Optimal packetisation of MPEG-4 using RTP over mobile networks

The introduction of third-generation wireless networks should result in real-time mobile video communications becoming a reality. Delivery of such video is likely to be facilitated by the realtime transport protocol (RTP). Careful packetisation of the video data is necessary to ensure the optimal trade-off between channel utilisation and error robustness. Theoretical analyses for two basic schemes of MPEG-4 data encapsulation within RTP packets are presented. Simulations over a GPRS (general packet radio service) network are used to validate the analysis of the most efficient scheme. Finally, a motion adaptive system for deriving MPEG-4 video packet sizes is presented. Further simulations demonstrate the benefits of the adaptive system

Modeling of Packet Streaming Services in Information Communication Networks

Application of the term video streaming in contemporary usage denotes compression techniques and data buffering, which can transmit video in real time over the network. There is currently a rapid growth and development of technologies using wireless broadband technology as a transport, which is a seri- ous alternative to cellular communication systems. Adverse effect of the aggressive environment used in wireless networks transmission results in data packets undergoing serious distortions and often get- ting lost in transit. All existing research in this area investigate the known types of errors separately. At present there are no standard approaches to determining the effect of errors on transmission quality of services. Besides, the spate in popularity of multimedia applications has led to the need for optimization of bandwidth allocation and usage in telecommunication networks. Modern telecommunication networks should by their definition be able to maintain the quality of different applications with different Quality of Service (QoS) levels. QoS requirements are generally dependent on the parameters of network and application layers of the OSI model. At the application layer QoS depends on factors such as resolution, bit rate, frame rate, video type, audio codecs, and so on. At the network layer, distortions (such as delay, jitter, packet loss, etc.) are introduced

Performance of TCP/UDP under Ad Hoc IEEE802.11

TCP is the De facto standard for connection oriented transport layer protocol, while UDP is the De facto standard for transport layer protocol, which is used with real time traffic for audio and video. Although there have been many attempts to measure and analyze the performance of the TCP protocol in wireless networks, very few research was done on the UDP or the interaction between TCP and UDP traffic over the wireless link. In this paper, we tudy the performance of TCP and UDP over IEEE802.11 ad hoc network. We used two topologies, a string and a mesh topology. Our work indicates that IEEE802.11 as a ad-hoc network is not very suitable for bulk transfer using TCP. It also indicates that it is much better for real-time audio. Although one has to be careful here since real-time audio does require much less bandwidth than the wireless link bandwidth. Careful and detailed studies are needed to further clarify that issue.Comment: 9 pages, 5 figures, ICT 2003 (10th International Conference on Telecommunication

Evaluation of the MDC and FEC over the quality of service and quality of experience for video distribution in ad hoc networks

Mobile ad hoc networks (MANETs) offer an excellent scenario for deploying communication applications because of the connectivity and versatility of this kind of networks. In contrast, the topology is usually extremely dynamic causing high rate of packet loss, so that ensuring a specific Quality of Service (QoS) for real-time video services becomes a hard challenge. In this paper, we evaluate the effect of using Multiple Description Coding (MDC) and Forward Error Correction (FEC) techniques for improving video quality in a multimedia content distribution system. A hybrid architecture using fixed and wireless ad hoc networks is proposed, which enables the use of multipoint-to-point transmission. MDC and FEC mechanisms can be combined with multipath transmission to increase the network efficiency and recover lost packets, improving the overall Quality of Experience (QoE) of the receiver. Simulations have been analyzed paying attention to objective parameters (Peak Signal to Noise Ratio, Packet Delivery Ratio, Decodable Frame Rate and interruptions) and subjective parameters. Results show that MDC increases the probability of packet delivery and FEC is able to recover lost frames and reduce video interruptions in moderate mobility scenarios, resulting in the improvement of video quality and the final user experience.This work was supported by project MIQUEL (TEC2007- 68119-C02-01/TCM) of the Spanish Ministry of Education and Science. The authors would like to thank the Editor and the reviewers for helpful suggestions to improve the quality of this paper.Acelas Delgado, P.; Arce Vila, P.; Guerri Cebollada, JC.; Castellanos Hernández, WE. (2014). Evaluation of the MDC and FEC over the quality of service and quality of experience for video distribution in ad hoc networks. Multimedia Tools and Applications. 68(3):969-989. https://doi.org/10.1007/s11042-012-1111-3969989683Apostolopoulos JG, Wong T, Tan W, Wee SJ (2002) On multiple description streaming with content delivery networks. IEEE INFOCOMBoukerche A (2009) Algorithms and protocols for wireless and mobile ad hoc networks. John Wiley & Sons IncChow CO, Ishii H (2007) Enhancing real-time video streaming over mobile ad hoc networks using multipoint-to-point communication. Comput Commun 30:1754–1764Clausen T, Jacquet P (2003) Optimized link state routing protocol (OLSR), RFC 3626Corrie B et al (2003) Towards quality of experience in advanced collaborative environments. Third Annual Workshop on Advanced Collaborative EnvironmentsGabrielyan E, Hersch R (2006) Reliable multi-path routing schemes for real-time streaming. International Conference on Digital Telecommunications, pp 65–65Gandikota VR, Tamma BR, Murthy CSR (2008) Adaptive-FEC based packet loss resilience scheme for supporting voice communication over adhoc wireless networks. IEEE Trans Mobile Comput 7:1184–1199Gharavi H (2008) Multi-channel for multihop communication links. International Conference on Telecommunications, pp 1–6Grega M, Janowski L, Leszczuk M, Romaniak P, Papir Z (2008) Quality of experience evaluation for multimedia services. Przegląd Telekomunikacyjny i Wiadomości Telekomunikacyjne 4:142–153Hsieh MY, Huang YM, Chian TC (2007) Transmission of layered video streaming via multi-path on ad hoc networks. Multimed Tool Appl 34:155–177ITU—International Telecommunication Union (2007) Definition of quality of experience (QoE)”, Reference: TD 109rev2 (PLEN/12)ITU-R Recommendation BT.500-12 (2009) Methodology for the subjective assessment of the quality of television pictures. International Telecommunication Union, GenevaITU-T Recommendation P.910 (2000) Subjective video quality assessment methods for multimedia applications. International Telecommunication Union, GenevaKao KL, Ke ChH, Shieh CH (2006) An advanced simulation tool-set for video transmission performance evaluation. IEEE Region 10 Conference, pp 1–40Ke CH et al (2006) A novel realistic simulation tool for video transmission over wireless network. Proceedings of the IEEE International Conference on Sensor Networks, Ubiquitous, and Trsutworthy ComputingKeisuke U, Cheeonn C, Hiroshi I (2008) A study on video performance of multipoint-to-point video streaming with multiple description coding over ad hoc networks. EEJ Trans Electron, Inf Syst 128:1431–1437Kilkki K (2008) Quality of experience in communications ecosystem. J Univers Comput Sci 14:615–624Li A (2007) RTP payload format for generic forward error correction. RFC 5109, Dec. 2007Li J, Blake C, Couto DD, Lee H, Morris R (2001) Capacity of ad hoc wireless networks. 7th Annual International Conference on Mobile Computing and Networking, pp 16–21Liao Y, Gibson JD (2011) Routing-aware multiple description video coding over mobile ad-hoc networks. IEEE Trans Multimed 13:132–142Lindeberg M, Kristiansen S, Plagemann T, Goebel V (2011) Challenges and techniques for video streaming over mobile ad hoc networks. Multimed Syst 17:51–82Mao S et al (2003) Video transport over ad hoc networks: multistream coding with multipath transport. IEEE J Sel Area Comm 21:1721–1737Ni P (2009) Towards Optimal Quality of Experience Via Scalable Video Coding. Mälardalen University Press Licentiate Theses, SwedenPinson MH, Wolf S (2004) A new standardized method for objectively measuring video quality. IEEE Trans Broadcast 50:312–322Rong B, Qian Y, Lu K, Hu RQ, Kadoch M (2010) Multipath routing over wireless mesh networks for multiple description video transmission. IEEE J Sel Area Comm 28:321–331Schierl T, Ganger K, Hellge C, Wiegand T, Stockhammer T (2006) SVC-based multisource streaming for robust video trans- mission in mobile ad hoc networks. IEEE Wireless Comm 13:96–103Schierl T, Stockhammer T, Wiegand T (2007) Mobile video transmission using scalable video coding. IEEE Trans Circ Syst Video Tech 17:1204–1217Schwarz H, Marpe D, Wiegand T (2007) Overview of the scalable video coding extension of the H.264/AVC standard. IEEE Trans Circ Syst Video Tech 17:1103–1120VQEG (2008) Video quality experts group. Available online: http://www.vqeg.orgWang Z et al (2004) Image quality assessment: from error visibility to structural similarity. IEEE Trans Image Process 13:600–612Wei W, Zakhor A (2004) Robust multipath source routing protocol (RMPSR) for video communication over wireless ad hoc net- works. 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MSPlayer: Multi-Source and multi-Path LeverAged YoutubER

Online video streaming through mobile devices has become extremely popular nowadays. YouTube, for example, reported that the percentage of its traffic streaming to mobile devices has soared from 6% to more than 40% over the past two years. Moreover, people are constantly seeking to stream high quality video for better experience while often suffering from limited bandwidth. Thanks to the rapid deployment of content delivery networks (CDNs), popular videos are now replicated at different sites, and users can stream videos from close-by locations with low latencies. As mobile devices nowadays are equipped with multiple wireless interfaces (e.g., WiFi and 3G/4G), aggregating bandwidth for high definition video streaming has become possible. We propose a client-based video streaming solution, MSPlayer, that takes advantage of multiple video sources as well as multiple network paths through different interfaces. MSPlayer reduces start-up latency and provides high quality video streaming and robust data transport in mobile scenarios. We experimentally demonstrate our solution on a testbed and through the YouTube video service.Comment: accepted to ACM CoNEXT'1

Streaming Video Performance and Enhancements in Resource-Constrained Wireless Networks

Streaming video is an increasingly popular application in wireless networks. The concept of a live streaming video yields several enticing possibilities: real-time video conferencing, television broadcasting, pay-per-view movie streaming, and more. These ideas have already been explored via the internet and have met with mixed success, largely due to the shortcomings of the underlying network. Taking streaming video to wireless networks, then, poses several significant challenges. Wireless networks are inherently more susceptible to failures and data corruption due to their unstable communications medium. This volatility suggests serious drawbacks for any implementation of streaming video. Video frame errors, jitter, and even complete sync loss are entirely conceivable in a wireless environment. Many of these issues have been undertaken and several approaches to mediation or even solution of these problems are underway. This thesis proposes to use advanced simulation techniques to properly exhaustively permute many vital parameters within a UMTS network and uncover, if they exist, bottlenecks in UMTS performance under considerable network load. This is accomplished via a described testing plan with simulation environment. Additionally this thesis proposes a new UDP-like transport layer specially optimized for streaming media over resource-constrained networks, tested to work with significant improvements under the UMTS cellular networking system. Finally this thesis provides several innovative new methods in the furtherance of the field of streaming media research in resourceconstrained and cellular environments. Overall this thesis makes several important contributes to an exciting and ever-growing field of active research and discussion

QoE-Based Low-Delay Live Streaming Using Throughput Predictions

Recently, HTTP-based adaptive streaming has become the de facto standard for video streaming over the Internet. It allows clients to dynamically adapt media characteristics to network conditions in order to ensure a high quality of experience, that is, minimize playback interruptions, while maximizing video quality at a reasonable level of quality changes. In the case of live streaming, this task becomes particularly challenging due to the latency constraints. The challenge further increases if a client uses a wireless network, where the throughput is subject to considerable fluctuations. Consequently, live streams often exhibit latencies of up to 30 seconds. In the present work, we introduce an adaptation algorithm for HTTP-based live streaming called LOLYPOP (Low-Latency Prediction-Based Adaptation) that is designed to operate with a transport latency of few seconds. To reach this goal, LOLYPOP leverages TCP throughput predictions on multiple time scales, from 1 to 10 seconds, along with an estimate of the prediction error distribution. In addition to satisfying the latency constraint, the algorithm heuristically maximizes the quality of experience by maximizing the average video quality as a function of the number of skipped segments and quality transitions. In order to select an efficient prediction method, we studied the performance of several time series prediction methods in IEEE 802.11 wireless access networks. We evaluated LOLYPOP under a large set of experimental conditions limiting the transport latency to 3 seconds, against a state-of-the-art adaptation algorithm from the literature, called FESTIVE. We observed that the average video quality is by up to a factor of 3 higher than with FESTIVE. We also observed that LOLYPOP is able to reach a broader region in the quality of experience space, and thus it is better adjustable to the user profile or service provider requirements.Comment: Technical Report TKN-16-001, Telecommunication Networks Group, Technische Universitaet Berlin. This TR updated TR TKN-15-00