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

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

Low-complexity video coding for receiver-driven layered multicast

In recent years, the “Internet Multicast Backbone,” or MBone, has risen from a small, research curiosity to a large- scale and widely used communications infrastructure. A driving force behind this growth was the development of multipoint audio, video, and shared whiteboard conferencing applications. Because these real-time media are transmitted at a uniform rate to all of the receivers in the network, a source must either run at the bottleneck rate or overload portions of its multicast distribution tree. We overcome this limitation by moving the burden of rate adaptation from the source to the receivers with a scheme we call receiver-driven layered multicast, or RLM. In RLM, a source distributes a hierarchical signal by striping the different layers across multiple multicast groups, and receivers adjust their reception rate by simply joining and leaving multicast groups. In this paper, we describe a layered video compression algorithm which, when combined with RLM, provides a comprehensive solution for scalable multicast video transmission in heterogeneous networks. In addition to a layered representation, our coder has low complexity (admitting an effi- cient software implementation) and high loss resilience (admitting robust operation in loosely controlled environments like the Inter- net). Even with these constraints, our hybrid DCT/wavelet-based coder exhibits good compression performance. It outperforms all publicly available Internet video codecs while maintaining comparable run-time performance. We have implemented our coder in a “real” application—the UCB/LBL videoconferencing tool vic. Unlike previous work on layered video compression and transmission, we have built a fully operational system that is currently being deployed on a very large scale over the MBone

Semi-synchronous video for deaf telephony with an adapted synchronous codec

Magister Scientiae - MScCommunication tools such as text-based instant messaging, voice and video relay services, real-time video chat and mobile SMS and MMS have successfully been used among Deaf people. Several years of field research with a local Deaf community revealed that disadvantaged South African Deaf people preferred to communicate with both Deaf and hearing peers in South African Sign Language as opposed to text. Synchronous video chat and video relay services provided such opportunities. Both types of services are commonly available in developed regions, but not in developing countries like South Africa. This thesis reports on a workaround approach to design and develop an asynchronous video communication tool that adapted synchronous video codecs to store-and-forward video delivery. This novel asynchronous video tool provided high quality South African Sign Language video chat at the expense of some additional latency. Synchronous video codec adaptation consisted of comparing codecs, and choosing one to optimise in order to minimise latency and preserve video quality. Traditional quality of service metrics only addressed real-time video quality and related services. There was no such standard for asynchronous video communication. Therefore, we also enhanced traditional objective video quality metrics with subjective assessment metrics conducted with the local Deaf community.South Afric

Robust P2P Live Streaming

Projecte fet en col.laboració amb la Fundació i2CATThe provisioning of robust real-time communication services (voice, video, etc.) or media contents through the Internet in a distributed manner is an important challenge, which will strongly influence in current and future Internet evolution. Aware of this, we are developing a project named Trilogy leaded by the i2CAT Foundation, which has as main pillar the study, development and evaluation of Peer-to-Peer (P2P) Live streaming architectures for the distribution of high-quality media contents. In this context, this work concretely covers media coding aspects and proposes the use of Multiple Description Coding (MDC) as a flexible solution for providing robust and scalable live streaming over P2P networks. This work describes current state of the art in media coding techniques and P2P streaming architectures, presents the implemented prototype as well as its simulation and validation results

AdamRTP: Adaptive multi-flow real-time multimedia transport protocol for Wireless Sensor Networks

Real-time multimedia applications are time sensitive and require extra resources from the network, e.g. large bandwidth and big memory. However, Wireless Sensor Networks (WSNs) suffer from limited resources such as computational, storage, and bandwidth capabilities. Therefore, sending real-time multimedia applications over WSNs can be very challenging. For this reason, we propose an Adaptive Multi-flow Real-time Multimedia Transport Protocol (AdamRTP) that has the ability to ease the process of transmitting real-time multimedia over WSNs by splitting the multimedia source stream into smaller independent flows using an MDC-aware encoder, then sending each flow to the destination using joint/disjoint path. AdamRTP uses dynamic adaptation techniques, e.g. number of flows and rate adaptation. Simulations experiments demonstrate that AdamRTP enhances the Quality of Service (QoS) of transmission. Also, we showed that in an ideal WSN, using multi-flows consumes less power than using a single flow and extends the life-time of the network

Video Multicasting Over 3g/umts Networks

Tez (Yüksek Lisans) -- İstanbul Teknik Üniversitesi, Fen Bilimleri Enstitüsü, 2009Thesis (M.Sc.) -- İstanbul Technical University, Institute of Science and Technology, 2009Bu çalışmada, UMTS şebekelerinde kullanılan farklı çeşitteki çoğa gönderim teknolojileri olumlu ve olumsuz yanları ile birlikte açıklanmıştır. 3GPP standardı, çoklu dağıtımı desteklemek için MBMS (Multimedia Broadcast/Multicast Services – Çoğul Ortam Yayın ve Çoklu Dağıtım Servisi) ile geliştirilmiştir. Bu tez esas olarak MBMS standardı, UMTS şebekelerine uygulanabilen video aktarım protokolleri ve teknikleri ile en önemli çoklu dağıtım servisi olarak görülen mobil televizyon uygulamasına odaklanmıştır. Teknolojik yeniliklerin başarısı ve kullanıcılar tarafından kabulü önemli ölçüde içeriğe dayalıdır. İçerik, kullanıcıların isteklerine göre tasarlanmalıdır ve mobil TV için önemli bir rol oynar. Bu tezde, kullanıcı istekleri ortaya konulmuş ve ayrıca mobil TV teknolojilerinin mevcut durumu, deneme sonuçları ve ticari olarak piyasaya sürülmesi tanımlanmıştır. Hangi durumlarda, nerelerde ve ne zaman bu hizmetlerin kullanılabileceğini tanımlamak için gerçekleştirilen araştırma sonuçları ortaya konulmuştur. Tezin bu konuya katkısı yeniden belirtilip, gelecekteki araştırmalara yön verecek bazı konulardan bahsedilmiştir.In this study different types of multicast technologies which are used in UMTS networks are introduced with their pros and cons. The 3GPP standard has been enhanced with MBMS (Multimedia Broadcast/Multicast Services) to support multicasting. This thesis mainly focuses on MBMS standard, video streaming protocols and techniques that are applicable to UMTS networks and especially the mobile TV service. Success and user acceptance of new technology innovations are highly depend on the content. It needs to be designed according to consumers’ demands and play an important role for mobile TV. In this thesis, users’ demands are introduced, also mobile TV technologies current status, trial results and commercial launches are described. Research results are presented which are performed to define in what situations, where and when the service can be used. The contributions are restated and some insight into future research directions is given.Yüksek LisansM.Sc

Real-time video streaming using peer-to-peer for video distribution

The growth of the Internet has led to research and development of several new and useful applications including video streaming. Commercial experiments are underway to determine the feasibility of multimedia broadcasting using packet based data networks alongside traditional over-the-air broadcasting. Broadcasting companies are offering low cost or free versions of video content online to both guage and at the same time generate popularity. In addition to television broadcasting, video streaming is used in a number of application areas including video conferencing, telecommuting and long distance education. Large scale video streaming has not become as widespread or widely deployed as could be expected. The reason for this is the high bandwidth requirement (and thus high cost) associated with video data. Provision of a constant stream of video data on a medium to large scale typically consumes a significant amount of bandwidth. An effect of this is that encoding bit rates are lowered and consequently video quality is degraded resulting in even slower uptake rates for video streaming services. The aim of this dissertation is to investigate peer-to-peer streaming as a potential solution to this bandwidth problem. The proposed peer-to-peer based solution relies on end user co-operation for video data distribution. This approach is highly effective in reducing the outgoing bandwidth requirement for the video streaming server. End users redistribute received video chunks amongst their respective peers and in so doing increase the potential capacity of the entire network for supporting more clients. A secondary effect of such a system is that encoding capabilities (including higher encoding bit rates or encoding of additional sub-channels) can be enhanced. Peer-to-peer distribution enables any regular user to stream video to large streaming networks with many viewers. This research includes a detailed overview of the fields of video streaming and peer-to-peer networking. Techniques for optimal video preparation and data distribution were investigated. A variety of academic and commercial peer-to-peer based multimedia broadcasting systems were analysed as a means to further define and place the proposed implementation in context with respect to other peercasting implementations. A proof-of-concept of the proposed implementation was developed, mathematically analyzed and simulated in a typical deployment scenario. Analysis was carried out to predict simulation performance and as a form of design evaluation and verification. The analysis highlighted some critical areas which resulted in adaptations to the initial design as well as conditions under which performance can be guaranteed. A simulation of the proof-of-concept system was used to determine the extent of bandwidth savings for the video server. The aim of the simulations was to show that it is possible to encode and deliver video data in real time over a peer-to-peer network. The proposed system achieved expectations and showed significant bandwidth savings for a sustantially large video streaming audience. The implementation was able to encode video in real time and continually stream video packets on time to connected peers while continually supporting network growth by connecting additional peers (or stream viewers). The system performed well and showed good performance under typical real world restrictions on available bandwith capacity.Dissertation (MEng)--University of Pretoria, 2009.Electrical, Electronic and Computer Engineeringunrestricte

Real-time Audio-Visual Media Transport over QUIC

We consider the problem of how to transport low-latency, interactive, real-time traffic over QUIC. This is needed to support applications like WebRTC, but difficult to support due to the reliable, unframed, nature of QUIC streams. We review the needs of low-latency real-time applications and how they have been supported in previous protocols, then propose a minimal set of extensions to QUIC to provide such support. Compared to a raw datagram service, our extensions provide meaningful support for partially reliable and real-time flows, in a backwards compatible manner