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

Congestion Control using FEC for Conversational Multimedia Communication

In this paper, we propose a new rate control algorithm for conversational multimedia flows. In our approach, along with Real-time Transport Protocol (RTP) media packets, we propose sending redundant packets to probe for available bandwidth. These redundant packets are Forward Error Correction (FEC) encoded RTP packets. A straightforward interpretation is that if no losses occur, the sender can increase the sending rate to include the FEC bit rate, and in the case of losses due to congestion the redundant packets help in recovering the lost packets. We also show that by varying the FEC bit rate, the sender is able to conservatively or aggressively probe for available bandwidth. We evaluate our FEC-based Rate Adaptation (FBRA) algorithm in a network simulator and in the real-world and compare it to other congestion control algorithms

Rate-control for conversational H.264 video communication in heterogeneous networks

The transmission bit rate available along a communication path in a heterogeneous network is highly variable. The wireless link quality may vary due to interference and fading phenomena and, peered with radio layer reconfiguration and link layer protection mechanisms, lead to varying error rates, latencies, and, most importantly, changes in the available bit rate. And in both fixed and wireless networks, varying amounts of cross traffic from other nodes (i.e., the total offered load on the individual links of a network path) may lead to fluctuations in queue size (reflected again in a path latency) and to congestion (reflected in packet drops from router quenes). Senders have to adapt dynamically to these network conditions and adjust their sending rate and possibly other transmission parameters (such as encoding or redundancy) to match the available bit rate while maximizing the media quality perceived at the receiver. We investigate congestion indicators and their characteristics in different multimedia environments. Taking these characteristics into account, we propose a rate-adaptation algorithm that works in the following environments: a) Mobile-Mobile, b) Internet-Internet and c) Heterogeneous, Mobile-Internet scenarios. Using metrics such as Peak Signal-to-Noise Ratio (PSNR), loss rate, bandwidth utilization and fairness, we compare the algorithm with other rate-control algorithms for conversational video communication

Protocols and Algorithms for Adaptive Multimedia Systems

The deployment of WebRTC and telepresence systems is going to start a wide-scale adoption of high quality real-time communication. Delivering high quality video usually corresponds to an increase in required network capacity and also requires an assurance of network stability. A real-time multimedia application that uses the Real-time Transport Protocol (RTP) over UDP needs to implement congestion control since UDP does not implement any such mechanism. This thesis is about enabling congestion control for real-time communication, and deploying it on the public Internet containing a mixture of wired and wireless links. A congestion control algorithm relies on congestion cues, such as RTT and loss. Hence, in this thesis, we first propose a framework for classifying congestion cues. We classify the congestion cues as a combination of: where they are measured or observed? And, how is the sending endpoint notified? For each there are two options, i.e., the cues are either observed and reported by an in-path or by an off-path source, and, the cue is either reported in-band or out-of-band, which results in four combinations. Hence, the framework provides options to look at congestion cues beyond those reported by the receiver. We propose a sender-driven, a receiver-driven and a hybrid congestion control algorithm. The hybrid algorithm relies on both the sender and receiver co-operating to perform congestion control. Lastly, we compare the performance of these different algorithms. We also explore the idea of using capacity notifications from middleboxes (e.g., 3G/LTE base stations) along the path as cues for a congestion control algorithm. Further, we look at the interaction between error-resilience mechanisms and show that FEC can be used in a congestion control algorithm for probing for additional capacity. We propose Multipath RTP (MPRTP), an extension to RTP, which uses multiple paths for either aggregating capacity or for increasing error-resilience. We show that our proposed scheduling algorithm works in diverse scenarios (e.g., 3G and WLAN, 3G and 3G, etc.) with paths with varying latencies. Lastly, we propose a network coverage map service (NCMS), which aggregates throughput measurements from mobile users consuming multimedia services. The NCMS sends notifications to its subscribers about the upcoming network conditions, which take these notifications into account when performing congestion control. In order to test and refine the ideas presented in this thesis, we have implemented most of them in proof-of-concept prototypes, and conducted experiments and simulations to validate our assumptions and gain new insights.

Quality aspects of Internet telephony

Internet telephony has had a tremendous impact on how people communicate. Many now maintain contact using some form of Internet telephony. Therefore the motivation for this work has been to address the quality aspects of real-world Internet telephony for both fixed and wireless telecommunication. The focus has been on the quality aspects of voice communication, since poor quality leads often to user dissatisfaction. The scope of the work has been broad in order to address the main factors within IP-based voice communication. The first four chapters of this dissertation constitute the background material. The first chapter outlines where Internet telephony is deployed today. It also motivates the topics and techniques used in this research. The second chapter provides the background on Internet telephony including signalling, speech coding and voice Internetworking. The third chapter focuses solely on quality measures for packetised voice systems and finally the fourth chapter is devoted to the history of voice research. The appendix of this dissertation constitutes the research contributions. It includes an examination of the access network, focusing on how calls are multiplexed in wired and wireless systems. Subsequently in the wireless case, we consider how to handover calls from 802.11 networks to the cellular infrastructure. We then consider the Internet backbone where most of our work is devoted to measurements specifically for Internet telephony. The applications of these measurements have been estimating telephony arrival processes, measuring call quality, and quantifying the trend in Internet telephony quality over several years. We also consider the end systems, since they are responsible for reconstructing a voice stream given loss and delay constraints. Finally we estimate voice quality using the ITU proposal PESQ and the packet loss process. The main contribution of this work is a systematic examination of Internet telephony. We describe several methods to enable adaptable solutions for maintaining consistent voice quality. We have also found that relatively small technical changes can lead to substantial user quality improvements. A second contribution of this work is a suite of software tools designed to ascertain voice quality in IP networks. Some of these tools are in use within commercial systems today

Performance analysis of topologies for Web-based Real-Time Communication (WebRTC)

Real-time Communications over the Web (WebRTC) is being developed to be the next big improvement for rich web applications. This enabler allow developers to implement real-time data transfer between browsers by using high level Application Programing Interfaces (APIs). Running real-time applications in browsers may lead to a totally new scenario regarding usability and performance. Congestion control mechanisms may influence the way this data is sent and metrics such as delay, bit rate and loss are now crucial for browsers. Some mechanisms that have been used in other technologies are implemented in those browsers to handle the internals of WebRTC adding complexity to the system but hiding it from the application developer. This new scenario requires a deep study regarding the ability of browsers to adapt to those requirements and to fulfill all the features that are enabled. We investigate how WebRTC performs in a real environment running over an current web application. The capacity of the internal mechanisms to adapt to the variable conditions of the path, consumption resources and rate. Taking those principles, we test a range of topologies and use cases that can be implemented with the current version of WebRTC. Considering this scenario we divide the metrics in two categories, host and network indicators. We compare the results of those tests with the expected output based on the defined protocol in order to evaluate the ability to perform real-time media communication over the browser

Adaptive header compression techniques for mobile multimedia networks

EThOS - Electronic Theses Online ServiceGBUnited Kingdo

A novel multimedia adaptation architecture and congestion control mechanism designed for real-time interactive applications

PhDThe increasing use of interactive multimedia applications over the Internet has created a problem of congestion. This is because a majority of these applications do not respond to congestion indicators. This leads to resource starvation for responsive flows, and ultimately excessive delay and losses for all flows therefore loss of quality. This results in unfair sharing of network resources and increasing the risk of network ‘congestion collapse’. Current Congestion Control Mechanisms such as ‘TCP-Friendly Rate Control’ (TFRC) have been able to achieve ‘fair-share’ of network resource when competing with responsive flows such as TCP, but TFRC’s method of congestion response (i.e. to reduce Packet Rate) is not ideally matched for interactive multimedia applications which maintain a fixed Frame Rate. This mismatch of the two rates (Packet Rate and Frame Rate) leads to buffering of frames at the Sender Buffer resulting in delay and loss, and an unacceptable reduction of quality or complete loss of service for the end-user. To address this issue, this thesis proposes a novel Congestion Control Mechanism which is referred to as ‘TCP-friendly rate control – Fine Grain Scalable’ (TFGS) for interactive multimedia applications. This new approach allows multimedia frames (data) to be sent as soon as they are generated, so that the multimedia frames can reach the destination as quickly as possible, in order to provide an isochronous interactive service. This is done by maintaining the Packet Rate of the Congestion Control Mechanism (CCM) at a level equivalent to the Frame Rate of the Multimedia Encoder.The response to congestion is to truncate the Packet Size, hence reducing the overall bitrate of the multimedia stream. This functionality of the Congestion Control Mechanism is referred to as Packet Size Truncation (PST), and takes advantage of adaptive multimedia encoding, such as Fine Grain Scalable (FGS), where the multimedia frame is encoded in order of significance, Most to Least Significant Bits. The Multimedia Adaptation Manager (MAM) truncates the multimedia frame to the size indicated by the Packet Size Truncation function of the CCM, accurately mapping user demand to available network resource. Additionally Fine Grain Scalable encoding can offer scalability at byte level granularity, providing a true match to available network resources. This approach has the benefits of achieving a ‘fair-share’ of network resource when competing with responsive flows (as similar to TFRC CCM), but it also provides an isochronous service which is of crucial benefit to real-time interactive services. Furthermore, results illustrate that an increased number of interactive multimedia flows (such as voice) can be carried over congested networks whilst maintaining a quality level equivalent to that of a standard landline telephone. This is because the loss and delay arising from the buffering of frames at the Sender Buffer is completely removed. Packets sent maintain a fixed inter-packet-gap-spacing (IPGS). This results in a majority of packets arriving at the receiving end at tight time intervals. Hence, this avoids the need of using large Playout (de-jitter) Buffer sizes and adaptive Playout Buffer configurations. As a result this reduces delay, improves interactivity and Quality of Experience (QoE) of the multimedia application