SIP Research Papers - Academia.edu (original) (raw)

The telecommunications industry has undergone major paradigm shifts in previous years. One of these is the shift from circuit-switched networks towards packet-switched networks. There is hardly any doubt that IP will be the ubiquitous... more

The telecommunications industry has undergone major paradigm shifts in previous years. One of these is the shift from circuit-switched networks towards packet-switched networks. There is hardly any doubt that IP will be the ubiquitous transport protocol for multimedia of the future. At present, the work towards Internet telephony has attracted a lot of attention. Two standards have been developed: SIP (session initiation protocol) by the Internet Engineering Task Force (IETF) and H.323 by the International Telecommunications Union (ITU). The rapid deployment of services on top of SIP is hindered by a well known problem, feature interactions, also well known in traditional telephony networks. In this paper, the technological changes in SIP compared to the traditional telephony network are highlighted as well as the impact of the changed business environment. In a multi-service provider environment (as encouraged by a SIP environment), the feature interaction problem is regarded to be an even more pressing issue. Services of different providers from a multitude of service creators are likely to interwork. This in turn leads to the fact that the ad-hoc approaches to service interaction handling, commonly applied in the industry today, will not be sufficient any more. In this paper, a new approach addressing these changes is introduced and the results of a case study are presented

—Denial of Service (DoS) or Distributed Denial of Service (DDoS) is a powerful attack which prevents the system from providing services to its legitimate users. Several approaches exist to filter network-level attacks, but... more

—Denial of Service (DoS) or Distributed Denial of Service (DDoS) is a powerful attack which prevents the system from providing services to its legitimate users. Several approaches exist to filter network-level attacks, but application-level attacks are harder to detect at the firewall. Filtering at application level can be computationally expensive and difficult to scale, while still creating bogus positives that block legitimate users. In this paper, authors show application layer DoS attack for SIP server using some open source DoS attack tools and also suggest a mechanism that can protect a given SIP server from application-level DoS attacks especially the attacks targeting the resources including CPU, sockets, memory of the victim server. In this paper author's attempt to illustrate application layer distributed denial of Service (DDoS) attack on SIP Server such as SIP flooding attack, real time transport (RTP) flooding attack using open source DDoS attack tools. We propose a new DDoS defence mechanism that protects SIP servers from application-level DDoS attacks based on the two methodologies: IPtables and fail2ban detection. The attack flow detection mechanism detects attach flows based on the symptom or stress at the server, since it is getting more difficult to identify bad flows only based on the incoming traffic patterns. A popular software known as Wireshark which is a network protocol analyzer is used to capture the packets during DoS attack from the victim server Ethernet interface to detect the attacking host IP address and analysis the types of attack. We evaluate the performance of the proposed scheme via experiment.

WebRTC technology that offers high eminence RTC applications being established for web and mobile platforms and permit them to interconnect via API's and similarly with a set of practices. WebRTC deliberations to be focused between at... more

WebRTC technology that offers high eminence RTC applications being established for web and mobile platforms and permit them to interconnect via API's and similarly with a set of practices. WebRTC deliberations to be focused between at least two endpoints by means of program based versatile/work area applications or gadget local portable applications. This paper explains about enabling online shoppers to have live audio/video chats through mobile with the vendors so that they can look at the product and clarity the uncertainties on spot, as though they are feeling outdoor shopping, the paper also proves such competence by leveraging evolving expertise like WebRTC and WSC

Voice over Internet Protocol (VoIP) is a new communication technology that uses internet protocol in providing phone services. VoIP provides various forms of benefits such as low monthly fee and cheaper rate in terms of long distance and... more

Voice over Internet Protocol (VoIP) is a new communication technology that uses internet protocol in providing phone services. VoIP provides various forms of benefits such as low monthly fee and cheaper rate in terms of long distance and international calls. However, VoIP is accompanied with novel security threats. Criminals often take advantages of such security threats and commit illicit activities. These activities require digital forensic experts to acquire, analyses, reconstruct and provide digital evidence. Meanwhile, there are various methodologies and models proposed in detecting, analysing and providing digital evidence in VoIP forensic. However, at the time of writing this paper, there is no model formalized for the reconstruction of VoIP malicious attacks. Reconstruction of attack scenario is an important technique in exposing the unknown criminal acts. Hence, this paper will strive in addressing that gap. We propose a model for reconstructing VoIP malicious attacks. To achieve that, a formal logic approach called Secure Temporal Logic of Action(S-TLA+) was adopted in rebuilding the attack scenario. The expected result of this model is to generate additional related evidences and their consistency with the existing evidences can be determined by means of S-TLA+ model checker.

This paper highlights the design and implementation aspects of a VoIP based asterisk voice exchange, developing a fully functional voice exchange requires to set up a server based on Asterisk, connecting clients to that server with the... more

This paper highlights the design and implementation aspects of a VoIP based asterisk voice exchange, developing a fully functional voice exchange requires to set up a server based on Asterisk, connecting clients to that server with the help of soft phones and then configuring the soft phones with the server. Here in our implementation we have connected the clients to the server with the help of IAX protocols. The first part of the paper contains some introductory concepts about VoIP, followed by asterisk's internal architecture. In the third part of the paper we discuss about the codecs and protocols used by the packet switching based PBX networks and finally we brush up about the design and implementation aspects.

This paper compares two most important protocols for signaling in voice over IP (VoIP) networks of today, namely SIP and H. 323. Several aspects of protocol functionality are covered. We have tried to focus on those aspects that are most... more

This paper compares two most important protocols for signaling in voice over IP (VoIP) networks of today, namely SIP and H. 323. Several aspects of protocol functionality are covered. We have tried to focus on those aspects that are most important for ...

IP Multimedia Subsystem (IMS) is considered to be one of the important features in Mobile Next Generation Networks (MNGN). It adds value to the mobile services and applications by integrating mobile network resources, such as location,... more

IP Multimedia Subsystem (IMS) is considered to be one of the important features in Mobile Next Generation Networks (MNGN). It adds value to the mobile services and applications by integrating mobile network resources, such as location, billing and authentication. This is achieved by enabling a third party access to network resources. In previous work [1] we have presented a testbed to be used as platform for testing mobile application prior to actual deployment. We have chosen a novel IMS based MObile Mass EXamination (MOMEX) system to showcase the benefit of designing an IMS based mobile application. We identify two aspects essential to of the application namely security threats and delay analysis. In this paper we identify MOMEX security threats and suggest strategies to mitigate system vulnerabilities. We then evaluate the performance of MOMEX system in terms of delay and security threats and vulnerabilities. The results presented show system performance limitation and tradeoffs.

This paper presents the design and development of a Linux based service platform for a wholesale service provider, which provides a web callback or click-to-call service to retail service providers. A web callback service, in its simplest... more

This paper presents the design and development of a Linux based service platform for a wholesale service provider, which provides a web callback or click-to-call service to retail service providers. A web callback service, in its simplest form, provides a button on a website that allows users to enter their telephone numbers, and receive callback calls from the call center operators. Web callback services are usually premium voice services that enable customers to make contacts by voice, from anywhere in the world, at no cost. Similar to the traditional toll free telephone service, the full cost of a web callback is paid by the website owner. This paper describes the design of such a commercial grade platform, including hardware requirement, database design, Voice over IP (VoIP) application development, voice infrastructure for interconnection with upstream providers via ISDN PRI, VoIP for worldwide termination, and APIs (application programming interface) for interconnection with multiple retail providers.

Presence information - expressing user willingness and ability to communicate with other users across a set of devices and tools - represents an essential prerequisite for real-time communications. Although presence-related protocols have... more

Presence information - expressing user willingness and ability to communicate with other users across a set of devices and tools - represents an essential prerequisite for real-time communications. Although presence-related protocols have primarily emerged in conjunction with instant messaging systems, presence is today viewed as a primer service offered by contemporary unified communication platforms, and is often referred to as the dial tone of the 21st century. The paper presents Presence@FER, an ecosystem for presence based on the Rich Presence Information Data format (RPID) which comprises both XMPP and SIP-based solutions for presence. Presence@FER supports context-aware collection and exposure of rich presence information, and offers fine-grained filtering of presence information in accordance with user context and predefined policies. It consolidates virtual, physical, and online presence into a single rich-presence platform which relies on content-based publish/subscribe service for efficient filtering and dissemination of presence information. We show a number of applications that can be built of top of such rich-presence platform, and provide details of our prototype implementation.