Research of Session Initial Protocol Security Mechanism on Public Key Technique

Two common kinds of security mechanisms used in session initial protocol （S1P） are analyzed. An improved HTTP digest authentication scheme is put forward based on the existing SIP authentication theories. This mechanism is combined with the merits of the HTTP digest authentication and the public key encryption, so the communicating parties complete two-way authentication and public key exchange in pre-calling, and the session key can be randomly generated in post-calling. The mixture of security encryption mechanism with public key encryption and symmetric-key encryption algorithm can ensure the security for network communication data. The emulation of the scheme is verified, and the security analysis is conducted in the end. The researches show that the simulations efficiency of this method is about 78% of HTTP＇s, and it can prevent four kinds of attacks including impersonating a server, offline password guessing attacks, relay-attack, and session monitoring.

Verification of Session Initiation Protocol Using Timed Colored Petri Net

In this work, Session Initiation Protocol model is established by using Timed Colored Petri Nets (TCPN). SIP (Session Initiation Protocol) is a protocol developed to assist in providing advanced telephony services across the Internet. The Session Initiation Protocol (SIP) has become the quasi-standard for Voiceover-Internet Protocol (VoIP) communications. SIP is based on a client–server infrastructure in which user agents represent the end-terminals as clients, proxy servers handle SIP message routing between the user agents, and registrar servers store the client’s contact information into a location service. By use of timed color set and useful time attributes in tokens defined in CPN tools, timer and time-related problems of SIP are modeled and analyzed. Timer is an important part for SIP, especially the INVITE transaction.

An ALM Model for Streaming Media Service in P2P-SIP Network

Application layer multicast (ALM) has been widely applied in Internet, as a substitute for IP multicast. However, it causes network load to increase because it adopts unicast in data transmission, which limits the application. In this article, in order to improve the ALM performance in P2P-SIP network, an ALM model was proposed which reduced network load via merging messages in concast mode. Finally network simulations prove that this model has better benefit on single node load and average network load. Therefore, it is suitable for streaming media service in P2P-SIP Network.

A P2PSIP System with Intelligent Routing Function on the Media Plane

Decentralized peer-to-peer session initiation protocol （P2PSIP） provides the same services as legacy SIPs such as IMS. However, in relatively open network, the requirement for route efficiency in a complex environment brings about undefined problems. To deploy a controllable P2PSIP network, perfect mechanisms have to be appended, especially in QoS, security, and management. Several proposals for QoS, network address translation （NAT）, and interworking have been put forward. In this paper, we propose an integrated architecture for a P2PSIP system as well as a proactive intelligent routing scheme on the media plane used in system. Implementation and simulation show that our solution is suitable for operation and management.

A Component-Based Conference Control Model and Implementation for Loosely Coupled Sessions

Conference control is a very important core part to compose a completeInternet multimedia conference system and has been a hot research area over the years, but there arecurrently no -widely accepted robust and scalable solutions and standards. This paper proposes acomponent-based conference control model for loosely coupled sessions in which media applicationscan collaborate with a Session Controller (SC) to provide the conference control. A SC prototype hasbeen built.

SIP Conformance Testing Based on TTCN-2

The session initiation protocol (SIP) is a signaling protocol for Internet telephony, multimedia conferencing, presence, event notification, and instant messaging. With the gaining popularity, more and more SIP implementations have been developed and deployed. How to guarantee the conformance of those SIP implementations is the key point of interconnection and interoperation among them. This paper proposes the test method and architecture for the SIP protocol based on the IPv6 tester system. Tree and tabular combined notation (TTCN-2) is adopted to describe the test suite. With an enhanced reference implementation, the data-processing ability to the original test system has been greatly improved. In the following test practices, some errors have been found in the SIP entities under test. It is proved that the proposed test method and architecture are effective to verify the conformance of the SIP entities, and the result of the conformance test may provide helpful reference to the development of SIP products.