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Versions: 00 01 02 03                                                   
   Internet Engineering Task Force                           Yunhong Gu
   Internet Draft                     University of Illinois at Chicago
   Intended status: Informational                        April 12, 2010
   Expires: October 12, 2010

                   UDT: UDP-based Data Transfer Protocol

Status of this Memo

   This Internet-Draft is submitted to IETF in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF), its areas, and its working groups.  Note that
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   and may be updated, replaced, or obsoleted by other documents at any
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   The list of current Internet-Drafts can be accessed at

   The list of Internet-Draft Shadow Directories can be accessed at

   This Internet-Draft will expire on October 15, 2010.

Copyright Notice

   Copyright (c) 2010 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.


   This document describes UDT, or the UDP based Data Transfer protocol.
   UDT is designed to be an alternative data transfer protocol for the
   situations when TCP does not work well. One of the most common cases,
   and also the original motivation of UDT, is to overcome TCP's

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   inefficiency in high bandwidth-delay product (BDP) networks. Another
   important target use scenario is to allow networking researchers,
   students, and application developers to easily implement and deploy
   new data transfer algorithms and protocols. Furthermore, UDT can also
   be used to better support firewall traversing.

   UDT is completely built on top of UDP. However, UDT is connection
   oriented, unicast, and duplex. It supports both reliable data
   streaming and partial reliable messaging. The congestion control
   module is an open framework that can be used to implement and/or
   deploy different control algorithms. UDT also has a native/default
   control algorithm based on AIMD rate control.

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Table of Contents

   1. Introduction...................................................4
   2. Packet Structures..............................................5
   3. UDP Multiplexer................................................8
   4. Timers.........................................................8
   5. Connection Setup and shutdown..................................9
      5.1 Client/Server Connection Setup............................10
      5.2 Rendezvous Connection Setup...............................10
      5.3 Shutdown..................................................11
   6. Data Sending and Receiving....................................11
      6.1 The Sender's Algorithm....................................11
      6.2 The Receiver's Algorithm..................................12
      6.3 Flow Control..............................................15
      6.4 Loss Information Compression Scheme.......................15
   7. Configurable Congestion Control (CCC).........................15
      7.1 CCC Interface.............................................15
      7.2 UDT's Native Control Algorithm............................16
   Security Considerations..........................................18
   Normative References.............................................18
   Informative References...........................................18
   Author's Addresses...............................................19

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   The Transmission Control Protocol (TCP) [RFC5681] has been very
   successful and greatly contributes to the popularity of today's
   Internet. Today TCP still contributes the majority of the traffic on
   the Internet.

   However, TCP is not perfect and it is not designed for every specific
   applications. In the last several years, with the rapid advance of
   optical networks and rich Internet applications, TCP has been found
   inefficient as the network bandwidth-delay product (BDP) increases.
   Its AIMD (additive increase multiplicative decrease) algorithm
   reduces the TCP congestion window drastically but fails to recover it
   to the available bandwidth quickly. Theoretical flow level analysis
   has shown that TCP becomes more vulnerable to packet loss as the BDP
   increases higher [LM97].

   To overcome the TCP's inefficiency problem over high speed wide area
   networks is the original motivation of UDT. Although there are new
   TCP variants deployed today (for example, BiC TCP [XHR04] on Linux
   and Compound TCP [TS06] on Windows), certain problems still exist.
   For example, none of the new TCP variants address RTT unfairness, the
   situation that connections with shorter RTT consume more bandwidth.

   Moreover, as the Internet continues to evolve, new challenges and
   requirements to the transport protocol will always emerge.
   Researchers need a platform to rapidly develop and test new
   algorithms and protocols. Network researchers and students can use
   UDT to easily implement their ideas on transport protocols, in
   particular congestion control algorithms, and conduct experiments
   over real networks.

   Finally, there are other situations when UDT can be found more
   helpful than TCP. For example, UDP-based protocol is usually easier
   for punching NAT firewalls. For another example, TCP's congestion
   control and reliability control is not desirable in certain
   applications of VOIP, wireless communication, etc. Application
   developers can use (with or without modification) UDT to suit their

   Due to all those reasons and motivations described above, we believe
   that it is necessary to design a well defined and developed UDP-based
   data transfer protocol.

   As its name suggest, UDT is built solely on the top of UDP [RFC768].
   Both data and control packets are transferred using UDP. UDT is
   connection-oriented in order to easily maintain congestion control,
   reliability, and security. It is a unicast protocol while multicast
   is not considered here. Finally, data can be transferred over UDT in

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   UDT supports both reliable data streaming and partial reliable
   messaging. The data streaming semantics is similar to that of TCP,
   while the messaging semantics can be regarded as a subset of SCTP

   This document defines UDT's protocol specification. The detailed
   description and performance analysis can be found in [GG07], and a
   fully functional reference implementation can be found at [UDT].

 Packet Structures

   UDT has two kinds of packets: the data packets and the control
   packets. They are distinguished by the 1st bit (flag bit) of the
   packet header.

   The data packet header structure is as following.

   0                   1                   2                   3
   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   |0|                     Packet Sequence Number                  |
   |FF |O|                     Message Number                      |
   |                          Time Stamp                           |
   |                    Destination Socket ID                      |

   The data packet header starts with 0. Packet sequence number uses the
   following 31 bits after the flag bit. UDT uses packet based
   sequencing, i.e., the sequence number is increased by 1 for each sent
   data packet in the order of packet sending. Sequence number is
   wrapped after it is increased to the maximum number (2^31 - 1).

   The next 32-bit field in the header is for the messaging. The first
   two bits "FF" flags the position of the packet is a message. "10" is
   the first packet, "01" is the last one, "11" is the only packet, and
   "00" is any packets in the middle. The third bit "O" means if the
   message should be delivered in order (1) or not (0). A message to be
   delivered in order requires that all previous messages must be either
   delivered or dropped. The rest 29 bits is the message number, similar
   to packet sequence number (but independent). A UDT message may
   contain multiple UDT packets.

   Following are the 32-bit time stamp when the packet is sent and the
   destination socket ID. The time stamp is a relative value starting

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   from the time when the connection is set up. The time stamp
   information is not required by UDT or its native control algorithm.
   It is included only in case that a user defined control algorithm may
   require the information (See Section 6).

   The Destination ID is used for UDP multiplexer. Multiple UDT socket
   can be bound on the same UDP port and this UDT socket ID is used to
   differentiate the UDT connections.

   If the flag bit of a UDT packet is 1, then it is a control packet and
   parsed according to the following structure.

   0                   1                   2                   3
   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   |1|             Type            |            Reserved           |
   |     |                    Additional Info                      |
   |                            Time Stamp                         |
   |                    Destination Socket ID                      |
   |                                                               |
   ~                 Control Information Field                     ~
   |                                                               |

   There are 8 types of control packets in UDT and the type information
   is put in bit field 1 - 15 of the header. The contents of the
   following fields depend on the packet type. The first 128 bits must
   exist in the packet header, whereas there may be an empty control
   information field, depending on the packet type.

   Particularly, UDT uses sub-sequencing for ACK packet. Each ACK packet
   is assigned a unique increasing 16-bit sequence number, which is
   independent of the data packet sequence number. The ACK sequence
   number uses bits 32 - 63 ("Additional Info") in the control packet
   header. The ACK sequence number ranges from 0 to (2^31 - 1).

   TYPE 0x0:  Protocol Connection Handshake
              Additional Info: Undefined
              Control Info:
              1) 32 bits: UDT version
              2) 32 bits: Socket Type (STREAM or DGRAM)
              3) 32 bits: initial packet sequence number
              4) 32 bits: maximum packet size (including UDP/IP headers)
              5) 32 bits: maximum flow window size
              6) 32 bits: connection type (regular or rendezvous)

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              7) 32 bits: socket ID
              8) 32 bits: SYN cookie
              9) 128 bits: the IP address of the peer's UDP socket

   TYPE 0x1:  Keep-alive
              Additional Info: Undefined
              Control Info: None

   TYPE 0x2:  Acknowledgement (ACK)
              Additional Info: ACK sequence number
              Control Info:
              1) 32 bits: The packet sequence number to which all the
                 previous packets have been received (excluding)
              [The following fields are optional]
              2) 32 bits: RTT (in microseconds)
              3) 32 bits: RTT variance
              4) 32 bits: Available buffer size (in bytes)
              5) 32 bits: Packets receiving rate (in number of packets
                          per second)
              6) 32 bits: Estimated link capacity (in number of packets
                          per second)

   TYPE 0x3:  Negative Acknowledgement (NAK)
              Additional Info: Undefined
              Control Info:
              1) 32 bits integer array of compressed loss information
                 (see section 3.9).

   TYPE 0x4:  Unused

   TYPE 0x5:  Shutdown
              Additional Info: Undefined
              Control Info: None

   TYPE 0x6:  Acknowledgement of Acknowledgement (ACK2)
              Additional Info: ACK sequence number
              Control Info: None

   TYPE 0x7:  Message Drop Request:
              Additional Info: Message ID
              Control Info:
              1) 32 bits: First sequence number in the message
              2) 32 bits: Last sequence number in the message

   TYPE 0x7FFF: Explained by bits 16 - 31, reserved for user defined
              Control Packet

   Finally, Time Stamp and Destination Socket ID also exist in the
   control packets.

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 UDP Multiplexer

   A UDP multiplexer is used to handle concurrent UDT connections
   sharing the same UDP port. The multiplexer dispatch incoming UDT
   packets to the corresponding UDT sockets according to the destination
   socket ID in the packet header.

   One multiplexer is used for all UDT connections bound to the same UDP
   port. That is, UDT sockets on different UDP port will be handled by
   different multiplexers.

   A multiplexer maintains two queues. The sending queue includes the
   sockets with at least one packet scheduled for sending. The UDT
   sockets in the sending queue are ordered by the next packet sending
   time. A high performance timer is maintained by the sending queue and
   when it is time for the first socket in the queue to send its packet,
   the packet will be sent and the socket will be removed. If there are
   more packets for that socket to be sent, the socket will be re-
   inserted to the queue.

   The receiving queue reads incoming packets and dispatches them to the
   corresponding sockets. If the destination ID is 0, the packet will be
   sent to the listening socket (if there is any), or to a socket that
   is in rendezvous connection phase. (See Section 5.)

   Similar to the sending queue, the receiving queue also maintains a
   list of sockets waiting for incoming packets. The receiving queue
   scans the list to check if any timer expires for each socket every
   SYN (SYN = 0.01 second, defined in Section 4).


   UDT uses four timers to trigger different periodical events. Each
   event has its own period and they are all independent. They use the
   system time as origins and should process wrapping if the system time

   For a certain periodical event E in UDT, suppose the time variable is
   ET and its period is p. If E is set or reset at system time t0 (ET =
   t0), then at any time t1, (t1 - ET >= p) is the condition to check if
   E should be triggered.

   The four timers are ACK, NAK, EXP and SND. SND is used in the sender
   only for rate-based packet sending (see Section 6.1), whereas the
   other three are used in the receiver only.

   ACK is used to trigger an acknowledgement (ACK). Its period is set by
   the congestion control module. However, UDT will send an ACK no

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   longer than every 0.01 second, even though the congestion control
   does not need timer-based ACK. Here, 0.01 second is defined as the
   SYN time, or synchronization time, and it affects many of the other
   timers used in UDT.

   NAK is used to trigger a negative acknowledgement (NAK). Its period
   is dynamically updated to 4 * RTT_+ RTTVar + SYN, where RTTVar is the
   variance of RTT samples.

   EXP is used to trigger data packets retransmission and maintain
   connection status. Its period is dynamically updated to N * (4 * RTT
   + RTTVar + SYN), where N is the number of continuous timeouts. To
   avoid unnecessary timeout, a minimum threshold (e.g., 0.5 second)
   should be used in the implementation.

   The recommended granularity of their periods is microseconds.
   However, accurate time keeping is not necessary, except for SND.

   In the rest of this document, a name of a time variable will be used
   to represent the associated event, the variable itself, or the value
   of its period, depending on the context. For example, ACK can mean
   either the ACK event or the value of ACK period.

 Connection Setup and shutdown

   UDT supports two different connection setup methods, the traditional
   client/server mode and the rendezvous mode. In the latter mode, both
   UDT sockets connect to each other at (approximately) the same time.

   The UDT client (in rendezvous mode, both peer are clients) sends a
   handshake request (type 0 control packet) to the server or the peer
   side. The handshake packet has the following information (suppose UDT
   socket A sends this handshake to B):
     1) UDT version: this value is for compatibility purpose. The
        current version is 4.
     2) Socket Type: STREAM (0) or DGRAM (1).
     3) Initial Sequence Number: It is the sequence number for the first
        data packet that A will send out. This should be a random value.
     4) Packet Size: the maximum size of a data packet (including all
        headers). This is usually the value of MTU.
     5) Maximum Flow Window Size: This value may not be necessary;
        however, it is needed in the current reference implementation.
     6) Connection Type. This information is used to differential the
        connection setup modes and request/response.
     7) Socket ID. The client UDT socket ID.
     8) Cookie. This is a cookie value used to avoid SYN flooding attack
     9) Peer IP address: B's IP address.

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  Client/Server Connection Setup

   One UDT entity starts first as the server (listener). The server
   accepts and processes incoming connection request, and creates new
   UDT socket for each new connection.

   A client that wants to connect to the server will send a handshake
   packet first. The client should keep on sending the handshake packet
   every constant interval until it receives a response handshake from
   the server or a timeout timer expires.

   When the server first receives the connection request from a client,
   it generates a cookie value according to the client address and a
   secret key and sends it back to the client. The client must then send
   back the same cookie to the server.

   The server, when receiving a handshake packet and the correct cookie,
   compares the packet size and maximum window size with its own values
   and set its own values as the smaller ones. The result values are
   also sent back to the client by a response handshake packet, together
   with the server's version and initial sequence number. The server is
   ready for sending/receiving data right after this step is finished.
   However, it must send back response packet as long as it receives any
   further handshakes from the same client.

   The client can start sending/receiving data once it gets a response
   handshake packet from the server. Further response handshake
   messages, if received any, should be omitted.

   The connection type from the client should be set to 1 and the
   response from the server should be set to -1.

   The client should also check if the response is from the server that
   the original request was sent to.

  Rendezvous Connection Setup

   In this mode, both clients send a connect request to each other at
   the same time. The initial connection type is set to 0. Once a peer
   receives a connection request, it sends back a response. If the
   connection type is 0, then the response sends back -1; if the
   connection type is -1, then the response sends back -2; No response
   will be sent for -2 request.

   The rendezvous peer does the same check on the handshake messages
   (version, packet size, window size, etc.) as described in Section
   5.1. In addition, the peer only process the connection request from
   the address it has sent a connection request to. Finally, rendezvous
   connection should be rejected by a regular UDT server (listener).

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   A peer initializes the connection when it receives -1 response.

   The rendezvous connection setup is useful when both peers are behind
   firewalls. It can also provide better security and usability when a
   listening server is not desirable.


   If one of the connected UDT entities is being closed, it will send a
   shutdown message to the peer side. The peer side, after received this
   message, will also be closed. This shutdown message, delivered using
   UDP, is only sent once and not guaranteed to be received. If the
   message is not received, the peer side will be closed after 16
   continuous EXP timeout (see section 3.5). However, the total timeout
   value should be between a minimum threshold and a maximum threshold.
   In our reference implementation, we use 3 seconds and 30 seconds,

 Data Sending and Receiving

   Each UDT entity has two logical parts: the sender and the receiver.
   The sender sends (and retransmits) application data according to the
   flow control and congestion control. The receiver receives both data
   packets and control packets, and sends out control packets according
   to the received packets and the timers.

   The receiver is responsible for triggering and processing all control
   events, including congestion control and reliability control, and
   their related mechanisms.

   UDT always tries to pack application data into fixed size packets
   (the maximum packet size negotiated during connection setup), unless
   there is not enough data to be sent.

   We explained the rationale of some of the UDT data sending/receiving
   schemes in [GHG04b].

  The Sender's Algorithm

   Data Structures and Variables:

   1) Sender's Loss List: The sender's loss list is used to store the
      sequence numbers of the lost packets fed back by the receiver
      through NAK packets or inserted in a timeout event. The numbers
      are stored in increasing order.

   Data Sending Algorithm:
   1) If the sender's loss list is not empty, retransmit the first

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      packet in the list and remove it from the list. Go to 5).
   2) In messaging mode, if the packets has been the loss list for a
      time more than the application specified TTL (time-to-live), send
      a message drop request and remove all related packets from the
      loss list. Go to 1).
   3) Wait until there is application data to be sent.
        a. If the number of unacknowledged packets exceeds the
           flow/congestion window size, wait until an ACK comes. Go to
        b. Pack a new data packet and send it out.
   5) If the sequence number of the current packet is 16n, where n is an
      integer, go to 2).
   6) Wait (SND - t) time, where SND is the inter-packet interval
      updated by congestion control and t is the total time used by step
      1 to step 5. Go to 1).

  The Receiver's Algorithm

   Data Structures and Variables:

   1) Receiver's Loss List: It is a list of tuples whose values include:
      the sequence numbers of detected lost data packets, the latest
      feedback time of each tuple, and a parameter k that is the number
      of times each one has been fed back in NAK. Values are stored in
      the increasing order of packet sequence numbers.
   2) ACK History Window: A circular array of each sent ACK and the time
      it is sent out. The most recent value will overwrite the oldest
      one if no more free space in the array.
   3) PKT History Window: A circular array that records the arrival time
      of each data packet.
   4) Packet Pair Window: A circular array that records the time
      interval between each probing packet pair.
   5) LRSN: A variable to record the largest received data packet
      sequence number. LRSN is initialized to the initial sequence
      number minus 1.
   6) ExpCount: A variable to record number of continuous EXP time-out

   Data Receiving Algorithm:
   1) Query the system time to check if ACK, NAK, or EXP timer has
      expired. If there is any, process the event (as described below
      in this section) and reset the associated time variables. For
      ACK, also check the ACK packet interval.
   2) Start time bounded UDP receiving. If no packet arrives, go to 1).
   1) Reset the ExpCount to 1. If there is no unacknowledged data
      packet, or if this is an ACK or NAK control packet, reset the EXP
   3) Check the flag bit of the packet header. If it is a control

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      packet, process it according to its type and go to 1).
   4) If the sequence number of the current data packet is 16n + 1,
      where n is an integer, record the time interval between this
      packet and the last data packet in the Packet Pair Window.
   5) Record the packet arrival time in PKT History Window.
        a. If the sequence number of the current data packet is greater
           than LRSN + 1, put all the sequence numbers between (but
           excluding) these two values into the receiver's loss list and
           send them to the sender in an NAK packet.
        b. If the sequence number is less than LRSN, remove it from the
           receiver's loss list.
   7) Update LRSN. Go to 1).

   ACK Event Processing:
   1) Find the sequence number prior to which all the packets have been
      received by the receiver (ACK number) according to the following
      rule: if the receiver's loss list is empty, the ACK number is LRSN
      + 1; otherwise it is the smallest sequence number in the
      receiver's loss list.
   2) If (a) the ACK number equals to the largest ACK number ever
      acknowledged by ACK2, or (b) it is equal to the ACK number in the
      last ACK and the time interval between this two ACK packets is
      less than 2 RTTs, stop (do not send this ACK).
   3) Assign this ACK a unique increasing ACK sequence number. Pack the
      ACK packet with RTT, RTT Variance, and flow window size (available
      receiver buffer size). If this ACK is not triggered by ACK timers,
      send out this ACK and stop.
   4) Calculate the packet arrival speed according to the following
         Calculate the median value of the last 16 packet arrival
         intervals (AI) using the values stored in PKT History Window.
         In these 16 values, remove those either greater than AI*8 or
         less than AI/8. If more than 8 values are left, calculate the
         average of the left values AI', and the packet arrival speed is
         1/AI' (number of packets per second). Otherwise, return 0.
   5) Calculate the estimated link capacity according to the following
         Calculate the median value of the last 16 packet pair
         intervals (PI) using the values in Packet Pair Window, and the
         link capacity is 1/PI (number of packets per second).
   6) Pack the packet arrival speed and estimated link capacity into the
      ACK packet and send it out.
   7) Record the ACK sequence number, ACK number and the departure time
      of this ACK in the ACK History Window.

   NAK Event Processing:
   Search the receiver's loss list, find out all those sequence numbers
   whose last feedback time is k*RTT before, where k is initialized as 2

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   and increased by 1 each time the number is fed back. Compress
   (according to section 6.4) and send these numbers back to the sender
   in an NAK packet.

   EXP Event Processing:
   1) Put all the unacknowledged packets into the sender's loss list.
   2) If (ExpCount > 16) and at least 3 seconds has elapsed since that
      last time when ExpCount is reset to 1, or, 3 minutes has elapsed,
      close the UDT connection and exit.
   3) If the sender's loss list is empty, send a keep-alive packet to
      the peer side.
   4) Increase ExpCount by 1.

   On ACK packet received:
   1) Update the largest acknowledged sequence number.
   2) Send back an ACK2 with the same ACK sequence number in this ACK.
   3) Update RTT and RTTVar.
   4) Update both ACK and NAK period to 4 * RTT + RTTVar + SYN.
   5) Update flow window size.
   6) If this is a Light ACK, stop.
   7) Update packet arrival rate: A = (A * 7 + a) / 8, where a is the
      value carried in the ACK.
   8) Update estimated link capacity: B = (B * 7 + b) / 8, where b is
      the value carried in the ACK.
   9) Update sender's buffer (by releasing the buffer that has been
   10) Update sender's loss list (by removing all those that has been

   On NAK packet received:
   1) Add all sequence numbers carried in the NAK into the sender's loss
   2) Update the SND period by rate control (see section 3.6).
   3) Reset the EXP time variable.

   On ACK2 packet received:
   1) Locate the related ACK in the ACK History Window according to the
      ACK sequence number in this ACK2.
   2) Update the largest ACK number ever been acknowledged.
   3) Calculate new rtt according to the ACK2 arrival time and the ACK
      departure time, and update the RTT value as: RTT = (RTT * 7 +
      rtt) / 8.
   4) Update RTTVar by: RTTVar = (RTTVar * 3 + abs(RTT - rtt)) / 4.
   5) Update both ACK and NAK period to 4 * RTT + RTTVar + SYN.

   On message drop request received:
   1) Tag all packets belong to the message in the receiver buffer so
      that they will not be read.
   2) Remove all corresponding packets in the receiver's loss list.

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   On Keep-alive packet received:
   Do nothing.

   On Handshake/Shutdown packet received:
   See Section 5.

   Flow Control

   The flow control window size is 16 initially.

   On ACK packet received:
   The flow window size is updated to the receiver's available buffer

  Loss Information Compression Scheme

   The loss information carried in an NAK packet is an array of 32-bit
   integers. If an integer in the array is a normal sequence number (1st
   bit is 0), it means that the packet with this sequence number is
   lost; if the 1st bit is 1, it means all the packets starting from
   (including) this number to (including) the next number in the array
   (whose 1st bit must be 0) are lost.

   For example, the following information carried in an NAK:
      0x00000002, 0x80000006, 0x0000000B, 0x0000000E
   means packets with sequence number 2, 6, 7, 8, 9, 10, 11, and 14 are

 Configurable Congestion Control (CCC)

   The congestion control in UDT is an open framework so that user-
   defined control algorithm can be easily implemented and switched.
   Particularly, the native control algorithm is also implemented by
   this framework.

   The user-defined algorithm may redefine several control routines to
   read and adjust several UDT parameters. The routines will be called
   when certain event occurs. For example, when an ACK is received, the
   control algorithm may increase the congestion window size.

  CCC Interface

   UDT allow users to access two congestion control parameters: the
   congestion window size and the inter-packet sending interval. Users
   may adjust these two parameters to realize window-based control,
   rate-based control, or a hybrid approach.

   In addition, the following parameters should also be exposed.

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   1) RTT
   2) Maximum Segment/Packet Size
   3) Estimated Bandwidth
   4) The latest packet sequence number that has been sent so far
   5) Packet arriving rate at the receiver side

   A UDT implementation may expose additional parameters as well. This
   information can be used in user-defined congestion control algorithms
   to adjust the packet sending rate.

   The following control events can be redefined via CCC (e.g., by a
   callback function).

   1) init: when the UDT socket is connected.
   2) close: when the UDT socket is closed.
   3) onACK: when ACK is received.
   4) onLOSS: when NACK is received.
   5) onTimeout: when timeout occurs.
   6) onPktSent: when a data packet is sent.
   7) onPktRecv: when a data packet is received.

   Users can also adjust the following parameters in the user-defined
   control algorithms.

   1) ACK interval: An ACK may be sent every fixed number of packets.
      User may define this interval. If this value is -1, then it means
      no ACK will be sent based on packet interval.
   2) ACK Timer: An ACK will also be sent every fixed time interval.
      This is mandatory in UDT. The maximum and default ACK time
      interval is SYN.
   3) RTO: UDT uses 4 * RTT + RTTVar to compute RTO. Users may redefine

   Detailed description and discussion of UDT/CCC can be found in

  UDT's Native Control Algorithm

   UDT has a native and default control algorithm, which will be used if
   no user-defined algorithm is implemented and configured. The native
   UDT algorithm should be implemented using CCC.

   UDT's native algorithm is a hybrid congestion control algorithm,
   hence it adjusts both the congestion window size and the inter-packet
   interval. The native algorithm uses timer-based ACK and the ACK
   interval is SYN.

   The initial congestion window size is 16 packets and the initial
   inter-packet interval is 0. The algorithm start with Slow Start phase

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   until the first ACK or NAK arrives.

   On ACK packet received:
   1) If the current status is in the slow start phase, set the
      congestion window size to the product of packet arrival rate and
      (RTT + SYN). Slow Start ends. Stop.
   2) Set the congestion window size (CWND) to: CWND = A * (RTT + SYN) +
   3) The number of sent packets to be increased in the next SYN period
      (inc) is calculated as:
         if (B <= C)
            inc = 1/PS;
            inc = max(10^(ceil(log10((B-C)*PS*8))) * Beta/PS, 1/PS);
      where B is the estimated link capacity and C is the current
      sending speed. All are counted as packets per second. PS is the
      fixed size of UDT packet counted in bytes. Beta is a constant
      value of 0.0000015.
   4) The SND period is updated as:
         SND = (SND * SYN) / (SND * inc + SYN).

   These four parameters are used in rate decrease, and their initial
   values are in the parentheses: AvgNAKNum (1), NAKCount (1),
   DecCount(1), LastDecSeq (initial sequence number - 1).

   We define a congestion period as the period between two NAKs in which
   the first biggest lost packet sequence number is greater than the
   LastDecSeq, which is the biggest sequence number when last time the
   packet sending rate is decreased.

   AvgNAKNum is the average number of NAKs in a congestion period.
   NAKCount is the current number of NAKs in the current period.

   On NAK packet received:
   1) If it is in slow start phase, set inter-packet interval to
      1/recvrate. Slow start ends. Stop.
   2) If this NAK starts a new congestion period, increase inter-packet
      interval (snd) to snd = snd * 1.125; Update AvgNAKNum, reset
      NAKCount to 1, and compute DecRandom to a random (average
      distribution) number between 1 and AvgNAKNum. Update LastDecSeq.
   3) If DecCount <= 5, and NAKCount == DecCount * DecRandom:
        a. Update SND period: SND = SND * 1.125;
        b. Increase DecCount by 1;
        c. Record the current largest sent sequence number (LastDecSeq).

   The native UDT control algorithm is designed for bulk data transfer
   over high BDP networks. [GHG04a]

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Security Considerations

   UDT's security mechanism is similar to that of TCP. Most of TCP's
   approach to counter security attack should also be implemented in

IANA Considerations

   This document has no actions for IANA.

Normative References

   [RFC768] J. Postel, User Datagram Protocol, Aug. 1980.

Informative References

   [RFC4987] W. Eddy, TCP SYN Flooding Attacks and Common Mitigations.

   [GG07] Yunhong Gu and Robert L. Grossman, UDT: UDP-based Data
      Transfer for High-Speed Wide Area Networks, Computer Networks
      (Elsevier). Volume 51, Issue 7. May 2007.

   [GG05] Yunhong Gu and Robert L. Grossman, Supporting Configurable
      Congestion Control in Data Transport Services, SC 2005, Nov 12 -
      18, Seattle, WA, USA.

   [GHG04b] Yunhong Gu, Xinwei Hong, and Robert L. Grossman, Experiences
      in Design and Implementation of a High Performance Transport
      Protocol, SC 2004, Nov 6 - 12, Pittsburgh, PA, USA.

   [GHG04a] Yunhong Gu, Xinwei Hong, and Robert L. Grossman, An Analysis
      of AIMD Algorithms with Decreasing Increases, First Workshop on
      Networks for Grid Applications (Gridnets 2004), Oct. 29, San Jose,
      CA, USA.

   [LM97] T. V. Lakshman and U. Madhow, The Performance of TCP/IP for
      Networks with High Bandwidth-Delay Products and Random Loss,
      IEEE/ACM Trans. on Networking, vol. 5 no 3, July 1997, pp. 336-

   [RFC5681] Allman, M., Paxson, V. and E. Blanton, TCP Congestion
      Control, September 2009.

   [RFC4960] R. Stewart, Ed. Stream Control Transmission Protocol.
      September 2007.

   [TS06] K. Tan, Jingmin Song, Qian Zhang, Murari Sridharan, A Compound
      TCP Approach for High-speed and Long Distance Networks, in IEEE
      Infocom, April 2006, Barcelona, Spain.

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                                 UDT                   April 12, 2010

   [UDT] UDT: UDP-based Data Transfer, URL http://udt.sf.net.

   [XHR04] Lisong Xu, Khaled Harfoush, and Injong Rhee, Binary Increase
      Congestion Control for Fast Long-Distance Networks, INFOCOM 2004.

Author's Addresses

   Yunhong Gu
   National Center for Data Mining
   University of Illinois at Chicago
   713 SEO, M/C 249, 851 S Morgan St
   Chicago, IL 60607, USA
   Phone: +1 (312) 413-9576
   Email: yunhong@lac.uic.edu

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