as well as opportunistic security solutions.

The session initiation protocol (SIP) is an IP-based telephony authentication mechanism for multimedia big data communications over the Internet. It is used to set up, and control voice and video calls, as well as for instant messaging. One of the concerns of this kind of open-text-based protocol is the security for user authentication. The HTTP digest-based challenge-response authentication process is used in the original SIP. However, this kind of authentication procedure is insecure and a pre-existing user configuration on the remote server is required. According to the literature, several authentication mechanisms for SIP are already devised, but none of these SIPs are robust against existing security attacks. Therefore, we design a three-factor SIP (TF-SIP) for multimedia big data communications, which is robust and flexible against existing known security issues. We show that our TF-SIP is provably secure in the random oracle model. We formally verify the mutual authentication and the freshness of the agreed session key between the user and the remote server using the BAN logic analysis. We found that the communication and computation costs are low, but the storage cost is slightly higher for our TF-SIP in comparison with other SIPs.

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.

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.

Modelo Sip Seguro para una Comunicación extremo a extremo sobre IPV6 Sip Security Model for End to End Communication on IPv6 Ross M. Benites, José L Quiroz, Raúl Villafani INICTEL-UNI, Lima 41 DOI: https://doi.org/10.33017/RevECIPeru2011.0024/ RESUMEN Las implementaciones VoIP hoy en día se han incrementado considerablemente, sin embargo no entodos los escenarios se tiene en cuenta los mecanismos de seguridad adecuados. Este último punto es muy importante a considerar el día de hoy , sobre todo por el agotamiento de las direcciones IPV4 y el despliegue hacia IPV6 de muchos de los servicios, donde aparecerán nuevas amenazas a la seguridad que trataran de opacar el gran auge de la tecnología VoIP. Si bien IPv6 fue desarrollado para solucionar muchas vulnerabilidades en seguridad que actualmente se ven presentes en IPv4, el hecho es que no logra alcanzar aún estas metas según pruebas realizadas. El protocolo SIP, el actor principal de la tecnología VoIP , requiere de la implementación de mecanismos de seguridad . Los escenarios actuales requieren terminales de usuario de alto rendimiento y soporte para adaptarse a mecanismos de seguridad heterogéneos o asumir relaciones de confianza. Sin embargo debemos tener en cuenta que hay varias combinaciones de soluciones de seguridad que son proporcionados por usuarios finales y los servidores. En este trabajo se expone un modelo de seguridad aplicado a un escenario experimental VoIP sobre el Internet de Próxima Generación (IPv6), que utiliza la seguridad salto a salto y extremo a extremo. El escenario propuesto se encuentra sobre una red local con direcciones IPv6. Utiliza dos servidores Asterisk implementados bajo las mismas características que cumplen la función de SIP Proxy, y se encuentran conectados mediante un enlace troncal SIP – TLS. Se utilizan además dos terminales de usuario (teléfonos IP) provenientes de una marca comercial conocida, registrados cada uno mediante el protocolo SIP-TLS a cada servidor Asterisk. En trabajos anteriores, se han realizado varios estudios sobre el rendimiento del uso de VoIP sobre IPv4 e IPv6 comparando los resultados [1], evaluación de mecanismos de seguridad para mantener la autenticación de usuario, confidencialidad e integridad de la señalización y media de los mensajes VoIP sobre las redes IPv4 [2] y [3]. Este trabajo se esboza en un marco de seguridad, donde se presenta un escenario basado en una red VoIP en IPV6 utilizando TLS y SRTP. TLS es utilizado para la seguridad en el establecimiento de la sesión con mecanismos de autenticación salto a salto y SRTP (Secure Real Time Protocol) para la seguridad del establecimiento del stream de media. Nos enfocaremos en analizar y evaluar la seguridad de los mensajes en este escenario sobre el protocolo de transporte seguro (TLS). Descriptores: sip, tls ,ipv6, srtp. ABSTRACT VoIP deployments today have increased considerably, but not all the scenarios consider appropriate security mechanisms. This last point is very important to consider today, especially the depletion of IPv4 addresses and the deployment of many services IPv6, where will new security threats to try to overshadow the great technology boom VoIP. Although IPv6 was developed to solve many security vulnerabilities are currently present in IPv4, the fact is that still fails to achieve these goals by testing. The SIP protocol, the main actor of VoIP technology requires the implementation of security mechanisms. The current scenarios require high-end user performance and support to adapt to heterogeneous security mechanisms or assume trust relationships. But keep in mind that there are various combinations of security solutions that are provided by end users and servers. This paper presents a security model applied to an experimental scenario VoIP over Next Generation Internet (IPv6), which uses hop by hop security and end to end. The proposed scenario is on a local network with IPv6 addresses. Use two Asterisk servers implemented under the same characteristics that act as SIP Proxy, and are connected via SIP trunk - TLS. They also use two user terminals (IP phones) from a known trade mark registered by each SIP-TLS protocol for each server Asterisk. In previous work, there have been several studies on the performance of VoIP using IPv6 and IPv4 and comparing the results [1], evaluation of security mechanisms to support user authentication, confidentiality and integrity of the signaling and media messages VoIP over IPv4 networks [2] and [3]. This paper outlines a framework of security, which presents a scenario based on a VoIP network in IPv6 using TLS and SRTP. TLS is used for the security session establishment authentication mechanisms hop by hop and SRTP (Secure Real Time Protocol) for the safety of the establishment of media stream. We will focus on analyzing and evaluating the security of the messages in this scenario the secure transport protocol (TLS) . Keywords: sip, tls ,ipv6, srtp.

The session initiation protocol is one the most popular protocols that is used in Internet protocol multimedia subsystems and adopted by a wide range of networking vendors. This research suggests a secure distributed session initiation protocol-based architectural model that can be deployed in service provider data centers to maintain the service availability, scalability, and security. This research proposes a secure session initiation protocol model. It called the redundant session initiation protocol model, and used to stop denial of service attacks. The proposed techniques should provide robust and secure implementations against today's vulnerabilities. The primary consideration of the predefined network is to acquire reliable mechanisms that handle the private information regarding legitimate users. The authors tried to suggest a secure distributed SIP registration mechanism that can be deployed in service provider data centers to maintain the service availability, security and reduce SIP attacks like IP spoofing.

Voice over Internet Protocol (VoIP) is spreading across the market rapidly due to its characteristics such as low cost, flexibility implementation, versatility of new applications, etc. However, the voice packets transmitted over the Internet are not protected in most VoIP environments, and then the user's information could be easily compromised by various malicious attacks. So an energy-efficient authenticated key agreement protocol for Session Initial Protocol (SIP) should be provided to ensure the confidentiality and integrity of data communications over VoIP networks. To simplify the authentication process, several protocols adopt a verification table to achieve mutual authentication, but the protocols require the SIP server to maintain a large verification table which not only increases energy consumption but also leads to some security issues. Although several attempts have been made to address the intractable problems, designing an energy-efficient authenticated key agreement protocol for SIP-based green VoIP networks is still a challenging task. In this study, we propose an efficient authentication protocol for SIP by using smartcards based on elliptic curve cryptography. With the proposed protocol, the SIP server needs not to store a password or verification table in its database, and so no energy is required for the maintenance of the verification table. Security analysis demonstrates that the proposed protocol can resist various attacks and provides efficient password updating. Furthermore, the experimental results show that the proposed protocol increases efficiency in comparison with other related protocols. We proposed an energy efficient authentication protocol for VoIP networks.No energy is required for the maintenance of verification tables in our protocol.Our protocol can resist various attacks and provides efficient password updating.The experimental results show that the proposed protocol increases efficiency.

Organisations have latched on to the need to secure SIP trunking solutions by implementing a Session Border Controller (SBC). The problem is that the vast majority of SBCs are considered not just one-off investments but also one-off deployments. Yet from denial of service attacks to toll fraud, SIP trunking is not only inherently vulnerable, that vulnerability continues to change and escalate. Few companies would fail to update anti-virus software, so why assume the SBC can protect against changing threats without similar routine updates? The fact is that in their current guise, most SBCs actually leave organisations with a false sense of security.

The smartphone industry has gained a large user base since the digital revolution. Due to this, even the Telecommunication operators have had a surge in customers. This has led to call tariffs being decided by the operators and frequently users end up paying for more than what they have used. This research paper explores the option of local call routing using femtocells instead of routing the call through a switching center that belongs to a licensed operator. It has been statistically verified that using VoIP based systems based on the Session Initiation Protocol is very efficient in cutting down costs. Gartner predicts that SIP trunking services can slash enterprise telecommunications expenses by up to 50 percent. It is estimated that VoIP systems can reduce initial startup costs to new businesses by up to 90 percent. The Session Initiation Protocol is the ideal candidate for efficient implementation of VoIP-based systems. The general idea being portrayed here is that the traffic generated in a network does not leave the network and terminates in a host within the same network. This ensures that the overhead to reach an operator is minimized. A call manager, specific to VoIP packets, may be developed to address hosts and efficiently route video and audio traffic in between them. Textual data can also be exchanged. This proves very useful during disaster relief as it is possible to set up such systems with relative ease.

In recent years, Voice over Internet Protocol (VoIP) has gained more and more popularity as an application of the Internet technology. For various IP applications including VoIP, the topic of Session Initiation Protocol (SIP) has attracted major concern from researchers. SIP is an advanced signaling protocol operating on Internet Telephony. SIP uses digest authentication protocols such as Simple Mail Transport Protocol (SMTP) and Hyper Text Transport Protocol (HTTP). When a user seeks SIP services, authentication plays an important role in providing secure access to the server only to the authorized access seekers. Being an insecure-channel-based protocol, a SIP authentication protocol is susceptible to adversarial threats. Therefore, security is a big concern in SIP authentication mechanisms. This paper reveals the security vulnerabilities of two recently proposed SIP authentication schemes for VoIP, Irshad et al.'s scheme [Multimed. Tools. Appl. doi:10.1007/s11042-013-1807-z] and Arshad and Nikooghadam's scheme [Multimed. Tools. Appl. DOI 10.1007/s11042-014-2282-x], the later scheme is based on the former scheme. Irshad et al.'s scheme suffers from password guessing, user impersonation and server spoofing attacks. Arshad and Nikooghadam's scheme can be threatened with server spoofing and stolen verifier attack. None of these two schemes achieve mutual authentication. It also fails to follow the single round-trip authentication design of Irshad et al.'s scheme. To overcome these weaknesses, we propose a provable secure single round-trip SIP authentication scheme for VoIP using smart card. We formally prove the security of the scheme in random oracle and demonstrate through discussion its resistance to various attacks. The comparative analysis shows that the proposed SIP authentication scheme offers superior performance with a little extra computational cost.

This work gives an over view of IP telephony and explain the existing difficulties in implementing VoIP services & migrate the TDM base telephony to IP telephony. When the architecture is deployed in IP based network. The causes of these problems are explained with three approaches used by carrier to solve interoperability. They are TDM base telephony to IP telephony by using soft switch and voice gateway, using different audio codec’s for better voice quality, band width used in IP telephony. General Terms Routing, Rate, Currentcalls, Server

This paper proposed an access control for computer-based integrated multimedia running on shepherd and SIP(Session Initiation Protocol). But, conventional framework for access control SIP environment has not yet fully progressed a shepherd and an access control for computer-based integrated multimedia running on SIP(Session Initiation Protocol). Session management include function of session creation, session end, late comer process, and access control. Therefore, this paper described access control based on a shepherd and SIP environment to maintain good session condition.

In this paper, we have studied and described the security of SIP Trunk on IMS network (IP Multimedia Subsystem) via SBC (Session Border Controller), introduced a model by installing SBC to connect between IMS Network and IP-PBX in concept of SIP Trunk and present new method to increase performance of SIP Trunk by specified concurrent call, setting the caller discrimination to filter subscriber number that are not specified in the system cannot call via SIP Trunk (To prevent Hijacked subscriber illegally) and limit minimum and maximum digit of subscriber in the system to protect the security from inside and outside attacker that can effected the performance of network. Furthermore, we have set the experiment with the live network by installing the system follow by system model consists of PBX server connected to IMS network (IP Multimedia Subsystem) via SBC (SBC is installed between IMS network and IP PBX, using wireshark to monitor and analyze signaling flow of each equipment. Finally, the result from experiment are show the system model can secure and protect system from inside and outside attacker including prevent Hijacked subscriber illegally and can be used to increase the performance of SIP Trunk security in live network efficiently.

Session Initiation Protocol (SIP), which is an IP based telephony protocol, is used mainly for the purpose of starting, sustaining and ending sessions related to multimedia communications on the Internet. The SIP protocol, which works on the top of TCP or UDP, is basically an open text-based protocol. Hence, to ensure security is of utmost importance. The original SIP used HTTP-digest based challenge-response authentication process. However, HTTP digest-based authentication is insecure and pre-existing user configuration on the remote server is needed. Moreover, it provides only one-way message authentication and replay protection, but not the support message integrity and confidentiality. Although, quite a few three factor SIP protocols using password, smartcard and biometric are existing in the literature, however, none of them are robust against known attacks. In this paper, a robust and cost-efficient VoIP based three-factor SIP is proposed based on the computational Diffie-Hellman problem.

The immense growth in wireless network applications has encouraged the researchers to enhance and propose new approaches that facilitate the deployment of the widely used services, protocols, and applications of the wired networking area in the wireless networks. Among which, the Session Initiation Protocol (SIP) is a signaling protocol that allows the establishment of multimedia sessions and calls between different parties. SIP functionality totally depends on a centralized infrastructure, and complexity arises when deploying such a protocol in a special type of wireless networks, namely Mobile Ad-hoc Networks (MANET), due to the absence of a centralized infrastructure, and the dynamic topology of the later. This paper highlights the the challenges arising when deploying SIP in MANETs and presents an overview of different proposed approaches to overcome these challenges.

IMS(IP Multi-Media Subsystem) is in the limelight as the Integrated wire and wireless Systems because of a sudden increase of smart mobile devices and growth of multimedia additional services such as IPTV. The structure of IMS is designed as a session control layer to provide various multimedia summative service using SIP based on IP communication network in order to carry out set-up, change and release by NGN of course, the existing voice services. But now It is broadly substituting in the IPTV, wire phone company and it is substituted in internet platform base on the soft-switch in currently. Especially, in currently, 4G LTE in a mobile communication company is rapidly growing in market. Therefore, in this study, we had designed and developed to the main prosser that can admit to 1000 user over and SIP gateway which can link the IMS Core that can link SIP Device which adopt the standard protocol on the SIP and to provide variable multimedia services. Key Words : IMS Core, 3GPP, PSTN, CSCF, SIP

Optimizing the transmission efficiency of data is becoming an indispensable job in voice radio communication system. IP Multimedia Subsystem (IMS) standardized Session Initiation Protocol (SIP), which categorize as a signaling protocol for management and maintenance of new multimedia IP-based services. Session setup delay is one of important index of it. In order to reduce SIP signaling delay, to begin with, we analyze and study the factors affecting the session setup delay from different points, and solve the problems in good ways. In the end of the article, we contrast five compression algorithms of SIP signaling.

Internet telephony is an Internet service which supports voice telephone using VoIP technology on the IP-based Internet. It has some advantages in that voice telephone services can be accompanied with multimedia services such as video communication and messaging services. In this paper we suggested an Asterisk-based Internet telephony system which can be easily scalable. Most current systems use text files to manage their configuration: SIP users, dialplans, IVR service and etc. But we designed the management system which introduces database tables for efficiency and scalability. It also supports web-based functions developed by using Asterisk, Apache, MySQL, jQuery, PHP and open source softwares. Key Words : Asterisk, Internet Telephony, IVR, Open Source Software, SIP, VoIP * 본 논문은 안식년 기간 중 연구되었음Received 27 May 2014, Revised 7 July 2014Accepted 20 August 2014Corresponding Author: Eun-Yong Ha(Anyang University) Email: eyha@anyang.ac.krⒸ The Society of Digital Policy & Management. All rights reserved. This is an open-access article distributed under the terms of the Creative Commons Attribution Non-Commercial License (http://creativecommons.org/licenses/by-nc/3.0), which permits unrestricted non-commercial use, distribution, and reproduction in any medium, provided the original work is ISSN: 1738-1916 properly cited.

Considering the widespread adoption of smartphones in mobile communications and the well-established resource sharing use in the networking community, we present a novel mechanism to achieve anonymity in the Global System for Mobile Communications GSM. We propose a Voice over Internet Protocol infrastructure using the Session Initiation Protocol SIP where a smartphone registers on a SIP registrar and can start GSM conversation through another smartphone acting as a GSM gateway, by using a SIP intermediate without an extra cost. The testbed that we developed for empirical evaluation revealed no significant quality of service degradation. Copyright © 2014 John Wiley & Sons, Ltd.

Through analysis of mechanism and characteristics of Session initiation protocol (short for SIP), it is found that SIP possesses prominent flexibility and expansibility, which meet the requirement of Internet infrastructure and media flow transmission. To some extent, tightcoupling model based on SIP reflects the fundamentals of distance education and session initiation. Furthermore, while key control and media mixer are functionally separated. A Web server is added to improve mutual communication. The model SIP-based distance education system with three centralized servers are constructed and the corresponding topology structure is proposed. The mechanism and education function of the model are embodied and realized through several aspects, such as distance class creation, participation and termination, state information notification and education resource control.

This article interprets the current development status of trunking mobile communication system at home and abroad, makes a detailed analysis of the two existing architectures of trunking technology, introduces the demand of digital trunking broadband in China on the basis of the analysis on current situation and development trend of trunking mobile communication system. It also presents two types of domestic technical scheme in combination with the features of broadband trunking mobile communication systemr abstract.

Mobile traffic is increasing a masse because of the propagation of the Internet and the development of wireless mobile technology. Accordingly, the Network Local Mobility Management (NETLMM) working group [1] of the Internet Engineering Task Force (IETF) has standardized Proxy Mobile IPv6 (PMIPv6) [2] as a protocol for accomplishing the transmissibility of mobile terminals. PMIPv6 is a network-led IP-based mobility management protocol, which can control terminal mobility without depending on the type of access system or the capability of the terminal. By combining PMIPv6 and the mobility of Session Initiation Protocol (SIP), we can establish terminal mobility and session mobility through a more effective route. The mobility function can be improved and the overlap of function reduced as compared to that in the case of independent operation. PMIPv6 is appropriate for a non-real-time service using TCP, and SIP is appropriate for a real-time service using RTP/UDP. Thus, in the case of a terminal using both services, an effective mobility management is possible only by using PMIPv6 together with SIP. In order to manage mobility in this manner, researches on PMIPv6-SIP are in progress. In line with this trend, this paper suggests a new PMIPv6-SIP architecture where when a mobile terminal conducts a handover, a network-led handover while maintaining the session without the addition of a special function or middleware is possible along with effective performance evaluation through mathematical modeling by comparing the delay and the packet loss that occur during the handover to the Pure-SIP.

Voice over Internet Protocol (VoIP) is a family of technologies for the transmission of voice over Internet. Voice is converted into digital signals and transmitted as data packets. The Session Initiation Protocol (SIP) is an IETF protocol for VoIP and other multimedia. SIP is an application layer protocol for creating, modifying and terminating sessions in VoIP communications. Since SIP is a more flexible and simple protocol, it is quite easy to add features to it. Distributed Denial of Service Attack (DDoS) floods the server with numerous requests from various hosts. Hence, the legitimate clients will not be able to get their intended services. A major concern in VoIP and almost in all network domains is availability rather than data consistency. Most of the surviving techniques could prevent VoIP network only after collision. This paper proposes a Recurrence Quantification based approach to detect and prevent VoIP from a DDoS attack. This model detects the attack at an earlier stage and also helps to prevent from further attacks. In addition, this techniques enables the efficient utilization of resources. QUALNET has been used to simulate the operation of the proposed technology.

Recently, Voice over Internet Protocol (VoIP) has been one of the more popular applications in Internet technology. For VoIP and other IP applications, issues surrounding Session Initiation Protocol (SIP) have received significant attention. SIP is a widely used signaling protocol and is capable of operating on Internet Telephony, typically using Hyper Text Transport Protocol (HTTP) digest authentication protocol. Authentication is becoming increasingly crucial because it accesses the server when a user asks to use SIP services. In this paper, we concentrate on the security flaws in the current SIP authentication procedure. We propose a secure ECC-based authentication mechanism to conquer many forms of attacks in previous schemes. By a sophisticated analysis of the security of the ECC-based protocol, we show that it is suitable for applications with higher security requirements.

Soft Phone is a VoIP soft phone that uses the Session Initiation Protocol. It is a powerful and unique SIP software telephone that lets users make phone and video calls using single software application using any Voice over IP (VoIP) telephony provider. As a fully featured SIP client, Soft Phone is designed from the ground up to work with current and future IP-based. Soft phone can be used in the place where we need establish the voice and video phone calls in the IP network. This can be installed at any personal computer. It uses the SIP protocol to establish the IP session and using RTP, RTCP and RTSP for media transmission. Using Soft phone they make call to any Soft Phone and Physical hardware SIP phone which can understand the SIP protocol.In computing, a soft phone is a software program for making telephone calls over the Internet using a general purpose computer, rather than using dedicated hardware. A soft phone is usually used with a headset connected to the sound card of the PC. To communicate, both end-points must have the same communication protocol and at least one common audio code. Many service providers use the Session Initiation Protocol (SIP) standardized by the Internet Engineering Task Force (IETF)

VoIP can make a real-time voice communication by using IP network. SIP protocol is an application-layer session control protocol which can implement VoIP system. Based on the research of VoIP related technology and SIP protocol, this paper proposes a design scheme of VoIP intelligent voice communication system which is based on C/S architecture and accords with the SIP protocol specifications. The design scheme is also implemented on the corresponding development platform. Besides, this paper also test and verify the conversation process of communication terminal and its functions. The system has the advantages of small investment, low cost, convenience and practicality, high reliability, good security, etc. It is of great application and popularization value.

The Session Initiation Protocol (SIP) has revolutionized the way of controlling Voice over Internet Protocol (VoIP) based communication sessions over an open channel. The SIP protocol is insecure for being an open text-based protocol inherently. Different solutions have been presented in the last decade to secure the protocol. Recently, Zhang et al. authentication protocol has been proposed with a sound feature that authenticates the users without any password-verifier database using smart card. However, the scheme has a few limitations and can be made more secure and optimized regarding cost of exchanged messages, with a few modifications. Our proposed key-agreement protocol makes a use of two server secrets for robustness and is also capable of authenticating the involved parties in a single round-trip of exchanged messages. The server can now authenticate the user on the request message received, rather than the response received upon sending the challenge message, saving another round-trip of exchanged messages and hence escapes a possible denial of service attack.

The rapidly evolving communication system enables applications for telephone communication to be carried over the data network known as VoIP (voice over internet protocol). SIP (session initiation protocol) as the signaling protocol is text-based VoIP which can be implemented easily in comparison with other signalingprotocols. The purpose of this paper is designing and implementing VoIP billing up to the company to provide additional facilities for enterprise customers. The methods start with data collection, analysis, design, development, and implementation. The result achieved is a system of VoIP with SIP and Asterisk software which has functions of PBX to provide additional facilities such as VoIP which is a plus for the company and customers. After implemented, the VoIP system and billing features are found work well.

The decision to move to SIP trunking appears straightforward: reduced costs, greater scalability, improved disaster recovery options and access to the productivity benefits enabled by Unified Communications (UC). But, as ever in the world of technology, the truth about SIP trunking is a little more complex than it may appear at first glance. How much up-front investment is required to migrate from the existing ISDN trunk? How big is the hardware investment? How scalable is the solution? And, critically, is it secure?But is it secure? This issue is too often overlooked by companies wanting to move to SIP. The result is a ticking time bomb for businesses, with threats ranging from denial of service to toll fraud (also known as call jacking); and the liability lies squarely with the business, not the provider, as Paul German of VoipSec explains.

To solve the system overloading problem of sip protocol based on centralized monitoring and management system, we propose a new protocol system based on cross-regional stratification sip monitoring integrated management platform, it uses the automatic control technology, networking and communications technology, video / audio compression and transmission technology, sensors and integrated control technology and software engineering techniques. It is an integrated platform of biometrics, intelligent video analysis, anti-theft alarm techniques, 3G video management, valuables management, multifunction video and voice network, multimedia video services, and that it can be used with existing IP network, LAN, internet, 3G network, the telephone network. It is fully compatible and can be used with existing databases, monitoring systems, television networks for docking to achieve a variety of monitoring elements of centralized management, control and interaction. The innovative applications of CSS regional stratification mechanisms, ActiveX terminal buffering mechanism and VM resource allocation mechanism can provide users real-time information. And according to a variety of real-time information, it aggregated emergency solutions of the overall regional security or certain high-risk target key places of security.

In the ‘Next Generation Networks’, ‘Session Initiation Protocol’ (SIP) is widely used to control multimedia (e.g. voice and video) sessions and ‘Internet Protocol version 6’ (IPv6) is adopted to provide enough addressing space. However, in the current stage of IPv6 deployment, the newly IPv6-enabled device [i.e. typically internet protocol version 4 (IPv4)/IPv6 dual-stack device] may connect to an existing IPv4 device. In the traditional ‘server-based solutions’, the SIP server is modified to perform the IPv4–IPv6 translations to the SIP messages and the real-time transport protocol (RTP) packets. However, the IPv4–IPv6 translations increase the call setup latency, the RTP transmission delay and the packet loss possibility. To reduce the drawbacks, this study proposes a ‘client-based solution’, where the client (i.e. the end device) instead of the SIP server performs the IPv4–IPv6 translation. The authors utilise the message flows to elaborate three server based and the proposed client-based solutions. In addition, the authors implement all solutions and deploy them in an IPv4–IPv6 interworking testbed. By using the testbed, the authors analyse these solutions in terms of the ‘UA modification, the SIP Server modification, the call setup complexity and the RTP translation’.

Legacy networks, both fixed and mobile, which were originally designed for voice communications, are progressively migrating to new infrastructures that promise to revolutionize the services offered. In this paper, we will cover this new generation of personal communication services, with an emphasis on the family of Internet protocol (IP)-based multimedia subsystem (IMS)-aided infrastructure that relies on the session initiation protocol (SIP). As a benefit, the end users will enjoy a new generation of personal communications services that are accessible anywhere and anytime. These services are directly related to the end users rather than to their diverse devices. It is anticipated that the new deployments of next-generation networks (all-IP based) will accelerate the adoption of the IMS technology.

SIP (Session Initiation Protocol) has been chosen as the core signaling protocol of the NGN (Next Generation Network), but the large SIP message which is text-based is an obstacle with the planned usage of SIP in wireless mobile networks. Based on the SigComp (Signaling Compression) framework, some further improvements are made to the Deflate algorithm according to the characteristics of SIP in this study. Experiments show that the improved Deflate algorithm can compress the SIP message greatly and reduce the bandwidth requirements signally, so it is highly valued in IMS (IP Multimedia Subsystem), PTT (Push To Talk) and other wireless real-time SIP applications.

In this article, a Session Initiation Protocol (SIP)-based instant messaging system is proposed to give an open protocol for instant messaging software to communicate with multi-platform and multi-service interoperability. The system will solve the issue of compatibility between current, mainstream, communication software. The system is based mainly on a multi-media scheduling server. This article is focused on the design of the SIP channel module, the instant messaging module, and presence service module with the goal of eventually achieving the function of instant communication between clients.

Voice over Internet Protocol (VoIP) uses Internet Protocol (IP) to transmit voice as packets over an IP network. It achieves desired functionality of Internet telephony using a signaling protocol known as Session Initiation Protocol (SIP). When users need to use SIP service, first server authenticates the user in order to provide the service. In this paper, a new and secure authentication scheme for SIP is proposed. Its major merits are 1) Provides mutual authentication. 2) Generates session key agreed between user and server in two steps. 3) Secure against various possible attacks induced by IP networks. Session Initiation Protocol (SIP) is a signaling protocol operating at application layer to initiate, maintain and terminates multimedia sessions across packet networks. Internet Engineering Task Force (IETF) proposed SIP as a signaling protocol for Internet Protocol (IP) based telephony. SIP is designed to be independent of underlying transport layer. It can operate on Transmission Control Protocol (TCP) as well as User Datagram Protocol (UDP) to handles all signaling requirements of Voice over Internet Protocol (VoIP) sessions. Today it is widely used to transmit voice and video over IP. Issue of security has become extremely important in today's computer networks environment. Two fundamental security services required by SIP are confidentiality and authentication. Whenever user wants to access SIP service from server, mutual authentication is required between two parties. An attacker can obtain user's secret information by forging the identity of server if mutual authentication is not performed. Confidentiality is usually provided by means of encryption. Only intended recipient can decrypt a message and obtain a tangible meaning out of it. Encryption/Decryption uses shared secrets agreed among the communicating entities. If it is different from session to each session, it is known as session key. Identifying caller and callee is an utmost important issue in SIP based application. To guarantee and enhance security features, several authentication schemes have been proposed (1)-(8). Rest of the paper is organized as follows. Section II presents an overview of previous work. Section III describes the proposed SIP authentication scheme. Section IV discusses security analysis of proposed scheme. Section V presents a comparative analysis of security features and

Security of session initiation protocol (SIP) servers is a serious concern of Voice over Internet (VoIP) vendors. The important contribution of our paper is an accurate and real-time attack classification system that detects: (1) application layer SIP flood attacks that result in denial of service (DoS) and distributed DoS attacks, and (2) Spam over Internet Telephony (SPIT). The major advantage of our framework over existing schemes is that it performs packet-based analysis using a set of spatial and temporal features. As a result, we do not need to transform network packet streams into traffic flows and thus save significant processing and memory overheads associated with the flow-based analysis. We evaluate our framework on a real-world SIP traffic—collected from the SIP server of a VoIP vendor—by injecting a number of application layer anomalies in it. The results of our experiments show that our proposed framework achieves significantly greater detection accuracy compared with existing state-of-the-art flooding and SPIT detection schemes.

Traditionally, call authentication and security have not raised user concerns because wiretapping requires physical access to the phone line and special equipment. However, Voice over Internet Protocol VoIP communications are becoming increasingly popular, and there is the perception that they may be easier to intercept or impersonate thus creating higher demand for security solutions, especially if the connection occurs over wireless links. The Session Initiation Protocol is widely used for managing voice and video communications over the Internet, and the Real-time Transport Protocol is used to carry voice and/or video streams. Session Initiation Protocol, however, was not designed with security in mind and is vulnerable toattacks.

In this modern era, communication through internet is popular. Business exchange on internet phone is thriving. However, current internet phones in the market do not guarantee voice quality. The purpose of this paper is to ascertain the effectiveness of different internet voice packet techniques towards voice quality and to select the technique that is more reliable and to realize closer to the original sound quality. Surveys were conducted Quality of Service for comparison of SIP and Red5 flow, and on voice quality acceptance level of the general public. The study concluded that: (1) while Red5 internet phone has higher transmission quality and occupying less web band, its voice quality is undesirable; (2) although SIP internet phone occupies larger band and its packets drops easily, voice quality is higher than Red5.

For more secure voice communication over Internet, the secure voice over internet protocol is proposed. The key agreement and the cipher text of voice packets transmission processes are related to the SIP, SDP and UDP. Parameters are negotiated using the SDP during the calling process for connection and key agreement, which is transported in the SIP packet body. The voice data are encrypted in groups and transported to secure communication server using UDP. Secure communication server forwards the cipher text packets to the receiver. Each secure voice communication over internet owns a different session key generated randomly and a different encryption key generated on the basis of key seeds according to random numbers and time-stamp, which guarantees security over Internet for voice communication users.

The implementation of the Session Initiation Protocol (SIP)-based Voice over Internet Protocol (VoIP) and multimedia over MANET is still a challenging issue. Many routing factors affect the performance of SIP signaling and the voice Quality of Service (QoS). Node mobility in MANET causes dynamic changes to route calculations, topology, hop numbers, and the connectivity status between the correspondent nodes. SIP-based VoIP depends on the caller’s registration, call initiation, and call termination processes. Therefore, the SIP signaling performance has an important role for the overall QoS of SIP-based VoIP applications for both IPv4 and IPv6 MANET. Different methods have been proposed to evaluate and benchmark the performance of the SIP signaling system. However, the efficiency of these methods vary and depend on the identified performance metrics and the implementation platforms. This survey examines the implementation of the SIP signaling system for VoIP applications over MANET and highlights the available performance enhancement methods.

In order to integrate the telecommunications network and IP network, providing companies with a unified voice services, a design of IP PBX phone system is proposed. On the basis of VoIP, this design implements most of the traditional PBX functions. It provides function-code functionality allows user to directly control phone; provides user-trunk-LLDC configurations to manage user’s privilege; provides a powerful and flexible IVR system; provides SIP phone and SIP trunk support. It also supports most call features such as black list, speed dial code, forward, transfer, pick-up, can meet most companies’ requirements.

Problem statement: This study aims to discuss several issues on session initiation protocol security and threats. An in-depth investigation related to SIP with the intention to categorize the wide variety of SIP security issues. Approach: Related papers to the infrastructure of SIP security were analyzed. Some of the identified issues are: Social threats, eavesdropping, delaying, modification of media session, service abuse threats, physical access threats and denied services threats. Results and Conclusion: A useful categorization of SIP security issues has been done. The vulnerabilities of existing SIP infrastructure and possible remedies are discussed. It is confirmed that, message attacks are the most dominant category of SIP attacks.

Fixed and mobile telephony with other telecommunication services are moving to a fundamentally different infrastructure as the transition proceeds from the public telephone network to the Internet. There are many opportunities to make significant progress on major developments such as separation of operator independent functions, modular design, and the integration of other desirable features of sustainable Internet solutions. This progress also means that the impacts of some shortcomings of currently competitive telephony services will decline—another benefit of the transition. However, because telephony services are essentially being “emulated” over the Internet infrastructure, other aspects of these services will be changed in ways that could affect the public good. In particular, trust-related issues such as authentication and validation, along with their business implications, present significant challenges. This paper explores these issues, including the opportunity that industry and the regulators now have to address them, and suggests approaches to sustainable solutions that can benefit all the stakeholders in this transition.

The IP multimedia subsystem (IMS) is a basis for a significant new architecture which offers network operators the opportunity to expand their services, by integrating voice and multimedia communications and delivering them into new environments with new purposes. Basically, the IMS is an overlay network on top of IP layer that uses Session Initiation Protocol (SIP) as the primary signaling mechanism. SIP works at the application layer in IP networks. It is thus faced to not only the IP-networks security issues, but also to new issues which are related to the SIP protocol directly. Consequently, using IMS bears several new security challenges. This paper presents the most relevant SIP-related security vulnerabilities and threats, and the implementation and simulation test bed to experiment two versions of the SIP Asterisk software to emphasize these threats. The different security mechanisms that can be deployed to overcome the SIP security issues while putting emphasis the most important ones are discussed. Afterwards, the authors propose adaptable solutions to the SIP threats already identified for a specific service (access information from anywhere) in IMS context. Finally, conclusions are drawn and some perspectives are introduced to improve the security of multimedia applications.

In recent years, Internet Protocol (IP) telephony has been a real alternative to the traditional Public Switched Telephone Networks (PSTN). IP telephony offers more flexibility in the implementation of new features and services. The Session Initiation Protocol (SIP) is becoming a popular signalling protocol for Voice over IP (VoIP) based applications. The SIP proxy server is a software application that provides call routing services by parsing and forwarding all the incoming SIP packets in an IP telephony network. The efficiency of this process can create large scale, highly reliable packet voice networks for service providers and enterprises. We established that the efficient design and implementation of the SIP proxy server architecture can enhance the performance characteristics of a SIP proxy server significantly. Since SIP proxy server performance can be characterised by its transaction states of each SIP session, we emulated the M/M/1 performance model of the SIP proxy server and studied some of the key performance benchmarks such as average response time to process the SIP calls, and mean number of SIP calls in the system. We showed its limitations, and then studied an alternative M/M/c based SIP proxy server performance model with enhanced performance model and studied additional key performance characteristics such as server utilisation, queue size and memory utilisation. Provided the comparative results between the predicted results with the experimental results conducted in a lab environment.