1,624 research outputs found
Providing End-to-End Connectivity to SIP User Agents Behind NATs
The widespread diffusion of private networks in SOHO scenarios is fostering an increased deployment of Network Address Translators (NATs). The presence of NATs seriously limits end-to-end connectivity and prevents protocols like the Session Initiation Protocol (SIP) from working properly. This document shows how the Address List Extension (ALEX), which was originally developed to provide dual-stack and multi-homing support to SIP, can be used, with minor modifications, to ensure end-to-end connectivity for both media and signaling flows, without relying on intermediate relay nodes whenever it is possibl
Roaming Real-Time Applications - Mobility Services in IPv6 Networks
Emerging mobility standards within the next generation Internet Protocol,
IPv6, promise to continuously operate devices roaming between IP networks.
Associated with the paradigm of ubiquitous computing and communication, network
technology is on the spot to deliver voice and videoconferencing as a standard
internet solution. However, current roaming procedures are too slow, to remain
seamless for real-time applications. Multicast mobility still waits for a
convincing design. This paper investigates the temporal behaviour of mobile
IPv6 with dedicated focus on topological impacts. Extending the hierarchical
mobile IPv6 approach we suggest protocol improvements for a continuous
handover, which may serve bidirectional multicast communication, as well. Along
this line a multicast mobility concept is introduced as a service for clients
and sources, as they are of dedicated importance in multipoint conferencing
applications. The mechanisms introduced do not rely on assumptions of any
specific multicast routing protocol in use.Comment: 15 pages, 5 figure
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The Impact of TLS on SIP Server Performance
This report studies the performance impact of using TLS as a transport protocol for SIP servers. We evaluate the cost of TLS experimentally using a testbed with OpenSIPS, OpenSSL, and Linux running on an Intel-based server. We analyze TLS costs using application, library, and kernel profiling, and use the profiles to illustrate when and how different costs are incurred, such as bulk data encryption, public key encryption, private key decryption, and MAC-based verification. We show that using TLS can reduce performance by up to a factor of 20 compared to the typical case of SIP over UDP. The primary factor in determining performance is whether and how TLS connection establishment is performed, due to the heavy costs of RSA operations used for session negotiation. This depends both on how the SIP proxy is deployed (e.g., as an inbound or outbound proxy) and what TLS options are used (e.g., mutual authentication, session reuse). The cost of symmetric key operations such as AES or 3DES, in contrast, tends to be small. Network operators deploying SIP over TLS should attempt to maximize the persistence of secure connections, and will need to assess the server resources required. To aid them, we provide a measurement-driven cost model for use in provisioning SIP servers using TLS. Our cost model predicts performance within 15 percent on average
Towards a scalable video interactivity solution over the IMS
Includes bibliographical references (leaves 72-76).Rapid increase in bandwidth and the interactive and scalability features of the Internet provide a precedent for a converged platform that will support interactive television. Next Generation Network platforms such as the IP Multimedia Subsystem (IMS) support Quality of Service (QoS), fair charging and possible integration with other services for the deployment of IPTV services. IMS architecture supports the use of the Session Initiation Protocol (SIP) for session control and the Real Time Streaming Protocol (RTSP) for media control. This study aims to investigate video interactivity designs over the Internet using an evaluation framework to examine the performance of both SIP and RTSP protocols over the IMS over different access networks. It proposes a Three Layered Video Interactivity Framework (TLVIF) to reduce the video processing load on a server
SIP protocol model for OMNeT++
The article describes our new SIP protocol implementation for the OMNeT++ simulation framework. OMNeT++ simulation framework provides an extensive support of IP related protocols, nevertheless a working SIP protocol implementation is missing. Real measurements were also done using a SIPp traffic generator and the results are compared to those obtained by our new SIP model. Since this work is a part of bigger project concerned strictly on measuring "first response times" over networks with a faulty transmission links, the actually collected statistics are focused only this way
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