In this paper we describe a set of video subjective tests with 498 student viewers in the age range 9-17. These tests are based on earlier tests that we carried out with adult viewers. The aim was not to validate our earlier results, but to use them as a reference against which we could compare the scores of the students, to allow us to address the questions of how sensitive students in this age range are to video quality issues, and how suitable they would be as viewers in subjective tests. We used the same test content as in our earlier tests, including content at high and ultra-high resolution, and with standard and high dynamic range.
Adaptive bit rate streaming using HTTP delivery is now widely used to provide live audio-visual content services. However, the quality of experience is often considered inferior to that of conventional broadcast services due to the high end to end latency, which is mostly due to the large amount of buffering required to provide resilience to variable network conditions. In this paper we present our initial research to address this issue of high latency. If each segment of content could be delivered in a consistent period of time, the amount of buffering required could be reduced. We present the results of simulations where we have changed the timing of the TCP congestion response to consider the timing requirements of content segments, while trying to retain fairness to competing flows. We show that with this simple modification to TCP the variation in the delivery time of content segments can be reduced and much lower end to end latency achieved. With no changes to the network or client devices required, this solution may be straightforward to deploy to make the user experience of streaming services much closer to that of conventional broadcast services.
This paper presents the ITU-T Study Group 15 development of H-series Recommendations that allow interworking between different audiovisual communication terminals manufactured by different equipment providers. The paper focuses on H.310 and H.321 systems for broad-band ATM environments and H.322 and H.323 systems for LAN environments where the quality of service may or may not be guaranteed. The paper first lists the Recommendations developed by the ITU-T for audiovisual communication systems and the network environments in which they may be used. It then describes the design philosophy, the network specific characteristics, and hardware trials for each system. Then it describes the communication control protocol defined in H.245 which is used commonly by different audiovisual communication systems. Finally, the paper discusses interworking scenarios for communication between the different types of terminal on different networks.
Increasingly, major live events are being delivered using HTTP Adaptive Streaming (HAS), which, being a unicast technology, causes such events to generate huge traffic demand peaks on broadband networks which in turn drive significant investment in the network to increase capacity. Multicast ABR (mABR) is a streaming technology that inserts multicast into the path of a unicast HAS stream, allowing the traffic peaks to be reduced while requiring little or no change to the client players. But the unmodified clients continue to behave as HAS clients, asynchronously requesting content segments, adapting the quality of content requested according to perceived network conditions, totally unaware of the use of multicast to deliver synchronously one or more of the available encoded representations of the content to a local proxy. This creates inefficiencies and challenges in system design. Segments delivered synchronously must be cached locally ready for asynchronous requests from the client. Adaptation by the client will cause unwanted segments to be received by multicast, and alternative representations to be requested by unicast. We exploit CDN log data for BT Sport channels delivered by HAS to explore these inefficiencies and challenges. We show that unicast traffic in the core network could be reduced by more than 75% by delivering a single HAS representation by multicast to each client proxy, with larger savings for HD than UHD streams. We show that a cache at the client proxy capable of storing seven segments of 6s duration would be sufficient to be able to satisfy the majority of asynchronous requests from clients. Finally, we evaluate the performance of four policies for a client proxy to join and leave multicast groups in terms of the competing metrics of savings in unicast traffic in the core network and the quantity of data delivered over the access network.
We describe an equitable quality video streaming system where the video server dynamically selects between multiple versions of video content coded at different fixed quality levels, and dynamically selects transmission rates for each video session so that the network bandwidth is divided between concurrent users such that they receive equal video quality at each moment in time. By sharing the bandwidth in this way and allowing delivery to get arbitrarily ahead of decoding, equitable quality video streaming can significantly outperform constant bit rate coding and delivery, allowing 100% more video sessions to be delivered at the same overall perceptual quality.
The results of a comparison of the compression efficiency of the H.263 and H.26L algorithms are presented when used to compress video from a variety of cameras. It is shown that H.26L achieves better compression performance in most cases, with larger compression gains being achieved when higher quality cameras are used.
HTTP Adaptive Streaming (HAS) technologies such as MPEG DASH are now used extensively to deliver television services to large numbers of viewers. In HAS, the client requests segments of content using HTTP, with an ABR algorithm selecting the quality at which to request each segment to trade-off video quality with the avoidance of stalling. This introduces significant end to end latency compared to traditional broadcast, due to the the client requiring a large enough buffer for the ABR algorithm to react to changes in network conditions in a timely manner. The recently standardised Common Media Application Format (CMAF) has helped address the issue of latency by defining segments as composed of independently transferable chunks. In this paper, we describe a simulation model we have developed to evaluate the performance of four popular ABR algorithms using DASH and CMAF in various low latency live streaming scenarios. Realistic network conditions are used for the evaluation, which are based on throughput data taken from the CDN logs of a commercial live TV service. We quantify the performance of the ABR algorithms using a selection of QoE metrics, and show that CMAF can significantly improve ABR performance in low delay scenarios.
For expanded coverage of this month’s topic “Evolving Distribution,” you can find the following paper in the Digital Edition. Visit the SMPTE digital library at http://journal.smpte.org to access the issue and to read this additional paper.
This paper describes the architecture and implementation of a trial multimedia multicast distribution mechanism for IP networks using layered audio and video compression that has been developed at BT Labs. The system consists of an array of servers capable of multicasting audiovisual content to large numbers of users who can view the content on their PC using a software client application. This could be used to access content including live television transmissions, business television, chairman's talks, and stored training material, etc.
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