Reducing Channel-Change Times with the Real-Time Transport Protocol

Begen, Ali C.; Glazebrook, Neil; Steeg, W. Ver

doi:10.1109/mic.2009.67

Cited by 14 publications

(13 citation statements)

References 2 publications

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“…• Retransmission of lost RTP data packets (RFC 4588), driven by negative acknowledgments sent in early RTCP reports • Rapid acquisition of multicast sessions [7] using RTCP to control rapid unicast delivery of reference information needed to synchronize with an ongoing multicast RTP session • Codec control messages (RFC 5104) to convey requests for full intra-coded frames or select the reference picture, and signal changes in the desired temporal/spatial trade-off and maximum media bit rate We expect IPTV systems to make heavy use of the rapid acquisition and retransmission extensions to support rapid channel change [8].…”

Section: Rtcp Reception Quality Feedbackmentioning

confidence: 99%

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On the use of RTP for monitoring and fault isolation in IPTV

Begen¹,

Perkins

Ott

2010

IEEE Network

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acket-based multimedia communication has come a long way since its early exploration in the 1970s (for audio over Ethernet) and the 1980s (for audiovisual conferencing over IP), and is nowadays a commodity for millions of users. The most prominent example for the success of IP-based multimedia is IP telephony, where commercial development started in the 1990s, and which has led to the conversion -although operators usually prefer to talk of convergence -of many time-division multiplexing (TDM)-based telecommunication networks into IP networks, from the core to the enterprise and increasingly to the home. With increasing availability of broadband access, the entertainment sector has also started to embrace the idea of running everything over IP, leading to the notion of triple play (Internet access, telephony, and IP television [IPTV]) offered in an integrated fashion by Internet service providers (ISPs). Non-ISP-based offerings have followed the trend with a multitude of web-based multimedia streaming offerings including Internet radio stations, TV broadcasters delivering selected programming, and user-generated content platforms such as YouTube.While the types of services, and the control protocols used to access and interact with them, may differ, the common theme across all services is the delivery of real-time (multi)media contents over a packet-switched network. The Real-Time Transport Protocol (RTP), Internet Engineering Task Force (IETF) Request for Comments (RFC) 3550, 1 was developed to serve this purpose, drawing on the experience of the early experimentation. While RTP's original target was multiparty multimedia conferencing in multicast networks, its design was carefully crafted for much broader applicability -RTP accommodates point-to-point IP telephony on one end of the scale, and broadcasting to millions of users on the other. The broad acceptance and deployment of IP telephony solutions using RTP prove the former. In this article we make the case for the latter.After an overview of the IPTV landscape, we review the basics of RTP, introduce the technical concepts of RTP specifically applied to IPTV, and present a new architecture for robust IPTV distribution. We then apply the reporting mechanisms offered by RTP to realize monitoring and fault tolerance, and subsequently show how service providers can use these mechanisms for troubleshooting and increasing customer satisfaction. We conclude with a brief summary and point to future developments. Our contribution is in outlining a multicast RTP-based IPTV distribution architecture that integrates scalable RTP Control Protocol (RTCP) feedback, feedback aggregation and fault isolation, rapid acquisition, and media repair to enhance subscriber quality of experience (QoE). IPTV in the Real WorldIPTV is the delivery of entertainment-quality video over an IP network. However, it is not just about how video content is carried in the network and delivered to consumers. The vision behind IPTV encompasses more than this: it extends the reachability of content...

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Section: Rtcp Reception Quality Feedbackmentioning

confidence: 99%

“…In either case, the FT system buffers RTP media packets that are later used to repair missing packets. In case of a channel change, not just a single packet is sent, but an RTP burst is generated and pushed to the receiver faster than real time to prime the receiver and allow it to begin playback virtually immediately [7,8].…”

Section: Use Of Rtp For Iptvmentioning

confidence: 99%

On the use of RTP for monitoring and fault isolation in IPTV

Begen¹,

Perkins

Ott

2010

IEEE Network

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“…Additional info on typical IPTV architectures can be found in [5,14], and additional info on FTTN networks can be found at [2]. In this paper, we use a simplified model of the FTTN architecture as shown in Fig.…”

Section: An Iptv Architecturementioning

confidence: 99%

Zebroid: using IPTV data to support STB-assisted VoD content delivery

et al. 2010

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IPTV, unlike Internet TV, delivers digital TV and multimedia services over IP-based networks with the required level of quality of service (QoS) and quality of experience (QoE). Linear programming channels in IPTV are delivered through multicast, which is highly scalable with the number of subscribers. Video-on-demand (VoD) content, on the other hand, is typically delivered using unicast, which places a heavy load on the VoD servers and all the network components leading to the end-user set-top boxes (STBs) as the demand increases. With the rapid growth of IPTV subscribers and the shift in video viewing habits, the need to efficiently disseminate large volumes of VoD content has prompted IPTV service providers to consider the use of STBs to assist in video content delivery. This paper describes our current research work on Zebroid, a potential VoD solution for fiber-to-the-node (FTTN) networks, which uses IPTV data on a recurring basis to determine how to select, stripe, and preposition popular content in selected STBs during idle hours. A STB requesting VoD content during the peak hours can then receive necessary stripes from participating STBs in the neighborhood. Recent VoD request access patterns, STB availability data, and capacity data on network components are taken into consideration in determining the parameters used in the striping algorithm of Zebroid. We show both by simulation and emulation on a realistic IPTV testbed that the VoD server load can be reduced by more than 70% during peak hours by allocating only 8 GB of storage on each STB. The savings achieved through Zebroid would also allow IPTV service providers to add more linear programming channels without expensive infrastructure upgrades.

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“…Therefore, the optimization of such components, in conjunction with the provisioning of additional techniques, such as predictive tuning methods (by pre-joining channels), secondary tune-in streams, as well as allocation and assignment of auxiliary servers, is essential to contribute to the decrease of the channel change delays. Up to date, several solutions (e.g., [Fuc08], [Beg09], [Bgc10] and [Ram11]), even IETF standards (e.g., RFC 6285 [Ste11] and RFC 6659 [Beg12]), have devised specific solutions to help decreasing channel change delays when using RTP/RTCP protocols.…”

Section: Reduction Of Channel Change Delaysmentioning

confidence: 99%

“…Therefore, in conjunction with the above techniques, the RTCP timing rules from RFC 6051 should be employed to enable rapid (inter-stream) synchronization. As an example, the works in [Beg09] and in [Bgc10] made use of a rapid acquisition technique (by employing an auxiliary retransmission server), combined with Early RTCP Feedback reporting rules, to decrease channel change delays when joining on-going RTP multicast sessions.…”

Section: Reduction Of Channel Change Delaysmentioning

confidence: 99%

Design, Development and Evaluation of an Adaptive and Standardized RTP/RTCP-Based Idms Solution

Climent¹

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ii Design, Development and Evaluation of an Adaptive and Standardized RTP/RTCP-based IDMS Solutioniii ABSTRACT Nowadays, we are witnessing a transition from physical togetherness towards networked togetherness around media content. Novel forms of shared media experiences are gaining momentum, allowing geographically distributed users to concurrently consume the same media content while socially interacting (e.g., via text, audio or video chat). Relevant use cases are, for example, Social TV, networked games and multi-party conferencing.However, realizing enjoyable shared media services faces many challenges. In particular, a key technological enabler is the concurrent synchronization of the media playout across multiple locations, which is known as Inter-Destination Multimedia Synchronization (IDMS).This PhD thesis presents an inter-operable, adaptive and accurate IDMS solution, based on extending the capabilities of RTP/RTCP standard protocols (RFC 3550). Concretely, two new RTCP messages for IDMS have been defined to carry out the necessary information to achieve IDMS. Such RTCP extensions have been standardized within the IETF, in RFC 7272. In addition, novel standard-compliant Early Event-Driven (EED) RTCP feedback reporting mechanisms have been also designed to enhance the performance in terms of interactivity, flexibility, dynamism and accuracy when performing IDMS.The designed IDMS solution makes use of globally synchronized clocks (e.g., using NTP) and can adopt different (centralized and distributed) architectural schemes to exchange the RTCP messages for IDMS. This allows efficiently providing IDMS in a variety of networked scenarios and applications, with different requirements (e.g., interactivity, scalability, robustness…) and available resources (e.g., bandwidth, latency, multicast support…). Likewise, various monitoring and control algorithms, such as dynamic strategies for selecting the reference timing to synchronize with, and fault tolerance mechanisms, have been added. Moreover, the proposed IDMS solution includes a novel Adaptive Media Playout (AMP) technique, which aims to smoothly adjust the media playout rate, within perceptually tolerable ranges, every time allowable asynchrony thresholds are exceeded.Prototypes of the IDMS solution have been implemented in both a simulation and in real media framework. The evaluation tests prove the consistent behavior and Mario Montagud Climent iv the satisfactory performance of each one of the designed components (e.g., protocols, architectural schemes, master selection policies, adjustment techniques…). Likewise, comparison results between the different developed alternatives for such components are also provided. In general, the obtained results demonstrate the ability of this RTP/RTCP-based IDMS solution to concurrently and independently maintain an overall synchronization status (within allowable limits) in different logical groups of users, while avoiding annoying playout discontinuities and hardly increasing the computational and traffic load. De...

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Reducing Channel-Change Times with the Real-Time Transport Protocol

Cited by 14 publications

References 2 publications

On the use of RTP for monitoring and fault isolation in IPTV

On the use of RTP for monitoring and fault isolation in IPTV

Zebroid: using IPTV data to support STB-assisted VoD content delivery

Design, Development and Evaluation of an Adaptive and Standardized RTP/RTCP-Based Idms Solution

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