Duplicating RTP streams

Packet loss is undesirable for real-time multimedia sessions but can occur due to a variety of reasons including unplanned network outages. In unicast transmissions, recovering from such an outage can be difficult depending on the outage duration, due to the potentially large number of missing packets. In multicast transmissions, recovery is even more challenging as many receivers could be impacted by the outage. For this challenge, one solution that does not incur unbounded delay is to duplicate the packets and send them in separate redundant streams, provided that the underlying network satisfies certain requirements. This document explains how Real-time Transport Protocol (RTP) streams can be duplicated without breaking RTP or RTP Control Protocol (RTCP) rule

Quality of Service optimisation framework for Next Generation Networks

Within recent years, the concept of Next Generation Networks (NGN) has become widely accepted within the telecommunication area, in parallel with the migration of telecommunication networks from traditional circuit-switched technologies such as ISDN (Integrated Services Digital Network) towards packet-switched NGN. In this context, SIP (Session Initiation Protocol), originally developed for Internet use only, has emerged as the major signalling protocol for multimedia sessions in IP (Internet Protocol) based NGN. One of the traditional limitations of IP when faced with the challenges of real-time communications is the lack of quality support at the network layer. In line with NGN specification work, international standardisation bodies have defined a sophisticated QoS (Quality of Service) architecture for NGN, controlling IP transport resources and conventional IP QoS mechanisms through centralised higher layer network elements via cross-layer signalling. Being able to centrally control QoS conditions for any media session in NGN without the imperative of a cross-layer approach would result in a feasible and less complex NGN architecture. Especially the demand for additional network elements would be decreased, resulting in the reduction of system and operational costs in both, service and transport infrastructure. This thesis proposes a novel framework for QoS optimisation for media sessions in SIP-based NGN without the need for cross-layer signalling. One key contribution of the framework is the approach to identify and logically group media sessions that encounter similar QoS conditions, which is performed by applying pattern recognition and clustering techniques. Based on this novel methodology, the framework provides functions and mechanisms for comprehensive resource-saving QoS estimation, adaptation of QoS conditions, and support of Call Admission Control. The framework can be integrated with any arbitrary SIP-IP-based real-time communication infrastructure, since it does not require access to any particular QoS control or monitoring functionalities provided within the IP transport network. The proposed framework concept has been deployed and validated in a prototypical simulation environment. Simulation results show MOS (Mean Opinion Score) improvement rates between 53 and 66 percent without any active control of transport network resources. Overall, the proposed framework comes as an effective concept for central controlled QoS optimisation in NGN without the need for cross-layer signalling. As such, by either being run stand-alone or combined with conventional QoS control mechanisms, the framework provides a comprehensive basis for both the reduction of complexity and mitigation of issues coming along with QoS provision in NGN

Sending multiple RTP streams in a single RTP session

This memo expands and clarifies the behavior of Real-time Transport Protocol (RTP) endpoints that use multiple synchronization sources (SSRCs). This occurs, for example, when an endpoint sends multiple RTP streams in a single RTP session. This memo updates RFC 3550 with regard to handling multiple SSRCs per endpoint in RTP sessions, with a particular focus on RTP Control Protocol (RTCP) behavior. It also updates RFC 4585 to change and clarify the calculation of the timeout of SSRCs and the inclusion of feedback messages

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

Un-constraining the medium: design software systems to support situated action

This dissertation is concerned with Computer Supported Cooperative Work (CSCW) and in particular with ways in which insights from ethnomethodology can be melded into the design of CSCW systems—a relationship that has been labelled technomethodology. The dissertation outlines a number of possible ways in which system design can learn from ethnomethodology and concentrates on one particular aspect—namely that CSCW should look closely at its foundational assumptions and, if necessary, re-specify any concepts which appear problematic in their formulation. [Continues.

Guidelines for using the multiplexing features of RTP to support multiple media streams

The Real-time Transport Protocol (RTP) is a flexible protocol that can be used in a wide range of applications, networks, and system topologies. That flexibility makes for wide applicability but can complicate the application design process. One particular design question that has received much attention is how to support multiple media streams in RTP. This memo discusses the available options and design trade-offs, and provides guidelines on how to use the multiplexing features of RTP to support multiple media streams

Sending Multiple Types of Media in a Single RTP Session

This document specifies how an RTP session can contain RTP streams with media from multiple media types such as audio, video, and text. This has been restricted by the RTP specifications (RFCs 3550 and 3551), and thus this document updates RFCs 3550 and 3551 to enable this behaviour for applications that satisfy the applicability for using multiple media types in a single RTP session