Principles of congestion control in Java

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Principles of congestion control
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Figure 36-3: Scenario 2: two hosts (with retransmissions) and a router with finite buffers
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Figure 36-4: Scenario 2 performance: (a) no retransmissions (b) only needed retransmisisons (c) extraneous, undeeded retransmissions The performance realized under scenario 2 will now depend strongly on how retransmission is performed First, consider the unrealistic case that Host A is able to somehow (magically!) determine whether or not a buffer is free in the router and thus sends a packet only when a buffer is free In this case, no loss would occur, in would be equal to in ' , and the throughput of the connection would be equal to in This case is shown in Figure 36-4(a) From a throughput standpoint, performance is ideal - everything that is sent is received Note that the average host sending rate can not exceed C/2 under this scenario, since packet loss is assumed never to occur
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Principles of congestion control
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Consider next the slightly more realistic case that the sender retransmits only when a packet is known for certain to be lost (Again, this assumption is a bit of a stretch However, it possiible that the sending host might set its timeout large enough to be virtually assured that a packet that has not been ACKed has been lost) In this case, the performance might look something like that shown in Figure 36-4(b) To appreciate what is happening here, consider the case that the offered load, in' (the rate of original data transmission plus retransmissions), equals 6C According to FIgure 36-4(b), at this value of the offered load, the rate at which data are delivered to the receiver application is C/3 Thus, out of the 6C units of data transmitted, 3333 bytes/sec (on average) are original data and 26666 bytes per second (on average) are retransmitted data We see here another "cost" of a congested network - the sender must perform retransmissions in order to compensate for dropped (lost) packets due to buffer overflow Finally, let us consider the more realistic case that the sender may timeout prematurely and retransmit a packet that has been delayed in the queue, but not yet lost In this case, both the original data packet and the retransmission may both reach the receiver Of course, the receiver needs but one copy of this packet and will discard the retransmission In this case, the "work" done by the router in forwarding the retransmitted copy of the original packet was "wasted," as the receiver will have already received the original copy of this packet The router would have better used the link transmission capacity transmitting a different packet instead Here then is yet another "cost" of a congested network unneeded retransmissions by the sender in the face of large delays may cause a router to use its link bandwidth to forward uneeded copies of a packet Figure 364(c) shows the throughput versus offered load when each packet is assumed to be forwarded (on average) at least twice by the router Since each packet is forwarded twice, the throughput achieved will be bounded above by the two-segment curve with the asymptotic value of C/4
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Scenario 3: Four senders, routers with finite buffers, and multihop paths
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In our final congestion scenario, four hosts transmit packets, each over overlapping two-hop paths, as shown in Figure 365 We again assume that each host uses a timeout/retransmission mechanism to implement a reliable data transfer service, that all hosts have the same value of in , and that all router links have capacity C bytes/sec
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Principles of congestion control
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Figure 36-5: Four senders, routers with finite buffers, and multihop paths Let us consider the connection from Host A to Host C, passing through Routers R1 and R2 The A-C connection shares router R1 with the D-B connection and shares router R2 with the B-D connection For extremely small values of in , buffer overflows are rare (as in congestion scenarios 1 and 2), and the throughput approximately equals the offered load For slightly larger values of in , the corresponding throughput is also larger, as more original data is being transmitted into the network and delivered to the destination, and overflows are still rate Thus, for small values of in , an increase in in results in an increase in out Having considered the case of extremely low traffic, let us next examine the case that in (and hence in') is extremely large Consider router R2 The A-C traffic arriving to router R2 (which arrives at R2 after being forwarded from R1) can have an arrival rate at R2 that is at most C, the capacity of the link from R1 to R2, regardless of the value of in If in' is extremely large for all connections (including the B-D connection), then the arrival rate of B-D traffic at R2 can be much larger than that of the A-C traffic Because the A-C and B-D traffic must compete at router R2 for the limited amount of buffer space, the amount of A-C traffic that successfully gets through R2 (ie, is not lost due to buffer overflow) becomes smaller and smaller as the offered load from B-D gets larger and larger In the limit, as the offered load approaches infinity, an empty buffer at R2 is immediately filled by a B-D packet and the throughput of the A-C connection at R2 goes to zero This, in turn, implies that the A-C end-end throughput goes to zero in the limt of heavy traffic These considerations give rise to the offered load versus throughput tradeoff shown below in Figure 36-6
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Figure 36-6: Scenario 2 performance with finite buffers and multihope paths The reason for the eventual decrease in throughput with increasing offered load is evident when one considers the amount of wasted "work" done by the network In the high traffic scenario outlined above, whenever a packet is dropped at a second-hop router, the "work" done by the first-hop router in forwarding a packet to the second-hop router ends up being "wasted" The network would have been equally well off (more accurately, equally as bad off) if the first router had simply discarded that packet and remained idle More to the point, the transmission capacity used at the first router to forward the packet to the second router could have been much more profitably used to transmit a different packet (For example, when selecting a packet for transmission, it might be better for a router to give priorty to packets that have already traversed some number of upstream routers) So here we see yet another cost of dropping a packet due to congestion - when a packet is dropped along a path, the transmission capacity that was used at each of the upstream routers to forward that packet to the point at which it is dropped ends up having been wasted
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