Carsten Griwodz

Video streaming using a location-based bandwidth-lookup service for bitrate planning

A lot of people around the world commute using public transportation and would like to spend this time viewing streamed video content such as news or sports updates. However, mobile wireless networks typically suffer from severe bandwidth fluctuations, and the networks are often completely unresponsive for several seconds, sometimes minutes. Today, there are several ways of adapting the video bitrate and thus the video quality to such fluctuations, for example, using scalable video codecs or segmented adaptive HTTP streaming that switches between nonscalable video streams encoded in different bitrates. Still, for a better long-term video playout experience that avoids disruptions and frequent quality changes while using existing video adaptation technology, it is desirable to perform bandwidth prediction and planned quality adaptation. This article describes a video streaming system for receivers equipped with a GPS. A receivers download rate is constantly monitored, and periodically reported back to a central database along with associated GPS positional data. Thus, based on the current location, a streaming device can use a GPS-based bandwidth-lookup service in order to better predict the near-future bandwidth availability and create a schedule for the video playout that takes likely future availability into account. To create a prototype and perform initial tests, we conducted several field trials while commuting using public transportation. We show how our database has been used to predict bandwidth fluctuations and network outages, and how this information helps maintain uninterrupted playback with less compromise on video quality than possible without prediction.

Commute path bandwidth traces from 3G networks: analysis and applications

In this dataset paper, we present and make available real-world measurements of the throughput that was achieved at the application layer when adaptive HTTP streaming was performed over 3G networks using mobile devices. For the streaming sessions, we used popular commute routes in and around Oslo (Norway) traveling with different types of public transportation (metro, tram, train, bus and ferry). We also have a few logs using a car. Each log provides a times-tamp, GPS coordinates and the measured number of bytes downloaded for approximately every second of the route. The dataset can be used in several ways, but the most obvious application is to emulate the same network bandwidth behavior (on specific geographical positions) for repeated experiments.

Reducing Internet Latency: A Survey of Techniques and Their Merits

Latency is increasingly becoming a performance bottleneck for Internet Protocol (IP) networks, but historically, networks have been designed with aims of maximizing throughput and utilization. This paper offers a broad survey of techniques aimed at tackling latency in the literature up to August 2014, as well as their merits. A goal of this work is to be able to quantify and compare the merits of the different Internet latency reducing techniques, contrasting their gains in delay reduction versus the pain required to implement and deploy them. We found that classifying techniques according to the sources of delay they alleviate provided the best insight into the following issues: 1) The structural arrangement of a network, such as placement of servers and suboptimal routes, can contribute significantly to latency; 2) each interaction between communicating endpoints adds a Round Trip Time (RTT) to latency, particularly significant for short flows; 3) in addition to base propagation delay, several sources of delay accumulate along transmission paths, today intermittently dominated by queuing delays; 4) it takes time to sense and use available capacity, with overuse inflicting latency on other flows sharing the capacity; and 5) within end systems, delay sources include operating system buffering, head-of-line blocking, and hardware interaction. No single source of delay dominates in all cases, and many of these sources are spasmodic and highly variable. Solutions addressing these sources often both reduce the overall latency and make it more predictable.

Improving the performance of quality-adaptive video streaming over multiple heterogeneous access networks

Devices capable of connecting to multiple, overlapping networks simultaneously are becoming increasingly common. For example, most laptops are equipped with LAN- and WLAN-interfaces, and smart phones can typically connect to both WLANs and 3G mobile networks. At the same time, streaming high-quality video is becoming increasingly popular. However, due to bandwidth limitations or the unreliable and unpredictable nature of some types of networks, streaming video can be subject to frequent periods of rebuffering and characterised by a low picture quality. In this paper, we present a client-side request scheduler that distributes requests for the video over multiple heterogeneous interfaces simultaneously. Each video is divided into independent segments with constant duration, enabling segments to be requested over separate links, utilizing all the available bandwidth. To increase performance even further, the segments are divided into smaller subsegments, and the sizes are dynamically calculated on the fly, based on the throughput of the different links. This is an improvement over our earlier subsegment approach, which divided segments into fixed size subsegments. Both subsegment approaches were evaluated with on-demand streaming and quasi-live streaming. The new subsegment approach reduces the number of playback interruptions and improves video quality significantly for all cases where the earlier approach struggled. Otherwise, they show similar performance.

The fun of using TCP for an MMORPG

Massive multi-player online games have become a popular, fast growing, multi-million industry with a very high user mass supporting hundreds or thousands of concurrent players. In many cases, these games are centralized and every player communicates with the central server through a time-critical unicast event stream. Funcoms Anarchy Online is one of these; it is based on TCP. We find that its kind of traffic has some interesting properties that inspire changes to protocol or architecture. In these game streams, TCP does not back off, using TCP does not have to be slower than using UDP, and almost only repeated timeouts ruin the game experience. Improving the latter in the sender implementation does not impose any remarkable penalty on the network. Alternatively, a proxy architecture for multiplexing could save about 40% resources at the server, allow congestion control to work and also reduce the lag of the game.

A network-layer proxy for bandwidth aggregation and reduction of IP packet reordering

With todays widespread deployment of wireless technologies, it is often the case that a single communication device can select from a variety of access networks. At the same time, there is an ongoing trend towards integration of multiple network interfaces into end-hosts, such as cell phones with HSDPA, Bluetooth and WLAN. By using multiple Internet connections concurrently, network applications can benefit from aggregated bandwidth and increased fault tolerance. However, the heterogeneity of wireless environments introduce challenges with respect to implementation, deployment, and protocol compatibility. Variable link characteristics cause reordering when sending IP packets of the same flow over multiple paths. This paper introduces a multilink proxy that is able to transparently stripe traffic destined for multihomed clients. Operating on the network layer, the proxy uses path monitoring statistics to adapt to changes in throughput and latency. Experimental results obtained from a proof-of-concept implementation verify that our approach is able to fully aggregate the throughput of heterogeneous downlink streams, even if the path characteristics change over time. In addition, our novel method of equalizing delays by buffering packets on the proxy significantly reduces IP packet reordering and the buffer requirements of clients.

Mobile video streaming using location-based network prediction and transparent handover

A well known challenge with mobile video streaming is fluctuating bandwidth. As the client devices move in and out of network coverage areas, the users may experience varying signal strengths, competition for the available resources and periods of network outage. These conditions have a significant effect on video quality. In this paper, we present a video streaming solution for roaming clients that is able to compensate for the effects of oscillating bandwidth through bandwidth prediction and video quality scheduling. We combine our existing adaptive segmented HTTP streaming system with 1) an application layer framework for creating transparent multi-link applications, and 2) a location based QoS information system containing GPS coordinates and accompanying bandwidth measurements, populated through crowd-sourcing. Additionally, we use real-time traffic information to improve the prediction by, for example, estimating the length of a commute route. To evaluate our prototype, we performed real-world experiments using a popular tram route in Oslo, Norway. The client connected to multiple networks, and the results show that our solution increases the perceived video quality significantly. Also, we used simulations to evaluate the potential of aggregating bandwidth along the route.

Bagadus: an integrated system for arena sports analytics: a soccer case study

Sports analytics is a growing area of interest, both from a computer system view to manage the technical challenges and from a sport performance view to aid the development of athletes. In this paper, we present Bagadus, a prototype of a sports analytics application using soccer as a case study. Bagadus integrates a sensor system, a soccer analytics annotations system and a video processing system using a video camera array. A prototype is currently installed at Alfheim Stadium in Norway, and in this paper, we describe how the system can follow and zoom in on particular player(s). Next, the system will playout events from the games using stitched panorama video or camera switching mode and create video summaries based on queries to the sensor system. Furthermore, we evaluate the system from a systems point of view, benchmarking different approaches, algorithms and tradeoffs.

Fine-grained scalable streaming from coarse-grained videos

Scalable video is an attractive option for adapting the bandwidth consumption of streaming video to the available bandwidth. Fine-grained scalability can adapt most closely to the available bandwidth, but this comes at the cost of a high compression penalty. In the context of VoD streaming to mobile end systems, we have therefore explored whether a similar adaptation to the available bandwidth can be achieved by performing layer switching in coarse-grained scalable videos. In this approach, enhancement layers of a video stream are switched on and off to achieve any desired longer-term bandwidth. We performed user studies to evaluate the idea, and came to the far-from-obvious conclusion that layer switching is viable way for bit-rate savings and fine-grained bit-rate adaptation even for rather short times between layer switches.

Low overhead container format for adaptive streaming

Current segmented HTTP streaming systems provide scalable and quality adaptive video delivery services to a huge number of users. However, while they support a wide range of bandwidths and enable arbitrary content-based composition, their current formats have shortcomings like large overheads, live streaming delays, etc. We have therefore developed an adaptive media player that works around these problems while still using standard components like H.264/AVC for video, and MP3 for audio. The systems adaptivity allows the player to pick a quality level that makes good use of available bandwidth and CPU resources while at the same time maintaining smooth uninterrupted playback, as well as offering near instant seek and startup times. This paper presents an appropriate way of coding the segments and a simple multimedia container format that is optimized for adaptive streaming and video composition over HTTP. We show that our format is sufficiently advanced to contain any payload type, while being trivial to parse and translate to other container formats. Additionally, we show that our format is second to none in terms of overhead, without incurring any penalties on live streaming.