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

Multimedia congestion control: circuit breakers for unicast RTP sessions

The Real-time Transport Protocol (RTP) is widely used in telephony, video conferencing, and telepresence applications. Such applications are often run on best-effort UDP/IP networks. If congestion control is not implemented in these applications, then network congestion can lead to uncontrolled packet loss and a resulting deterioration of the user's multimedia experience. The congestion control algorithm acts as a safety measure by stopping RTP flows from using excessive resources and protecting the network from overload. At the time of this writing, however, while there are several proprietary solutions, there is no standard algorithm for congestion control of interactive RTP flows. This document does not propose a congestion control algorithm. It instead defines a minimal set of RTP circuit breakers: conditions under which an RTP sender needs to stop transmitting media data to protect the network from excessive congestion. It is expected that, in the absence of long-lived excessive congestion, RTP applications running on best-effort IP networks will be able to operate without triggering these circuit breakers. To avoid triggering the RTP circuit breaker, any Standards Track congestion control algorithms defined for RTP will need to operate within the envelope set by these RTP circuit breaker algorithms

Design, Development and Assessment of Control Schemes for IDMS in a Standardized RTCP-based Solution

[EN] Currently, several media sharing applications that allow social interactions between distributed users are gaining momentum. In these networked scenarios, synchronized playout between the involved participants must be provided to enable truly interactive and coherent shared media experiences. This research topic is known as Inter-Destination Media Synchronization (IDMS). This paper presents the design and development of an advanced IDMS solution, which is based on extending the capabilities of RTP/RTCP standard protocols. Particularly, novel RTCP extensions, in combination with several control algorithms and adjustment techniques, have been specified to enable an adaptive, highly accurate and standard compliant IDMS solution. Moreover, as different control or architectural schemes for IDMS exist, and each one is best suited for specific use cases, the IDMS solution has been extended to be able to adopt each one of them. Simulation results prove the satisfactory responsiveness of our IDMS solution in a small scale scenario, as well as its consistent behavior, when using each one of the deployed architectural schemes.This work has been financed, partially, by Universitat Politecnica de Valencia (UPV), under its R&D Support Program in PAID-01-10. TNO's work has been partially funded by European Community's Seventh Framework Programme (FP7/2007-2013) under Grant Agreement No. ICT-2011-8-318343 (STEER Project). CWI's work has been partially funded by the European Community's Seventh Framework Programme (FP7/2007-2013) under Grant Agreement No. ICT-2011-7-287723 (REVERIE Project).Montagud Aguar, M.; Boronat Segui, F.; Stokking, H.; Cesar, P. (2014). Design, Development and Assessment of Control Schemes for IDMS in a Standardized RTCP-based Solution. Computer Networks. 70:240-259. https://doi.org/10.1016/j.comnet.2014.06.004S2402597

Enhanced adaptive RTCP-based inter-destination multimedia synchronization approach for distributed applications

[EN] Newer social multimedia applications, such as Social TV or networked multi-player games, enable independent groups (or clusters) of users to interact among themselves and share services within the context of simultaneous media content consumption. In such scenarios, concurrently synchronized playout points must be ensured so as not to degrade the user experience on such interaction. We refer to this process as Inter-Destination Multimedia Synchronization (IDMS). This paper presents the design, implementation and evaluation of an evolved version of an RTCP-based IDMS approach, including an Adaptive Media Playout (AMP) scheme that aims to dynamically and smoothly adjust the playout timing of each one of the geographically distributed consumers in a specific cluster if an allowable asynchrony threshold between their playout states is exceeded. For that purpose, we previously had also to develop a full implementation of RTP/RTCP protocols for NS-2, in which we included the IDMS approach as an optional functionality. Simulation results prove the feasibility of such IDMS and AMP proposals, by adopting several dynamic master reference selection policies, to maintain an overall synchronization status (within allowable limits) in each cluster of participants, while minimizing the occurrence of long-term playout discontinuities (such as skips/pauses) which are subjectively more annoying and less tolerable to users than small variations in the media playout rate.This work has been financed, partially, by Universitat Politecnica de Valencia (UPV), under its R&D Support Program in PAID-05-11-002-331 Project and in PAID-01-10. Authors also would like to thank the anonymous reviewers that helped to significantly improve the quality of the paper with their constructive comments.Montagud, M.; Boronat, F. (2012). Enhanced adaptive RTCP-based inter-destination multimedia synchronization approach for distributed applications. Computer Networks. 56(12):2912-2933. https://doi.org/10.1016/j.comnet.2012.05.00329122933561

Media Transport and Use of RTP in WebRTC

The framework for Web Real-Time Communication (WebRTC) provides support for direct interactive rich communication using audio, video, text, collaboration, games, etc. between two peers' web browsers. This memo describes the media transport aspects of the WebRTC framework. It specifies how the Real-time Transport Protocol (RTP) is used in the WebRTC context and gives requirements for which RTP features, profiles, and extensions need to be supported

On the use of adaptive media playout for inter-destination synchronization

[EN] Inter-Destination Multimedia Synchronization (IDMS) is essential in most of the emerging social multimedia applications. In this paper we present a novel Adaptive Media Playout (AMP) scheme that aims to acquire an overall synchronization status between distributed receivers by means of smoothly adjusting their playout timing. Simulation results show that the proposed solution minimizes long-term playout discontinuities (skips and/or pauses) which are subjectively more annoying to users than small variations in the media playout rate.This work was supported by the Universitat Politecnica de Valencia (UPV), under its R&D Support Program in the PAID-01-10 Project, and by Generalitat Valenciana, under its R&D Support Program in the GV 2010/009 Project.Montagud, M.; Boronat, F. (2011). On the use of adaptive media playout for inter-destination synchronization. IEEE Communications Letters. 15(9):863-865. https://doi.org/10.1109/LCOMM.2011.061611.110072S86386515

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

Multicast Mobility in Mobile IP Version 6 (MIPv6) : Problem Statement and Brief Survey

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