Skype Video congestion control: An experimental investigation

Proceedings of IEEE INFOCOM, 2001

Streaming video has become a popular form of transferring video over wireless networks. TCP-friendly rate control (TFRC) is used as a streaming video transport protocol. Using both TCP congestion response function and current network conditions, TFRC adjusts its sending rate to yield the maximum TCP-friendly throughput. Since TFRC was designed for applications that would prefer to maintain a slowly-changing sending rate, it is less responsive to changes in handover between next-generation wireless overlay network. This paper shows a new TFRC method for vertical handover (VHO) over wireless overlay network. The proposed TFRC method uses explicit VHO messages for faster rate adaptation to a new target network. The simulation results show that the proposed one provides better QoS and throughput supports than the conventional TFRC scheme during VHO.

IEEE Transactions on Multimedia, 2000

This paper presents a media-and TCP-friendly ratebased congestion control algorithm (MTFRCC) for scalable video streaming in the Internet. The algorithm integrates two new techniques: i) a utility-based model using the rate-distortion function as the application utility measure for optimizing the overall video quality; and ii) a two-timescale approach of rate averages (longterm and short-term) to satisfy both media and TCP-friendliness. We evaluate our algorithm through simulation and compare the results against the TCP-friendly rate control (TFRC) algorithm. For assessment, we consider five criteria: TCP fairness, responsiveness, aggressiveness, overall video quality, and smoothness of the resulting bit rate. Our simulation results manifest that MTFRCC performs better than TFRC for various congestion levels, including an improvement of the overall video quality.

Proc IEEE International Conference …, 2006

Existing congestion controllers have been designed with TCP traffic in mind. Changing traffic patterns on the Internet imply that on some tight links all UDP video streams will occur. Three different congestion controllers (RAP, TFRC, and fuzzy-logic based), already successful in avoiding instability in current TCPdominated internets, were tested across a tight link in which video traffic dominated. Congestion control is either achieved by modulating the sending rate in response to feedback of packet loss rates and/or round-trip delays (RAP/TFRC) or a congestion level based on packet dispersion across a network path (fuzzy controller). The controllers were found to differ in the smoothness of resulting video clip streams, with the fuzzy and TFRC controllers, in that order, producing the smoothest received video. Tests also demonstrated that, when controlled flows of different types compete across a tight link, it is possible for the sending rate of TFRC to exceed the available bandwidth, resulting in excess packet loss and implying quite poor video quality at the receiver. The results show that fuzzy-logic control is more flexible when video streams dominate.

Lecture Notes in Computer Science, 2002

In the digital era, the increase of network bandwidth and the ubiquitous wireless access facilitates the creation of more and more innovative network services. Among these services, Voice over Internet Protocol (VoIP) is surely one of the most popular and successful real-time multimedia services on the Internet. For decades, User Datagram Protocol (UDP) has been adopted to transport the voice streams of VoIP applications on the Internet. In this thesis, we presented how the Datagram Congestion Control Protocol (DCCP), which was recently developed by Internet Engineering Task Force (IETF), can be utilized by the Real-time Transport Protocol (RTP) to transport real-time audio streams. Unlike UDP which is a connectionless protocol, DCCP is a connection-oriented protocol which requires both ends to establish a connection before they can begin sending and receiving packets. We proposed the design and architecture of a VoIP application running on DCCP, and take Linphone, an open-source Internet VoIP phone on Linux, as an example to illustrate how to apply DCCP to establish a bidirectional Session Initiation Protocol (SIP) communication.

2012

Abstract Standardisation of Web Real-Time Communications (WebRTC) and Telepresence will likely lead to greatly increased deployment of interactive multimedia applications on the Internet. The resulting high-rate multimedia traffic may cause congestion, but few congestion control solutions have been proposed. We suggest that the congestion control, codec, and application be coupled and engage in a dialogue on the correct response to congestion; and that congestion control consider semantic feedback, not just transport-layer metrics.

The stability and performance of the Internet to date have in a large part been due to the congestion control mechanism employed by TCP. However, while the TCP congestion control is appropriate for traditional applications such as bulk data transfer, it has been found less than ideal for multimedia applications. In particular, audio and video streaming applications have difficulties managing the rate halving performed by TCP in response to congestion. To this end, the majority of multimedia applications use either a congestion control scheme which reacts less drastic to congestion and therefore often is more aggressive than TCP, or, worse yet, no congestion control whatsoever. Since the Internet community strongly fears that a rapid deployment of multimedia applications which do not behave in a fair and TCP-friendly manner could endanger the current stability and performance of the Internet, a broad spectrum of TCP-friendly congestion control schemes have been proposed. In this report, a survey over contemporary proposals of TCP-friendly congestion control mechanisms for multimedia traffic in the Internet is presented. A classification scheme is outlined which shows how the majority of the proposed congestion control schemes emanate from a relatively small number of design principles. Furthermore, we illustrate how these design principles have been applied in a selection of congestion control scheme proposals and actual transport protocols.

Computer Networks, 2007

In this paper, we present a new end-to-end protocol, namely Scalable Streaming Video Protocol (SSVP), which operates on top of UDP and is optimized for unicast video streaming applications. SSVP employs Additive Increase Multiplicative Decrease (AIMD)-based congestion control and adapts the sending rate by properly adjusting the inter-packet-gap (IPG). The smoothness-oriented modulation of AIMD parameters and IPG adjustments reduce the magnitude of AIMD oscillation and allow for smooth transmission patterns, while TCP-friendliness is maintained. Our experimental results demonstrate that SSVP eventually adapts to the vagaries of the network and achieves remarkable performance on real-time video delivery. In the event where awkward network conditions impair the perceptual video quality, we investigate the potential improvement via a layered adaptation mechanism that utilizes receiver buffering and adapts video quality along with long-term variations in the available bandwidth. The adaptation mechanism sends a new layer based on explicit criteria that consider both the available bandwidth and the amount of buffering at the receiver, preventing wasteful layer changes that have an adverse effect on user-perceived quality. Quantifying the interactions of SSVP with the specific adaptation scheme, we identify notable gains in terms of video delivery, especially in the presence of limited bandwidth.

Journal of Visual Communication and Image …, 1998

We present a new technique for streaming real time video on today's Internet, based on dynamic rate shaping and TCP ow control. Dynamic rate shaping is a signal processing technique that adapts the rate of compressed video MPEG-1, MPEG-2, H.26x to dynamically varying bandwidth constraints. This provides an interface or lter between the source and the network, with which the encoder's output either live or stored can be perfectly matched to the network's available bandwidth. We couple this adaptation capability with the use of a new semi-reliable protocol that uses the TCP congestion window to pace the delivery of data into the network, but without using other TCP algorithms that are poorly suited to real time media. Use of TCP ow control ensures that the protocol competes fairly with all other TCP data, and that it optimally shares the available bandwidth. We also describe a real application that uses this approach to stream MPEG video on the Internet. We present several experiments, performed in both a controlled environment and the wide area Internet, that were used to evaluate the e ectiveness and fairness of the scheme. The results show that the proposed solution achieves superior video quality while at the same time providing fairness by equally sharing bandwidth with other non-real-time connections. It also avoids the latency problems commonly associated with TCP.

2016 12th International Conference on Network and Service Management (CNSM)

Video streaming services have continuously gained popularity over the last decades, accounting for about 70% of all consumer Internet traffic in 2016. All of these video streaming sessions have strict delivery deadlines in order to avoid playout interruptions, detrimentally impacting the Quality of Experience (QoE). However, the vast majority of this traffic uses TCP at the transport layer, which is known to be far from minimizing the number of deadline-missing streams. By introducing deadlineawareness at the transport layer, video delivery can be optimized by prioritizing specific flows. This paper proposes a deadlineaware congestion control mechanism, based on a parametrization of the traditional TCP New Reno congestion control strategy. By taking into account the available deadline information, the modulation of the congestion window is dynamically adapted to steer the aggressiveness of a considered stream. The proposed approach has been thoroughly evaluated in both a video-ondemand (VoD)-only scenario and a scenario where VoD streams co-exist with live streaming sessions and non-deadline-aware traffic. It was shown that in a video streaming scenario the minimal bottleneck bandwidth can be reduced by 16% on average when using deadline-aware congestion control. In coexistence with other TCP traffic, a bottleneck reduction of 11% could be achieved.

IEEE Transactions on Automatic Control, 2000

Nowadays the Internet is changing from being only an efficient platform for data delivering to become a major platform for many other applications such as Voice over IP. The stability of the traditional data Internet is due to the congestion control algorithm developed by V. Jacobson for the TCP. However, the TCP congestion control is not optimal for VoIP applications because of its retransmission mechanism and additive increase multiplicative decrease sliding window control. As a consequence, VoIP applications often employ proprietary and hidden congestion control algorithms executed over the UDP protocol. In this paper we focus on Skype audio, which is the most known and used VoIP application, in order to derive a mathematical model of its congestion control algorithm. To the purpose, the controller input/output variables are first identified and then their dynamic relation is described in the form of a hybrid automaton. Main findings are: (1) Skype does not implement a delay based control; (2) the loss ratio is the main input that affects the sending rate; (3) the sending rate matches the available bandwidth with a finite error.