ALEX: Improving SIP Support in Systems with Multiple Network Addresses

The successful and increasingly adopted session initiation protocol (SIP) does not adequately support hosts with multiple network addresses, such as dual-stack (IPv4-IPv6) or IPv6 multi-homed devices. This paper presents the Address List Extension (ALEX) to SIP that adds effective support to systems with multiple addresses, such as dual-stack hosts or multi-homed IPv6 hosts. ALEX enables IPv6 transport to be used for SIP messages, as well as for communication sessions between SIP user agents (UAs), whenever possible and without compromising compatibility with ALEX-unaware UAs and SIP servers

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

Delivering Live Multimedia Streams to Mobile Hosts in a Wireless Internet with Multiple Content Aggregators

We consider the distribution of channels of live multimedia content (e.g., radio or TV broadcasts) via multiple content aggregators. In our work, an aggregator receives channels from content sources and redistributes them to a potentially large number of mobile hosts. Each aggregator can offer a channel in various configurations to cater for different wireless links, mobile hosts, and user preferences. As a result, a mobile host can generally choose from different configurations of the same channel offered by multiple alternative aggregators, which may be available through different interfaces (e.g., in a hotspot). A mobile host may need to handoff to another aggregator once it receives a channel. To prevent service disruption, a mobile host may for instance need to handoff to another aggregator when it leaves the subnets that make up its current aggregator�s service area (e.g., a hotspot or a cellular network).\ud In this paper, we present the design of a system that enables (multi-homed) mobile hosts to seamlessly handoff from one aggregator to another so that they can continue to receive a channel wherever they go. We concentrate on handoffs between aggregators as a result of a mobile host crossing a subnet boundary. As part of the system, we discuss a lightweight application-level protocol that enables mobile hosts to select the aggregator that provides the �best� configuration of a channel. The protocol comes into play when a mobile host begins to receive a channel and when it crosses a subnet boundary while receiving the channel. We show how our protocol can be implemented using the standard IETF session control and description protocols SIP and SDP. The implementation combines SIP and SDP�s offer-answer model in a novel way

A new scheme to reduce session establishment time in session initiation protocol (SIP)

The session Initiation Protocol (SIP) has been developed by Internet Engineering Taskforce standard (IETF) with the main purpose of establishing and managing sessions between two or more parties wishing to communicate. SIP is a signaling protocol which is used for the current and future Internet Protocol (IP) telephony services, video services, and integrated web and multimedia services. SIP is an application layer protcol, thus it can run over Transmission Control Protocol(TCP) or User Datagram Protocol (UDP). When the packets are sent over the network, a form of congestion control mechanism is necessary to prevent from network collapse. TCP is a reliable protocl and provides the congestion control by adjusting the size of the congestion windows. UDP is an unreliable protocol and no flow control mechanism is provided. Many applications of the Internet require the establishment and management of sessions. The purpose of the thesis is to study the session establishnment procedure in SIP and try to reduce the time taken for the session setup in two different conditions. One, when there is no congestion in the network, and the other is when there is a network congestion. We have simulated the behaviour of session establishment in SIP using Network Simulator (NS2). UDP is used as the transport protocol. We have created different network topologies. In the topology we had created SIP user agents who wants to communicte, proxy servers for forwarding the requests on behalf of the user agents, and a Domain Name Server (DNS) which maintains the location information of all proxy servers. We tried to reduce the time taken for the session establishment. As UDP does not provide any congestion control mechanisms, we used the binary exponential backoff (BEB) algorithm to set the timers. In our network topolgy when there is no packet loss in the network, the time taken for the session establishment is reduced from 0.86 sec to 0.574 sec. In case of network congestion the setup time is reduced from 4.55 sec to 2.86 sec. From the simulation, we conclude that the session establishment time can be reduced by reducing the number of message exchanges required for session setup

Study, Analysis and Modeling of IP Multimedia systems on next generation networks providing mobile and fixed multimedia services

Not availabl

DCCP Simultaneous-Open Technique to Facilitate NAT/Middlebox Traversal

https://datatracker.ietf.org/doc/rfc5595/Publisher PD

An Architecture for Global Distributed SIP Network Using IPv4 Anycast

Tato diplomová práce se zabývá metodami pro výběr nejbližší RTP proxy k VoIP klientům s použitím IP anycastu. RTP proxy servery jsou umístěny v síti Internetu a přeposílají RTP data pro VoIP klienty za síťovými překladači adres(NAT). Bez zeměpisně rozmístěných RTP proxy serverů a metod pro nalezení nejbližšího RTP proxy serveru by došlo ke zbytečnému poklesu kvality přenosu médialních dat a velkému zpoždení. Tento dokument navrhuje 4 metody a jejich porovnání s podrobnějšími rozbory metod s využitím DNS resolvování a přímo SIP protokolu. Tento dokument také obsahuje měření chování IP anycastu v porovnání mezi metrikami směrování a metrikami časovými. Nakonec dokumentu je také uvedena implemetace na SIP Express Router platformě.This thesis is about using IP anycast-based methods for locating RTP proxy servers close to VoIP clients. The RTP proxy servers are hosts on the public Internet that relay RTP media between VoIP clients in a way that accomplishes traversal over Network Address Translators (NATs). Without geographically-dispersed RTP proxy servers and methods to find one in client's proximity, voice latency may be unbearably long and dramatically reduce perceived voice quality. This document proposes four methods their comparison with further design of DNS-based and SIP-based methods. It includes IP anycast measurements that provides an overview of IP anycast behaviour in terms of routing metrics and latency metrics. It also includes implementation on SIP Express Router platform.

Managing ClientInitiated Connections

The Session Initiation Protocol (SIP) allows proxy servers to initiate TCP connections or to send asynchronous UDP datagrams to User Agents in order to deliver requests. However, in a large number of real deployments, many practical considerations, such as the existence of firewalls and Network Address Translators (NATs) or the use of TLS with server-provided certificates, prevent servers from connecting to User Agents in this way. This specification defines behaviors for User Agents, registrars, and proxy servers that allow requests to be delivered on existing connections established by the User Agent. It also defines keep-alive behaviors needed to keep NAT bindings open and specifies the usage of multiple connections from the User Agent to its registrar. Status of This Memo This document specifies an Internet standards track protocol for the Internet community, and requests discussion and suggestions for improvements. Please refer to the current edition of the "Internet Official Protocol Standards " (STD 1) for the standardization state and status of this protocol. Distribution of this memo is unlimited. Copyright Notice Copyright (c) 2009 IETF Trust and the persons identified as th

MRTP - A novel real-time protocol for patient-doctor online meetings and monitoring

This paper presents the design of a novel Voice over IP (VoIP) protocol that carries real-time voice and medical data in the same stream while preserving good quality of service (QoS) for online medical care. The work deliberates the reasons for the choice of the Session Initiation Protocol (SIP). The proposed medical real-time protocol (MRTP) is designed to be compatible with the existing Real Time Protocol (RTP) and has advanced features for medical data transmission. The MRTP leverages the real-time performance due to less overhead i.e., less delays. A major challenge is that some medical data is life-critical i.e., losses may cause harm. Hence, reliability and satisfactory QoS are crucial. A major result is the decrease in average-delay by around 74% compared to concurrently utilizing RTP for the audio part and TCP-connections (sockets) for the same medical data.Peer ReviewedPostprint (author's final draft