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Internet Engineering Task Force                             Yogesh Swami
INTERNET DRAFT                                                  Khiem Le
File: draft-swami-tsvwg-tcp-dclor-02.txt           Nokia Research Center
                                                                  Dallas
                                                      September 24, 2003
                                               Expires:   March 24, 2004


          DCLOR: De-correlated Loss Recovery using SACK option
                         for spurious timeouts.


Status of this Memo

   This document is an Internet-Draft and is in full conformance with
   all provisions of Section 10 of [RFC2026].

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF), its areas, and its working groups.  Note that
   other groups may also distribute working documents as Internet-
   Drafts.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   The list of current Internet-Drafts can be accessed at
   http://www.ietf.org/ietf/1id-abstracts.txt

   The list of Internet-Draft Shadow Directories can be accessed at
   http://www.ietf.org/shadow.html

Abstract

   A spurious timeout in TCP forces the sender to unnecessarily
   retransmit one complete congestion window of data into the network.
   In addition, TCP uses the rate of arrival of ACKs as the basic
   criterion for congestion control. TCP makes the assumption that the
   rate at which ACKs are received reflects the end-to-end state of the
   network in terms of congestion. However, ACKs after a spurious
   timeout don't reflect the end-to-end congestion state of the network;
   they only reflect the congestion state of a part of the network. In
   these cases, the slow-start behavior after a timeout can further add
   to network congestion. In this draft we propose changes to the TCP
   sender that can be used to solve the problem of both redundant-
   retransmission and network congestion after a spurious timeout.




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1. Introduction

   The response of a TCP sender after a retransmission timeout is
   governed by the underlying assumption that a mid-stream timeout can
   occur only if there is heavy congestion--manifested as packet
   loss--in the network. TCP therefore assumes that a timeout is a
   sufficient indication to a) recover all the packets in flight, and b)
   to initiate a congestion response (slow start in this case) suited
   for heavy congestion scenarios.

   Even though timeout is often a sufficient indication for recovering
   all the packets in flight and initiating slow start, the loss
   recovery algorithm should be separate from the congestion control
   decisions. The loss recovery algorithm should only answer the
   question of "what" data (i.e., what sequence numbers) to send. On the
   other hand, the congestion control algorithm should answer the
   question of "how much" data to send. But after a timeout, TCP
   addresses the issues of loss recovery and congestion control using a
   single mechanism--send one packet per round trip timeout (RTO)
   (answers the "how much" question) until an acknowledgment is
   received; the single segment sent is always the first unacknowledged
   outstanding packet in the retransmission queue (answers the "what"
   question).  Since the present TCP's loss recovery and congestion
   control algorithms are coupled together, we call this "Correlated
   Loss Recovery (CLOR)."

   Although the assumption that a timeout can occur only if there is
   severe congestion is valid for traditional wire-line networks, it
   does not hold good for some other types of networks--networks where
   packets can be stalled "in the network" for a significant duration
   without being discarded. Typical examples of such networks are
   cellular networks. In cellular networks, the link layer can
   experience a relatively long disruption due to errors, and the link
   layer protocol can keep these packets-in-error buffered as long as
   the link layer disruption lasts.

   In this document we present an alternative approach to loss recovery
   and congestion control that "De-Correlates" Loss Recovery from
   congestion congestion and allows independent choice on using a
   particular TCP sequence number without compromising on the congestion
   control principles of [RFC2581][RFC2914][RFC2861].


2. Problem Description.

   Let us assume that a TCP sender has sent N packets, p(1) ...  p(N),
   into the network and it's waiting for the ACK of p(1) (Figure-1). Due
   to bad network conditions or some other problem, these packets are



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   excessively delayed at some some intermediary node RTR-1. Unlike
   standard IP routers, RTR-1 keeps these packets buffered for a
   relatively long period of time until these packets are forwarded to
   their intended recipient.  This excessive delay forces the TCP sender
   to timeout and enter slow start.

   As far as the sender is concerned, a timeout is always interpreted as
   heavy congestion. The TCP sender therefore makes the assumption that
   all packets between p(1) and p(N) were lost in the network. To
   recover from this misconstrued loss, the TCP sender retransmits P1(1)
   ( Px(k) represents the xth retransmission of packet with sequence
   number k), and waits for the ACK a(1).

   After some period of time when the network conditions at RTR-1
   improve, the queued in packets are finally dispatched to their
   intended recipient; in response to the packet the TCP receiver
   generates the ACK a(1). When the TCP sender receives a(1), it's
   fooled into believing that a(1) was generated in response to the
   retransmitted packet p1(1), while in reality a(1) was generated in
   response to the originally transmitted packet p(1). When the sender
   receives a(1), it increases its congestion window to two, and
   retransmits p1(2) and p1(3). As the sender receives more
   acknowledgments, it continues with retransmissions and finally starts
   sending new data.

   The following two sub sections examine the problems associated with
   the above-mentioned TCP behavior.

2.1 Redundant Data Retransmission

   The obvious and relatively easy-to-solve inefficiency of the above
   algorithm is that the entire congestion window worth of data is
   unnecessarily retransmitted. Although such retransmissions are
   harmless to high-bandwidth, well-provisioned, backbone links (so long
   they are infrequent), it could severely degrade the performance of
   slow links.

   In cases where bandwidth is a commodity at a premium, (e.g., cellular
   networks), unnecessary retransmission can also be costly.

2.2 Congestion after Spurious Timeout

   To analyze network congestion after spurious timeout, we compute the
   worst case scenario packet loss in the system--assuming only TCP
   connections to be present.

   After the spurious timeout, the TCP sender sets its SS_THRESH to N/2.
   Therefore, for the first N/2 ACKs received (i.e., ACK a(1) to a(N/2)



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   ), the TCP sender will grow its congestion window by one and reach
   the SS_THRESH value of N/2.  For each ACK received, the TCP sender
   sends 2 packets. Therefore, by the end of the slow start, the TCP
   sender would have sent 2*(N/2) packets into the network. For the
   remaining N/2 ACKs (i.e., ACKs between a(N/2+1) to a(N)) the TCP
   sender will remain in the congestion avoidance phase and send one
   packet for each ACK received--sending N/2 more data segments. The net
   amount of data sent is therefore N/2 + N = 3N/2.

   Please note that the entire 3N/2 packets are injected into the
   network within a time period less than or equal to RTT in most cases.
   The number of data segments that left the network during this time is
   only N. Therefore, N/2 packets out of 3N/2 packets will be lost with
   a very high probability. These N/2 lost packets, however, need not
   come from the same connection, and such a data-burst will
   unnecessarily penalize all the competing TCP connections that share
   the same bottleneck router.

   Going further ahead, let us assume there are M competing TCP
   connections that share the same bottleneck router(s) with C(0) (each
   connection is numbered C(0) ... C(M-1)). During the period of time
   while C(0) is stalled, the TCP sender does not use its network
   resources--the buffer space--on the bottleneck router(s). The
   competing connections, C(1)...  C(M), however see this lack of
   activity as resource availability and start growing their window by
   at least one segment per RTT during this time period (by virtue of
   linear window increase during congestion avoidance phase). For
   simplicity reasons, we assume that each of these connections has the
   same round trip time of RTT, and the idle time for C(0) is k*RTT
   (where k > RTO/RTT). Under these assumptions, each of these competing
   connections will increase their congestion window by k segments.
   Therefore the amount of packets lost in the network due to slow start
   can be as high as:

                   N/2 + M*k       ... (4)

   the first term in the above equation is the packet loss due to slow
   start, while the second term is the loss due to window growth of
   completing connections (if the competing connections were in slow
   start the response could have been worse).

   Based on the above equation, we note that the congestion state of the
   network depends upon the duration of spurious timeout. In our response
   algorithm we therefore take the time duration of spurious timeout
   into account to reduce the data rate by half every RTO. Please note
   that this scheme works well only when the number of competing
   connections M does not vary too much while C(0) was stalled. A more
   conservative response algorithm should reduce the data rate to



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   INIT_WINDOW if M is not bounded.

   In the following sections we describe an algorithm that solves the
   problem of both redundant retransmission and packet loss after a
   spurious timeout.

3. De-correlated Loss Recovery (DCLOR)

   The basic idea behind DCLOR is to send a new data segment from
   outside the sender's retransmission queue and wait for the ACK or
   SACK of the new data before initiating the response algorithm. Unlike
   slow-start where the response algorithm starts immediately after
   receiving the first ACK, DCLOR waits for the ACK/SACK of the new data
   sent after timeout before initiating loss recovery. The SACK block
   for new data contains sufficient information to determine all the
   packets that were lost into the network. Once the sequence number of
   lost packets is determined, the TCP sender grows its congestion
   window as determined by the SS_THRESH and it's congestion window.

3.1  Probe phase after a timeout

   The following steps describe the response of a TCP sender on a
   timeout:

     1. If the timeout occurs before the 3 way handshake is complete,
        the TCP sender's behavior is unchanged,

     2. After each timeout, the TCP sender MUST set its congestion
        window to:

                        cwnd = max( cwnd/2, INIT_WINDOW).

        The value of SS_THRESH MUST be left unchanged at this point. The
        TCP sender should also count the number of packets in flight at
        this time, and keep it in a state variable stale_outstanding.

     3. The TCP sender SHOULD also reset all the SACK tag bits in its
        retransmission queue if this the first timeout.

     4. Instead of sending the first unacknowledged packet P1 after a
        timeout, the TCP sender should *disregard* its congestion window
        and sends ONE new MSS size data (Pn+1).

        The TCP sender should also store the sequence number of the new
        segment in a new state variable called SS_PTR (for slow start
        pointer).

        If the sender does not have any new data outside its



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        retransmission queue, or if the receiver's flow control window
        cannot sustain any new data, the TCP sender SHOULD send the
        highest sequence numbered MSS sized data chunk from its
        retransmission queue (i.e., it should send the last packet from
        its retransmission queue).

     5. A TCP sender MUST repeat step-2 to step-4 until it enters the
        Timeout-Recovery state as described in step 6.

3.2 Congestion Control After the probe phase

     6. For each ACK received with ACK-sequence number less than
        SS_PTR, the TCP sender SHOULD NOT grow it's congestion window.
        If the ACK contains a new SACK block, the SACK tag SHOULD be set
        in the corresponding data packet, and the number of packets in
        flight should be updated. If a pure ACK is received, the packet
        should be removed from the retransmission queue and the value of
        packets in flight should be updated.

        After making the above mentioned changes, the TCP sender SHOULD
        send new data (i.e., data from outside the retransmission queue)
        if the number of packets in flight is less than the congestion
        window. In addition, the TCP sender should keep a variable
        'new_packets' which counts the number of bytes (packets if
        congestion window is maintained as a count of packets) sent that
        have a sequence number greater than or equal to SS_PTR.

        In addition, the TCP sender SHOULD NOT take any timer sample for
        the stale ACKs. (NOTE: We do not attempt to change the RTT
        calculation in an ad-hoc manner; we believe that this is a
        research problem that needs better network modeling before an
        appropriate timer calculation can be found)

     7. Step-6 continues until the TCP sender receives an ACK
        with a  sequence number greater than SS_PTR, or a SACK block
        covering the sequence number greater than SS_PTR.

        If the sender receives a SACK block containing SS_PTR, i.e., if
        there is a packet loss in the stalled window, it SHOULD follow
        step-8.

        If the sender receives an ACK that acknowledges SS_PTR, i.e., if
        no packets were lost from the stalled window, it SHOULD go to
        step-10.

NOTE: In our previous experiments we had set the congestion window
   to one MSS after a spurious timeout, however this algorithm performs
   better if there is moderate load on the routers and the number of



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   competing connections do not vary a lot 0 the stalling period. In
   case of heavy load, setting the congestion window to INIT_WINDOW
   still performs better. We believe that using the present congestion
   response makes a fair compromise for different scenarios.

3.3 Timeout-Recovery: recovering lost packets after timeout

     8. The TCP sender traverses the retransmission queue and marks
        all the packets without any SACK tag as lost. The TCP sender
        also updates its packets in flight based on the SACK tags and
        the lost segment information (the packets-in-flight  should be
        ZERO after the update).

        Please note that unlike Fast-Retransmit and Fast-recovery, DCLOR
        uses only one SACK block containing SS_PTR to mark packets as
        lost.  This is because we do not expect packet reordering to
        exist over the period of RTO.

     9. The TCP sender should update its SS_THRESH, as:

                        SS_THRESH= stale_outstanding/2

    10. The TCP sender SHOULD set cwnd=new_packets+1. (Note that if
        all packets were lost, the value of 'new_packets' will be 1, and
        therefore the congestion window will become 2, which is the
        value for a timeout due to congestion.)  If packets were lost in
        the network (i.e., if a SACK for SS_PTR was received), the TCP
        sender should start by sending packets with lowest sequence
        number; else it should continue with new data.

        The sender should follow the normal window growth strategy based
        on the value of SS_THRESH after this step.

   Please note that with a pure ACK acknowledging SS_PTR, the TCP sender
   does not update the SS_THRESH value (it directly enters step-10 from
   step-7). This prevents a TCP sender from setting its SS_THRESH to a
   very small values if the spurious timeout occurs at the start of the
   connection.

4. Data Delivery To Upper Layers

   If a TCP sender loses its entire congestion window worth of data,
   sending new data after timeout prevents a TCP receiver from
   forwarding the new data to the upper layers immediately.  However,
   once the SACK for this new data is received, the TCP sender will send
   the first lost segment. This essentially means that data delivery to
   the upper layers could be delayed by at most one RTT when all the
   packets are lost in the network.



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   This, however, does not affect the throughput of the connection in
   any way. If a timeout has occurred, then the data delivery to the
   upper layers has already been excessively delayed.  Delaying it by
   another round trip is not a serious problem. Please note that
   reliability and timeliness are two conflicting issues and one cannot
   gain on one without sacrificing something else on the other.

5. Security Considerations

   The TCP SACK information is meant to be advisory, and a TCP receiver
   is allowed--though strongly discouraged--to discard data blocks the
   receiver has already SACKed [RFC2018]. Please note however that even
   if the TCP sender discards the data block it received, it MUST still
   send the SACK block for at least the recent most data received.
   Therefore in spite of SACK reneging, DCLOR will work without any
   deadlocks.

   A SACK implementation is also allowed not to send a SACK block even
   though the TCP sender and receiver might have agreed to SACK-
   Permitted option at the start of the connection. In these cases,
   however, if the receiver sends one SACK block, it must send SACK
   blocks for the rest of the connection. Because of the above mentioned
   leniency in implementation, its possible that a TCP receiver may
   agree on SACK-Permitted option, and yet not send any SACK blocks. To
   make DCLOR robust under these circumstances, DCLOR SHOULD NOT be
   invoked unless the sender has seen at least one SACK block before
   timeout. We, however, believe that once the SACK-Permitted option is
   accepted, the TCP sender MUST send a SACK block--even though that
   block might finally be discarded.  Otherwise, the SACK-Permitted
   option is completely redundant and serves little purpose. To the best
   of our knowledge, almost all SACK implementations send a SACK block
   if they have accepted the SACK-Permitted option.



6. References

     [RFC2581] M. Allman, V. Paxson, W. Stevens. "TCP Congestion
               Control," Apr, 1999.

     [RFC2914] S. Floyd, "Congestion Control Principles," Sep 2002.

     [RFC2861] M. Handley, J. Padhye, S. Floyd. "TCP Congestion
               Window Validation," Jun 2000.

     [RFC3517] E. Blanton, M. Allman, K. Fall, L. Wang, "Conservative
               SACK-based Loss Recovery Algorithm for TCP," Apr 2003.




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     [RFC2018] M. Mathis, J. Mahdavi, S. Floyd, A. Romanow, "TCP
               Selective Acknowledgment Options," Oct 1996.

     [RFC2883] S. Floyd, J. Mahdavi, M. Mathis, M. Podolsky, "An
               Extension to the Selective Acknowledgment (SACK) Option
               for TCP," Jul 2000.

     [RFC3522] R. Ludwig, M. Meyer. "The Eiffel Detection Algorithm
               for TCP," Apr 2003.

     [LG03]    R. Ludwig, A. Gurtov, "The Eifel Response Algorithm for
               TCP." Internet draft; work in progress, draft-ietf-tsvwg-
               tcp-eifel-response-03.txt, Mar 2003.

     [SK03]    P. Sarolahti, M. Kojo. "F-RTO: A TCP RTO Recovery
               Algorithm for Avoiding Unnecessary Retransmissions."
               Internet draft; work in progress.  draft-sarolahti-tsvwg-
               tcp-frto-03.txt, Jan 2003.

     [RFC2988] V. Paxon, M. Allman. "Computing TCP's Retransmission
               Timer," Nov 2000.

     [BA02]    E. Blanton, M. Allman, "Using TCP DSACKs and SCTP
               Duplicate TSNs to Detect Spurious Retransmissions,"
               Internet draft; work in progress, draft-blanton-dsack-
               use-02.txt, Oct 2002.



7. IPR Statement

   The IETF has been notified of intellectual property rights claimed in
   regard to some or all of the specification contained in this
   document. For more information consult the on-line list of claimed
   rights at http://www.ietf.org/ipr.

Author's  Address:

   Yogesh Prem Swami                       Khiem Le
   Nokia Research Center                   Nokia Research Center
   6000 Connection Drive                   6000 Connection Drive
   Irving TX-75063                         Irving TX-75063
   USA                                     USA

   Phone: +1 972-374-0669                  Phone: +1 972-894-4882
   Email: yogesh.swami@nokia.com           Email: khiem.le@nokia.com





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