TFRC and RTT thresholds interdependence in a selective retransmission scheme (original) (raw)
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TFRC-Based Selective Retransmission for Multimedia Applications
Multimedia applications are becoming increasingly popular in IP networks, while in mobile networks the limited bandwidth and the higher error rate arise in spite of its popularity. These are restraining factors for mobile clients using multimedia applications such as voice over IP (VoIP) or video streaming. In some conditions the retransmission of lost and corrupted packets should increase the quality of the multimedia service, but these retransmissions should be enabled only if the network is not in congested state. Otherwise the retransmitted packet will intensify the congestion and it will have negative effect on the audio/video quality. Our proposed mechanism selectively retransmits the corrupted packets based on the TCP-Friendly Rate Control (TFRC) and the actual video bit rate.
Computer Communications, 2008
In the past few years we have witnessed an explosive growth in the usage of media streaming applications. The newly appeared audio/video applications are becoming increasingly popular in IP networks, while in mobile environment the limited bandwidth and the higher error rate arise in spite of its popularity. Retransmission-based error recovery is considered inappropriate for multimedia applications, because of its latency. However, this solution can be attractive because it requires minimal network bandwidth, processing cost and efficiently improves the quality. Despite its latency, retransmission can be used successfully in many cases, especially if playout buffering is employed. Only the successfully retransmitted packets will improve the quality parameters of the multimedia stream, therefore it is worth to examine which packets should be retransmitted. In this paper a source controlled selective retransmission algorithm is presented with a decision algorithm based on the actual RTT and sending rate determined by the TFRC. In our scheme the transmitter determines the playout delay caused by the playout buffer using the proposed Flood method. The needed information about the network congestion state and the network delay are provided by the TFRC (TCP Friendly Rate Control) algorithm. Our proposal does not need additional administration messages because the decision procedure and its inputs are at the transmitter. The obtained results show that significant quality improvement can be achieved with the proposed selective retransmission scheme.
Content-Aware Selective Retransmission Scheme in Heavy Loaded Wireless Networks
IFIP International Federation for Information Processing, 2008
Streaming media is becoming increasingly prominent on the Internet, although multimedia applications have very stringent bandwidth, delay and loss requirements. In mobile environment the limited bandwidth and the higher error rate arise as an obstacle of its popularity. In many cases retransmission-based error recovery can be an attractive solution to improve the quality of the video stream, because it requires minimal network bandwidth and processing cost. In this paper we propose a content-aware selective retransmission scheme which allows the retransmission of all packets when the risk of congestion is low, but as it rises the retransmission is disabled step-by-step, but not all at once, in order of packet importance. In this work the heterogenity of H.264 streams were utilized for the determination of packet importance. The advantage of this transmitter controlled procedure is that all the needed information is available at the source due to DCCP transport protocol and its congestion control algorithm. The effectiveness of the proposed method was examined in Ns2 network simulator.
DCCP-based multiple retransmission technique for multimedia streaming
Proceedings of the 6th International Conference on Advances in Mobile Computing and Multimedia - MoMM '08, 2008
Retransmission-based error recovery is the simplest technique to minimize the overall packet loss ratio in order to increase the quality of the applications. Multimedia applications are becoming increasingly popular in IP networks, while in mobile environment the limited bandwidth and the higher error rate arise in spite of its popularity. Retransmission can be also used for loss recovery in media applications, but the number of retransmissions is limited by the playout buffer and the recent network delay.
An adaptive multiple retransmission technique for continuous media streams
Proceedings of the 14th international workshop on Network and operating systems support for digital audio and video, 2004
Retransmission can be used for loss recovery in continuous media applications but the number of retransmission attempts is bounded by the size of the playout buffer. For efficient recovery, a protocol must attempt as many retransmissions as possible but avoid late retransmissions. This typically requires that the playout buffer be sized in round-trip time (RTT) multiples plus some margin for error. RTT-based timers are then used to trigger retransmissions. However, this approach is problematic due to (i) the high variation in RTT commonly encountered in the Internet, which makes accurate estimation difficult, and (ii) the granularity of timers typically used prevents precise control. We present two new retransmission-based protocols, for unicast and multicast respectively, which eliminate RTT estimation and timer-triggered events. As a result, our protocols are immune to errors due to jitter and timer granularity and recover more losses, while better suppressing unnecessary retransmission requests and retransmissions than timer-based protocols. At the same time, our protocols are simpler to implement and degrade more gracefully than timer-based protocols.
A Taxonomy and Survey of Retransmission Based Partially Reliable Protocols
The mismatch between the services offered by the two standard transport protocols in the Internet, TCP and UDP, and the services required by distributed multimedia applications has led to the development of a large number of partially reliable transport protocols. That is, protocols which in terms of reliability places themselves between TCP and UDP. This paper presents a taxonomy for retransmission based, partially reliable transport protocols, i.e., the subclass of partially reliable transport protocols that performs error recovery through retransmissions. The taxonomy comprises two classification schemes: one that classifies retransmission based, partially reliable transport protocols with respect to the reliability service they offer and one that classifies them with respect to their error control scheme. The objective of our taxonomy is fourfold: to introduce a unified terminology; to provide a framework in which retransmission based, partially reliable transport protocols can be examined, compared, and contrasted; to make explicit the error control schemes used by these protocols; and, finally, to gain new insights into these protocols and thereby suggest avenues for future research. Based on our taxonomy, a survey was made of existing retransmission based, partially reliable transport protocols. The survey shows how protocols are categorized according to our taxonomy, and exemplifies the majority of reliability services and error control schemes detailed in our taxonomy.
Using Spurious Retransmissions to Adapt the Retransmission Timeout
2008
This report describes a method for using spurious retransmission timeouts to determine when the retransmission timeout is not accurately capturing the delay variance in the network. We account for this by adapting the way TCP's retransmission timeout is computed in an effort to avoid subsequent unnecessary retransmissions.
Evaluation of retransmission strategies in a local area network environment
ACM SIGMETRICS Performance Evaluation Review, 1989
We present an evaluation of retransmission strategies over local area networks. Expressions are derived for the expectation and the variance of the transmission time of the go-back-n and the selective repeat protocols in the presence of errors. These are compared to the expressions for blast with full retransmission on error (BFRE) derived by Zwaenepoel [Zwa 85]. We conclude that go-back-n performs almost as well as selective repeat and is very much simpler to implement while BFRE is stable only for a limited range of messages sizes and error rates. We also present a variant of BFRE which optimally checkpoints the transmission of a large message. This is shown to overcome the instability of ordinary BFRE. It has a simple state machine and seems to take full advantage of the low error rates of local area networks. We further investigate go-back-n by generalizing the analysis to an upper layer transport protocol, which is likely to encounter among other things, variable delays due to ...
CA-RTO: a contention-adaptive retransmission timeout
Proceedings. 14th International Conference on Computer Communications and Networks, 2005. ICCCN 2005., 2005
We show that TCP timers, based solely on RTT estimations and measurements, cannot capture with precision the level of flow contention. We notice that increased contention may stabilize RTT variation, minimize the deviation and, in turn, shorten the timeout. We show that this behavior is undesirable indeed, since it leads to unfair resource utilization. We propose CA-RTO, an algorithm that incorporates a contention parameter and a randomization technique into the Retransmission Timeout. We report significant improvement in fairness, great reduction of retransmitted packets and slight improvements in application goodput.