Congestion Control in Real Time Applications (original) (raw)
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TFRC for Congestion Control in Real Time Applications
The World Academy of Research in Science and Engineering
The amount of traffic generated by Real Time Applications (RTA) has increased substantially over the years. RTA will face congestion while there's any form of bottleneck restricting traffic, this can lead to packet loss or delayed traffic that is unacceptable for RTAs. Therefore it is desirable for RTA to implement congestion control mechanism to improve the steadiness of networks. The congestion pr oblem has been addressed successfully by Transmission Control Protocol (TCP). TCP is connection oriented protocol that presents reliable and ordered delivery of packets and also presents end - to - end congestion control mechanism. But its congestion control m echanism does not suit the characteristics of the RTA. User Datagram Protocol (UDP) applications can send data at constant bit rate. It is non TCP based protocol. It cannot adjust its flow rate when congestion is detected and it continues to send at origin al rate. So these non TCP applications don't have congestion control mechanism and don't share bandwidth fairly with TCP based applications. A new congestion control protocol for datagram transport was defined i.e., TCP Friendly Rate Control (TFRC) standar dized by Internet Engineering Task Force (IETF). TFRC is a congestion control algorithm that supplies a smooth transmission rate for RTAs. TFRC is a congestion control mechanism for unicast flows functioning in a best effort Internet environment. It's reas onably fair when competing for bandwidth with TCP flows in congested network, although encompasses a lot of lower variation of throughput over time compared with TCP
Proceedings of the First International ICST Conference on Simulation Tools and Techniques for Communications Networks and Systems, 2008
In this paper, we present a modification of the ns2 code for the RTP/RTCP protocols. The legacy RTP/RTCP code in ns2 has not yet been validated but it provides a framework of the protocol's specification for experimental use. We have modified the code by adding all the RTP/RTCP protocol's attributes that are defined in RFC 3550 and related to QoS metrics. We have also implemented additional algorithms and functions in order to enhance our modified code with TCP friendly bandwidth share behavior. Our protocol, named RTPUP ("UP" stands for the University of Patras), is offered as a package and is fully documented so that it can be used for simulations and research within the ns2 simulation environment.
Demonstration of the Functioning of TCP Protocol Used for Network Congestion Control
Congestion can occur when the quality of service in a network reduces as a result of a node or link conveying too many data. TCP is the most widely used protocol for Internet traffic, including email, web browsing, data and an increasing portion of multimedia content delivered in real time using the HTTP/TCP protocols. Performances of existing TCP congestion control algorithms degrade significantly when deployed over wireless networks. TCP was designed primarily for reliability as opposed to real time delivery, but the problem is particularly severe for real time applications, such as, HTTP/TCP based streaming. In this paper, we carried out a research on the TCP’s four related congestion control algorithms, namely: slow-start, congestion avoidance, fast retransmit and fast recovery. We studied the behaviour and implementation of slow-start and congestion avoidance algorithms, as well as the modifications to the fast retransmit and fast recovery. We used the OPNET Network Model as our methodology. The TCP performance on the network was modeled, first without background traffic and then with background traffic. We compared these algorithms using the same network model to deterministically check several scenarios; and simulations were conducted to ascertain the differences between the congestion control algorithms studied and OPNET’s software. The results gotten showed that using different algorithms, traffic could actually be fine tuned in the network being modeled so as to achieve higher Performance. The adjustments were done in the OPNET simulator. Call for Papers: https://sites.google.com/site/ijcsis/
RTP Control Protocol (RTCP) Feedback for Congestion Control
2021
An effective RTP congestion control algorithm requires more fine-grained feedback on packet loss, timing, and Explicit Congestion Notification (ECN) marks than is provided by the standard RTP Control Protocol (RTCP) Sender Report (SR) and Receiver Report (RR) packets. This document describes an RTCP feedback message intended to enable congestion control for interactive realtime traffic using RTP. The feedback message is designed for use with a sender-based congestion control algorithm, in which the receiver of an RTP flow sends back to the sender RTCP feedback packets containing the information the sender needs to perform congestion control.
Congestion control for real-time communications: A comparison between NADA and GCC
2016 24th Mediterranean Conference on Control and Automation (MED), 2016
Congestion control for Web real-time communication (WebRTC) is a hot topic currently addressed at the IETF. Differently from congestion control for TCP, congestion control for WebRTC not only aims at containing packet losses, but also aims at minimizing queuing delays to provide interactivity. In this paper we describe two algorithms under discussion at IETF: Network Assisted Dynamic Adaptation (NADA) proposed by Cisco Systems and Google Congestion Control (GCC) proposed by Google. A performance comparison in a simulation environment is carried out. Results show that GCC exhibits slow convergence whereas NADA exhibits a remarkable oscillating behavior.
IEEE/ACM Transactions on Networking, 2005
Currently there is no control for real-time traffic sources in IP networks. This is a serious problem because real-time traffic can not only congest the network but can also cause unfairness and starvation of TCP traffic. However, it is not possible to apply current solutions for Internet to the networks with high bandwidth-delay products and high bit error rates. The channel errors may result in inaccurate congestion control decisions and unnecessary rate throttles leading to severe performance degradation. This problem is amplified in the links with high bandwidth-delay products, since the link is inefficiently utilized for a very long time until the unnecessary rate throttle is recovered. In this paper, a new Rate Control Scheme, RCS, is introduced for real-time interactive applications in networks with high bandwidth-delay products and high bit error rates. RCS is based on the concept of using dummy packets to probe the availability of network resources. Dummy packets are treated as low priority packets and consequently they do not affect the throughput of actual data traffic. Therefore, RCS requires all the routers in the connection path to support some priority policy. A new algorithm is also proposed to improve the robustness of the RCS to temporal signal loss conditions. The delay-bound considerations for real-time traffic sources using RCS rate control scheme are also investigated. Simulation experiments show that in environments with high bandwidth-delay products and high bit error rates, RCS achieves high throughput performance without penalizing TCP connections. Index Terms-Flow control, high bandwidth-delay products, high bit error rates, real-time protocols.
Performance Analysis of TCP Congestion Control Algorithms
The demand for fast transfer of large volumes of data, and the deployment of the network infrastructures is ever increasing. However, the dominant transport protocol of today, TCP, does not meet this demand because it favors reliability over timeliness and fails to fully utilize the network capacity due to limitations of its conservative congestion control algorithm. The slow response of TCP in fast long distance networks leaves sizeable unused bandwidth in such networks. A large variety of TCP variants have been proposed to improve the connection's throughput by adopting more aggressive congestion control algorithms. Some of the flavors of TCP congestion control are loss-based, high-speed TCP congestion control algorithms that uses packet losses as an indication of congestion; delay-based TCP congestion control that emphasizes packet delay rather than packet loss as a signal to determine the rate at which to send packets. Some efforts combine the features of loss-based and delay-based algorithms to achieve fair bandwidth allocation and fairness among flows. A comparative analysis between different flavors of TCP congestion control namely Standard TCP congestion control (TCP Reno), loss-based TCP congestion control (HighSpeed TCP, Scalable TCP, CUBIC TCP), delay-based TCP congestion control (TCP Vegas) and mixed loss-delay based TCP congestion control (Compound TCP) is presented here in terns of congestion window verses elapsed time after the connection is established.
Comparative performance analysis of TCP-based congestion control algorithms
International Journal of Communication Networks and Distributed Systems, 2016
In order to curtail the escalating packet loss rates caused by an exponential increase in network traffic, active queue management techniques such as Random Early Detection (RED) have come into picture. Flow Random Early Drop (FRED) keeps state based on instantaneous queue occupancy of a given flow. FRED protects fragile flows by deterministically accepting flows from low bandwidth connections and fixes several shortcomings of RED by computing queue length during both arrival and departure of the packet. Stochastic Fair Queuing (SFQ) ensures fair access to network resources and prevents a busty flow from consuming more than its fair share. In case of (Random Exponential Marking) REM, the key idea is to decouple congestion measure from performance measure (loss, queue length or delay). Stabilized RED (SRED) is another approach of detecting nonresponsive flows. In this paper, we have shown a comparative analysis of throughput, delay and queue length for the various congestion control algorithms RED, SFQ and REM. We also included the comparative analysis of loss rate having different bandwidth for these algorithms.
Exploration and evaluation of traditional TCP congestion control techniques
TCP or Transmission Control Protocol represents one of the prevailing ''languages'' of the Internet Protocol Suite, complementing the Internet Protocol (IP), and therefore the entire suite is commonly referred to as TCP/IP. TCP provides reliability to data transferring in all end-to-end data stream services on the internet. This protocol is utilized by major internet applications such as the e-mail, file transfer, remote administration and world-wide-web. Other applications which do not require reliable data stream service may use the User Datagram Protocol (UDP), which provides a datagram service that emphasizes reduced latency over reliability. The task of determining the available bandwidth of TCP packets flow is in fact, very tedious and complicated. The complexity arises due to the effects of congestion control of both the network dynamics and TCP. Congestion control is an approved mechanism used to detect the optimum bandwidth in which the packets are to be sent by TCP sender. The understanding of TCP behaviour and the approaches used to enhance the performance of TCP in fact, still remain a major challenge. In conjunction to this, a considerable amount of researches has been made, in view of developing a good mechanism to raise the efficiency of TCP performance. The article analyses and investigates the congestion control technique applied by TCP, and indicates the main parameters and requirements required to design and develop a new congestion control mechanism.