863 research outputs found
In-Order Delivery Delay of Transport Layer Coding
A large number of streaming applications use reliable transport protocols
such as TCP to deliver content over the Internet. However, head-of-line
blocking due to packet loss recovery can often result in unwanted behavior and
poor application layer performance. Transport layer coding can help mitigate
this issue by helping to recover from lost packets without waiting for
retransmissions. We consider the use of an on-line network code that inserts
coded packets at strategic locations within the underlying packet stream. If
retransmissions are necessary, additional coding packets are transmitted to
ensure the receiver's ability to decode. An analysis of this scheme is provided
that helps determine both the expected in-order packet delivery delay and its
variance. Numerical results are then used to determine when and how many coded
packets should be inserted into the packet stream, in addition to determining
the trade-offs between reducing the in-order delay and the achievable rate. The
analytical results are finally compared with experimental results to provide
insight into how to minimize the delay of existing transport layer protocols
Optimization flow control -- I: Basic algorithm and convergence
We propose an optimization approach to flow control where the objective is to maximize the aggregate source utility over their transmission rates. We view network links and sources as processors of a distributed computation system to solve the dual problem using a gradient projection algorithm. In this system, sources select transmission rates that maximize their own benefits, utility minus bandwidth cost, and network links adjust bandwidth prices to coordinate the sources' decisions. We allow feedback delays to be different, substantial, and time varying, and links and sources to update at different times and with different frequencies. We provide asynchronous distributed algorithms and prove their convergence in a static environment. We present measurements obtained from a preliminary prototype to illustrate the convergence of the algorithm in a slowly time-varying environment. We discuss its fairness property
Recommended from our members
Intra- and Inter-Session Network Coding in Wireless Networks
In this paper, we are interested in improving the performance of constructive
network coding schemes in lossy wireless environments.We propose I2NC - a
cross-layer approach that combines inter-session and intra-session network
coding and has two strengths. First, the error-correcting capabilities of
intra-session network coding make our scheme resilient to loss. Second,
redundancy allows intermediate nodes to operate without knowledge of the
decoding buffers of their neighbors. Based only on the knowledge of the loss
rates on the direct and overhearing links, intermediate nodes can make
decisions for both intra-session (i.e., how much redundancy to add in each
flow) and inter-session (i.e., what percentage of flows to code together)
coding. Our approach is grounded on a network utility maximization (NUM)
formulation of the problem. We propose two practical schemes, I2NC-state and
I2NC-stateless, which mimic the structure of the NUM optimal solution. We also
address the interaction of our approach with the transport layer. We
demonstrate the benefits of our schemes through simulations
Network Coded TCP (CTCP) Performance over Satellite Networks
We show preliminary results for the performance of Network Coded TCP (CTCP)
over large latency networks. While CTCP performs very well in networks with
relatively short RTT, the slow-start mechanism currently employed does not
adequately fill the available bandwidth when the RTT is large. Regardless, we
show that CTCP still outperforms current TCP variants (i.e., Cubic TCP and
Hybla TCP) for high packet loss rates (e.g., >2.5%). We then explore the
possibility of a modified congestion control mechanism based off of H-TCP that
opens the congestion window quickly to overcome the challenges of large latency
networks. Preliminary results are provided that show the combination of network
coding with an appropriate congestion control algorithm can provide gains on
the order of 20 times that of existing TCP variants. Finally, we provide a
discussion of the future work needed to increase CTCP's performance in these
networks.Comment: 4 pages, 4 figures, Accepted at SPACOMM 201
RED behavior with different packet sizes
We consider the adaptation of random early detection (RED) as a buffer
management algorithm for TCP traffic in Internet gateways where different
maximum transfer units (MTUs) are used. We studied the two RED variants
described in [4] and point out a weakness in both. The first variant where drop
probability is independent from the packet size discriminates connections with
smaller MTUs. The second variant results in a very high packet loss ratio
(PLR), and as a consequence low goodput, for connections with higher MTUs. We
show that fairness in terms of loss and goodput can be supplied through an
appropriate setting of the RED algorithm.Comment: 13 pages, submitted to IEEE symposium on computer communications
(ISCC2000
- …