





Network Working Group                                          M. Pullen
Request for Comments: 4410                                       F. Zhao
Category: Experimental                                 George Mason Univ
                                                                D. Cohen
                                                        Sun Microsystems
                                                           February 2006


             Selectively Reliable Multicast Protocol (SRMP)

Status of This Memo

   This memo defines an Experimental Protocol for the Internet
   community.  It does not specify an Internet standard of any kind.
   Discussion and suggestions for improvement are requested.
   Distribution of this memo is unlimited.

Copyright Notice

   Copyright (C) The Internet Society (2006).

Abstract

   The Selectively Reliable Multicast Protocol (SRMP) is a transport
   protocol, intended to deliver a mix of reliable and best-effort
   messages in an any-to-any multicast environment, where the best-
   effort traffic occurs in significantly greater volume than the
   reliable traffic and therefore can carry sequence numbers of reliable
   messages for loss detection.  SRMP is intended for use in a
   distributed simulation application environment, where only the latest
   value of reliable transmission for any particular data identifier
   requires delivery.  SRMP has two sublayers: a bundling sublayer
   handling message aggregation and congestion control, and a
   Selectively Reliable Transport (SRT) sublayer.  Selection between
   reliable and best-effort messages is performed by the application.
















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Table of Contents

   1. Introduction ....................................................3
      1.1. Terminology ................................................3
   2. Protocol Description ............................................4
   3. Message Formats .................................................6
      3.1. Bundle Message Format: .....................................6
      3.2. Bundle Header Format .......................................7
      3.3. Feedback Message Format ....................................9
      3.4. SRT Mode 0 Header Format ..................................10
      3.5. SRT Mode 1 Header Format ..................................11
      3.6. SRT Mode 2 Header Format ..................................11
      3.7. SRT NACK Format ...........................................12
      3.8. User-Configurable Parameters ..............................13
   4. TFMCC Operation ................................................13
      4.1. TCP Rate Prediction Equation for TFMCC ....................13
      4.2. Bundling ..................................................13
      4.3. Congestion Control ........................................14
      4.4. Any-Source Multicast ......................................14
      4.5. Multiple Sources ..........................................14
      4.6. Bundle Size ...............................................15
      4.7. Data Rate Control .........................................15
      4.8. Mode 1 Loss Detection .....................................16
           4.8.1. Sending a Negative Acknowledgement .................16
      4.9. Unbundling ................................................17
      4.10. Heartbeat Bundle .........................................17
   5. SRT Operation ..................................................17
      5.1. Mode 0 Operation ..........................................18
           5.1.1. Sending Mode 0 Messages ............................18
           5.1.2. Receiving Mode 0 Messages ..........................18
      5.2. Mode 1 Operation ..........................................18
           5.2.1. Sending Mode 1 Data Messages .......................19
           5.2.2. Receiving Mode 1 Data Messages .....................19
           5.2.3. Sending a Negative Acknowledgement .................20
           5.2.4. Receiving a Negative Acknowledgement ...............21
      5.3. Mode 2 Operation ..........................................21
           5.3.1. Sending Mode 2 Data Messages .......................21
           5.3.2. Receiving Mode 2 Data Messages .....................22
           5.3.3. Sending a Positive Acknowledgement .................23
           5.3.4. Receiving a Positive Acknowledgement ...............23
   6. RFC 2357 Analysis ..............................................23
      6.1. Scalability ...............................................23
      6.2. Congestion ................................................24
   7. Security Considerations ........................................25
   8. List of Acronyms Used ..........................................26
   9. Contributions ..................................................27
   10. References ....................................................27




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1.  Introduction

   There is no viable generic approach to achieving reliable transport
   over multicast networks.  Existing successful approaches require that
   the transport protocol take advantage of special properties of the
   traffic in a way originally proposed by Cohen [10].  The protocol
   described here is applicable to real-time traffic containing a mix of
   two categories of messages: a small fraction requiring reliable
   delivery, mixed with a predominating flow of best-effort messages.
   This sort of traffic is associated with distributed virtual
   simulation (RFC 2502 [4]) and also with some forms of distributed
   multimedia conferencing.  These applications typically have some data
   that changes rarely, or not at all, so the best efficiency will be
   achieved by transmitting that data reliably (the external appearance
   of a simulated vehicle is an excellent example).  They also require
   real-time transmission of a best-effort stream (for example, the
   position and orientation of the vehicle).  There is no value to
   reliable transmission of this stream because typically new updates
   arrive faster than loss identification and retransmission could take
   place.  By piggy-backing the sequence number (SN) of the latest
   reliable transmission on each bundle of traffic, the reliable and
   best-effort traffic can co-exist synergistically.  This approach is
   implemented in the Selectively Reliable Multicast Protocol (SRMP).

   The IETF has conducted a successful working group on Reliable
   Multicast Transport (RMT) that has produced RFCs 2357 [6], 2887 [11],
   and 3450 through 3453 [12 - 15], which define building block
   protocols for reliable multicast.  Selectively reliable multicast is
   similar in spirit to these protocols and in fact uses one of them,
   TCP-Friendly Multicast Congestion Control (TFMCC).  This document
   provides the basis for specifying SRMP with TFMCC for use on an
   experimental basis.  Key requirements of the RMT process that is
   carried forward here are specified in RFC 2357 [6].  These generally
   relate to scalability and congestion control, and are addressed in
   section 6 of this document.

1.1.  Terminology

   In this document, the key words "MUST", "MUST NOT", "REQUIRED",
   "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY",
   and "OPTIONAL" are to be interpreted as described in RFC 2119 [1] and
   indicate requirement levels for compliant implementations.









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2.  Protocol Description

   The Selectively Reliable Multicast Protocol (SRMP) has two major
   components: Selectively Reliable Transport (SRT) and a "bundling
   sublayer" that implements TCP-Friendly Multicast Congestion Control
   (TFMCC), as proposed by Widmer and Handley [2], in order to meet the
   requirements of RFC 2357 [6] for congestion avoidance.

   SRMP is capable of reliable message delivery over multicast networks,
   when the messages to be delivered reliably represent a fraction of a
   larger, associated best-effort flow and only the latest reliable
   message must be delivered.  The basic strategy for SRMP is to trade
   as little network capacity as possible for reliability by buffering
   the most recently sent reliable message at each sender and piggy-
   backing its sequence number on associated best-effort messages.  For
   this purpose, three modes of sending are defined:

   o  Mode 0 messages.  These will be delivered best-effort; if lost, no
      retransmission will be done.

   o  Mode 1 messages.  When a Mode 1 message loss is detected, the
      receiver will send back a NACK to the sender, where SRMP will
      retransmit the latest reliable message from that sender.  Senders
      define data identifiers (dataIDs), allowing multiple reliable
      message streams to be supported.  Mode 1 messages may be up to
      131,071 bytes long; SRMP provides for segmentation and reassembly,
      but only for the latest Mode 1 message for any given
      <sourceAddress, multicastAddress, dataID>.

   o  Mode 2 messages.  Through Mode 2 messages, SRMP provides for a
      lightweight, reliable, connectionless peer-to-peer unicast
      transaction exchange between any two members of the multicast
      group.  This is a unicast message requiring positive
      acknowledgement (ACK).

      | Application   |
      -----------------       ----------
      |      SRT      |
      -----------------   ->     SRMP
      |Bundling(TFMCC)|
      -----------------       ----------
      |      UDP      |

   The bundling sublayer is transparent to the Selectively Reliable
   Transport (SRT) sublayer.  It implements congestion control both by
   dropping Mode 0 messages at the source when needed and by bundling
   multiple short messages that are presented by applications within a
   short time window.  It also performs NACK suppression.



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   A bundling sublayer data unit is called a bundle.  A bundle is made
   up of a bundle header and one or more Mode 0 and Mode 1 SRMP
   messages.  Retransmission of Mode 1 messages does not imply
   retransmission of the original bundle; the retransmitted message
   becomes part of a new bundle.

   The TFMCC layer's behavior follows the mechanism described by Widmer
   and Handley.  This is an equation-based multicast congestion control
   mechanism: in a multicast group, each receiver determines its loss
   rate with regard to the sender, and calculates a desired source
   sending rate based on an equation that models the steady-state
   sending rate of TCP.  A distributed feedback suppression mechanism
   restricts feedback to those receivers likely to report the lowest
   desired rates.  Congestion control is achieved by dropping best-
   effort (Mode 0) messages at random.  For example, in distributed
   simulation, Mode 0 messages are part of a stream of state updates for
   dynamic data such as geographic location; therefore, the application
   can continue to function (with lower fidelity) when they are dropped.

   As described by its authors, TFMCC's congestion control mechanism
   works as follows:

   o  Each receiver measures the loss event rate and its Round-Trip Time
      (RTT) to the sender.

   o  Each receiver then uses this information, together with an
      equation for TCP throughput, to derive a TCP-friendly sending
      rate.

   o  Through a distributed feedback suppression mechanism, only a
      subset of the receivers is allowed to give feedback to prevent a
      feedback implosion at the sender.  The feedback mechanism ensures
      that receivers reporting a low desired transmission rate have a
      high probability of sending feedback.

   o  Receivers whose feedback is not suppressed report the calculated
      transmission rate back to the sender in so-called receiver
      reports.  The receiver reports serve two purposes: they inform the
      sender about the appropriate transmit rate, and they allow the
      receivers to measure their RTT.

   o  The sender selects the receiver that reports the lowest rate as
      the current limiting receiver (CLR).  Whenever feedback with an
      even lower rate reaches the sender, the corresponding receiver
      becomes the CLR and the sending rate is reduced to match that
      receiver's calculated rate.  The sending rate increases when the
      CLR reports a calculated rate higher than the current sending
      rate.



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   TFMCC was intended for fixed-size packets with variable rate.  SRMP
   applies it to variable-size SRMP messages that are mostly the same
   size because the best-effort updates typically all represent the same
   sort of simulation information and are grouped into bundles of size
   just under one MTU during periods of heavy network activity.  Future
   developments in TFMCC for variable-size messages will be of high
   value for inclusion in SRMP if, as expected, they prove to be
   appropriate for the types of traffic SRMP is intended to support.

   SRMP is intended for general use under applications that need its
   services and may exist in parallel instances on the same host.  The
   UDP port is therefore established ad hoc from available application
   ports; accordingly, it would not be appropriate to have a well-known
   port for SRMP.

3.  Message Formats

3.1.  Bundle Message Format:

   --------------------------------------------------------------------
   | bundle header | SRT Message 0 | SRT message 1 | SRT message 2 |...
   --------------------------------------------------------------------

   A bundle is an aggregation of multiple SRMP messages destined for the
   same multicast address.  A bundle can contain only Mode 0 and Mode 1
   messages; Mode 2 messages are exchanged using unicast addresses.

   SRMP identifies the sender and receiver using their 32-bit Sender_ID,
   which may be an IPv4 address.  For use with IPv6, a user group will
   need to establish a unique identifier per host.  There is no
   requirement for this identifier to be unique in the Internet; it only
   needs to be unique in the communicating group.



















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3.2.  Bundle Header Format

      0              8              16             24             32
      +--------------+--------------+--------------+--------------+
      |Version| Type |fb_nr | flag  |        bundle_SN            |
      +--------------+--------------+--------------+--------------+
      |                       Sender_ID                           |
      +--------------+--------------+--------------+--------------+
      |                       Receiver_ID                         |
      +--------------+--------------+--------------+--------------+
      |       Sender_Timestamp      |    Receiver_Timestamp       |
      +--------------+--------------+--------------+--------------+
      |            x_supp           |            R_max            |
      +--------------+--------------+--------------+--------------+
      |  DSN_count   |   padding    |           Length            |
      +--------------+--------------+--------------+--------------+
      |     0 to 255 DSN: <dataID, SN, NoSegs> of this sender     |
      +-----------------------------------------------------------+

   Version:
      4 bits   currently 0010

   Type:
      4 bits   0000 - indicates bundle

   fb_nr:
      4 bits   feedback round, range 0-15

   flag:
      4 bits   0001 Is_CLR
               other bits reserved

   bundle_SN:
      16 bits   range 0-65535

   Sender_Timestamp:
      16 bits   Representing the time that the bundle was sent out (in
                milliseconds) based on the sender's local clock.

   Receiver_Timestamp:
      16 bits   Echo of the Receiver_Time_Stamp field (in milliseconds)
                of the receiver feedback message.  If the sender has
                time delay between receiving the feedback and echoing
                the timestamp, it MUST adjust the Receiver_Timestamp
                value to compensate.






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   Receiver_ID
      32 bits   Unique identifier for the receiver within the multicast
                group.  IPv4 addresses may be used.

   Sender_ID:
      32 bits   Unique identifier for the sender within the multicast
                group.  IPv4 addresses may be used.

   X_supp:
      16 bits   The suppression rate corresponding to the sender, in
                bits/s.  Only those receivers whose desired rate is less
                than the suppression rate, or whose RTT is larger than
                R_max, may send feedback information to the sender.  The
                suppression rate is represented as a 16-bit floating
                point value with 8 bits for the unsigned exponent and 8
                bits for the unsigned mantissa.

   R_max:
      16 bits   The maximum of the RTTs of all receivers, in
                milliseconds.  The Maximum RTT should be represented as
                a 16-bit floating point value with 8 bits for the
                unsigned exponent and 8 bits for the unsigned mantissa.

   DSN_count:
      8 bits    The count of DSN blocks following the header.

   Length:
      16 bits   Range from 0~65535.  The total length of the bundle
                in octets (including the header).

   DSN:
      32 bits   There can be up to 256 of these in a header.  An SRMP
                implementation MUST support a minimum of 1.  Each DSN
                consists of three fields:

      dataID:
         16 bits   A unique number associated with a particular data
                   element on the sending host, used to identify a
                   Mode 1 message.
      SN:
         9 bits    Sequence number associated with a particular Mode 1
                   transmission of a particular dataID.
      NoSegs:
         7 bits    Number of segments, if the dataID was long enough
                   to require segmentation; otherwise 0x0.






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   Note that the number of DSNs reflects the number of different Mode 1
   DataIDs being supported at this time by this instance of SRMP, and is
   not the count of SRMP messages bundled in this transmission.

   Note also that 16-bit timestamps will wrap around in 65536
   milliseconds.  This should not be a problem unless an RTT is greater
   than 65 seconds. If a timestamp is less than its predecessor
   (treating the 16 bits as an unsigned integer), its value must be
   increased by 65536 for comparisons against the predecessor.

3.3.  Feedback Message Format

      0              8              16             24             32
      +--------------+--------------+--------------+--------------+
      |Version| Type | fb_nr| flag  |             X_r             |
      +--------------+--------------+--------------+--------------+
      |       Sender_Timestamp      |    Receiver_Timestamp       |
      +--------------+--------------+--------------+--------------+
      |                       Sender_ID                           |
      +--------------+--------------+--------------+--------------+
      |                      Receiver_ID                          |
      +--------------+--------------+--------------+--------------+

   Version:
      4 bits   currently 0010

   Type:
      4 bits   value 0001

   fb_nr:
      4 bits   current feedback round of the sender

   flag:
      4 bits
         0001 - have_RTT
         0010 - have_loss
         0100 - receiver_leave
         other values reserved

   X_r:
      16 bits   desired sending rate X_r in bits/s, calculated by the
                receiver to be TCP-friendly, 16 bit floating point
                value with 8 bits for the unsigned exponent and 8 bits
                for the unsigned mantissa.







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   Sender_Timestamp:
      16 bits   Echo of the Sender_Timestamp in bundle header.  If the
                receiver has time delay between receiving the bundle and
                echoing the timestamp, it MUST adjust the
                Sender_Timestamp value correspondently.

   Receiver_Timestamp:
      16 bits   The time when the feedback message was sent out from the
                receiver.

   Receiver_ID:
      32 bits   Unique identifier for the receiver within the multicast
                group.  IPv4 addresses may be used.  (Identifies the
                receiver that sends the feedback message).

   Sender_ID:
      32 bits   Unique identifier for the sender within the multicast
                group.  IPv4 addresses may be used.  (Identifies the
                sender that is the destination of the current feedback
                message.)

3.4.  SRT Mode 0 Header Format

      0              8              16             24             32
      +--------------+--------------+--------------+--------------+
      |Version| Type | 000 |  00000000  |        Length           |
      +--------------+--------------+--------------+--------------+

   Version:
      4 bits   currently 0010

   Type:
      4 bits   0000

   Mode:
      3 bits   000

   Padding:
      8 bits   00000000

   Length:
      11 bits  Length of the payload data in octets (does not include
               the header).








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3.5.  SRT Mode 1 Header Format

      0              8              16             24             32
      +--------------+--------------+--------------+--------------+
      |Version| Type | 001 |  SegNo    |            Length        |
      +--------------+--------------+--------------+--------------+
      |                            DSN                            |
      +--------------+--------------+--------------+--------------+

   Version:
      4 bits   currently 0010

   Type:
      4 bits   0000

   Mode:
      3 bits   001

   SegNo:
      7 bits   The index number of this segment.

   Length:
      14 bits   Length of the payload data in octets (does not include
                the header).

   DSN:
      32 bits   Same as in the bundle header.  Note that this contains
                NoSegs, whereas SegNo is a separate element.

3.6.  SRT Mode 2 Header Format

      0              8              16             24             32
      +--------------+--------------+--------------+--------------+
      |Version| Type |010 |  00000  |            Length           |
      +--------------+--------------+--------------+--------------+
      |                            SN                             |
      +--------------+--------------+--------------+--------------+

   Version:
      4 bits   currently 0010

   Type:
      4 bits   0010

   Mode:
      3 bits   010





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   Padding:
      5 bits   00000

   Length:
      16 bits  Length of the payload data in octets (does not the
               include header).

   SN:
      32 bits   Same as in bundle header.

3.7.  SRT NACK Format

      0              8              16             24             32
      +--------------+--------------+--------------+--------------+
      |Version| Type |111 |  00000  |          reserved           |
      +--------------+--------------+--------------+--------------+
      |                            DSN                            |
      +--------------+--------------+--------------+--------------+
      |                      Sender Address                       |
      +--------------+--------------+--------------+--------------+

   Version:
      4 bits   currently 0010

   Type:
      4 bits   0010

   Mode:
      3 bits   111

   Padding:
      5 bits   00000

   Reserved:
      16 bits

   DSN:
      32 bits  sequence number

   Sender Address:
      The IP address of the sender of the message being NACKed.










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3.8.  User-Configurable Parameters

   Name                 Minimum Value   Recommended Value       Units

   DSN_Max                 1                 32                messages
   dataID_Timeout         none              none                 ms
   Segment_Timeout         50                250                 ms
   Bundle_Timeout          1                 10                  ms
   Heartbeat_Interval      1                none                 s
   Mode2_Max               1                none               messages
   ACK_Threshold          none         worst RTT in group        ms

4.  TFMCC Operation

4.1.  TCP Rate Prediction Equation for TFMCC

   The RECOMMENDED throughput equation for SRMP is a slightly simplified
   version of the throughput equation for Reno TCP from [5]:

                                      8*s
      X = ------------------------------------------------------   (1)
            R * (sqrt(2*p/3) + (3*sqrt(6*p) * p * (1+32*p^2)))

   (the formula may be simplified for implementation), where

      X is the transmit rate in bits/second.

      s is the message size in bytes.

      R is the round-trip time in seconds.

      p is the loss event rate, between 0.0 and 1.0, of the number of
        loss events as a fraction of the number of messages transmitted.

   In the future, different TCP formulas may be substituted for this
   equation.  The requirement is that the throughput equation be a
   reasonable approximation of the sending rate of TCP for conformant
   TCP congestion control.

4.2.  Bundling

   Multiple SRMP messages will be encapsulated into a bundle.  When a
   new SRMP message (Mode 0 or Mode 1) arrives, the SRMP daemon will try
   to add the new message into the current bundle.

   The SRMP daemon MUST keep a timer, which will be reset when the first
   SRMP message is added into the bundle.  After Bundle_Timeout, the
   timer will time out, and the current bundle should be transmitted



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   immediately.  A new bundle will then be initialized to hold new SRMP
   messages.  Bundle_Timeout SHALL NOT be less than 1 ms.  The
   recommended value is 10 ms.

   Also, the bundle length MUST NOT exceed LENGTH_MAX.  If adding a new
   SRMP message will produce a greater length, the SRMP daemon MUST
   initialize a new bundle for the new SRMP messages, and the current
   bundle should be transmitted immediately.  The recommended value for
   LENGTH_MAX is 1454 bytes (Ethernet MTU minus IP and UDP header
   lengths).

   In a bundle, there may exist multiple SRMP messages with the same
   dataID.  In this case, only the latest version of that dataID is
   useful.  SRMP may check for duplicate dataIDs in the same bundle and
   delete all but the latest one.  If a Mode 1 message appears in the
   outgoing bundle, then the corresponding DSN should not appear in the
   bundle header.

   The bundle header contains the DSN <dataID,SN,NoSegs> for Mode 1
   messages from this sender.  The absolute maximum number of DSN is
   255; however, an implementation may apply a user-specified DSN_Max,
   no smaller than 1.  An implementation may support a user-defined
   dataID_Timeout, after which a given dataID will not be announced in
   the bundle header unless a new Mode 1 message has been sent.  If the
   sender has more dataIDs sent (and not timed out) than will fit in the
   bundle header, the DSNs MUST be announced on a round-robin basis,
   with the exception that no bundle header will announce a DSN for a
   Mode 1 message contained within that bundle.  If a duplicate DSN is
   received, it may be silently discarded.

4.3.  Congestion Control

   The congestion control mechanism operates as described in [7].

4.4.  Any-Source Multicast

   SRMP uses the Any-Source Multicast Mode.  Each sender will determine
   its maximum RTT, suppression data rate, and sending rate with respect
   to each sender.  Each receiver will measure its RTT and desired rate
   to each sender in the group, and send feedback to every sender by
   sending to the multicast group.

4.5.  Multiple Sources

   Under SRMP, each group member in a multicast group is a sender as
   well as a receiver.  Each receiver may need to participate in TFMCC
   information exchange with all senders.  Thus, when a receiver sends a




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   feedback message, it must identify to which source the message should
   be sent using the "Sender ID" field in the header.

   The feedback is multicast to the group.  Depending on the network
   situation, senders may select different receivers to provide
   feedback.  Feedback messages from receivers that are not among those
   selected by the local TFMCC to provide feedback should be silently
   discarded.

4.6.  Bundle Size

   TFMCC is designed for traffic with a fixed message size.  The maximum
   bundle size (including header) for SRMP is set to a configurable
   maximum, typically 1454 bytes (Ethernet MTU minus IP and UDP header
   lengths).  The bundle size will be used in a TCP throughput equation,
   to get a desired source rate.  However, in SRMP, the message size is
   variable because:

   1. After bundle time out, the current bundle will not wait for new
      SRMP messages.  This happens with sources sending at a slow rate.

   2. In long messages, there is no further space in the current bundle
      for new SRMP messages.  This will happen with sources sending at a
      high rate or sending messages with a length over half of the
      bundle payload size.

   The case 1 bundle size is likely to be much smaller than that of case
   2.

   Therefore, in SRMP, the mean value of the 10 most recent bundles'
   sizes will be used as the bundle size in the TCP throughput equation.
   This mean value is independent from the network condition and
   reflects current activity of the source.

4.7.  Data Rate Control

   Each host will have a single instance of SRMP supporting all of its
   applications.  Thus, the sender's source rate is the sum of the rates
   of all the clients of the same multicast group.

   If the source rate is larger than the sender's desired transmission
   rate, it is the sender's responsibility to do traffic shaping.  Any
   method that conforms to the target sending rate may be used.  The
   RECOMMENDED method is to randomly discard enough Mode 0 messages to
   meet the target rate.






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4.8.  Mode 1 Loss Detection

   Bundle header processing includes checking each DSN in the bundle
   header and scheduling a NACK for each DSN bearing a dataID for which
   some application has indicated interest, if the SN/SegNo in that DSN
   indicates that a NACK is needed.  NACKs are sent in bundles and may
   be bundled with data messages.  A NACK is required if:

   o  the SN is one or more greater (mod 512) than the latest received
      Mode 1 message for that dataID, or

   o  the SegNo has not been received, some segment of the <dataID,SN>
      has been received, and a user-defined Segment_Timeout, which SHALL
      NOT be less than 50 ms, has expired since receipt of the first
      SegNo for the <dataID,SN>.

   The bundling sublayer will pass the DSN list in any received bundle
   header to the SRT sublayer.  It also will suppress NACKs in outgoing
   bundles, as described in the next section.

4.8.1.  Sending a Negative Acknowledgement

   Negative acknowledgements are used by SRMP for multicast messages in
   order to avoid the congestion of an "ACK implosion" at the original
   sender that would likely occur if positive acknowledgements were used
   instead.  However, with a large multicast group spread out over a
   congested wide-area network, there is the potential for enough
   members of the multicast group to fail to receive the message and
   generate NACKs to cause considerable congestion at the original
   sender despite the use of negative acknowledgements instead of
   positive acknowledgements.  For this reason, SRMP uses a NACK
   suppression mechanism to reduce the number of NACKs generated in
   response to any single lost message.

   The NACK suppression mechanism uses the Bundle_Timeout to distribute
   NACKs over an appropriate time window.  This assumes that the user
   has selected a bundle timeout appropriate for the needs of the
   application for real-time responsiveness.

   When the bundling sublayer is ready to send a bundle, it removes from
   the bundle any NACKs for which a response has been sent by another
   member of the multicast group within the NACK_Repeat_Timeout window.
   If the original Bundle_Timeout has not expired, transmission of the
   bundle may then be delayed until the original Bundle_Timeout expires
   or the bundle is full, whichever happens first.






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4.9.  Unbundling

   After a receiver completes congestion control processing on a bundle,
   it parses the bundle into SRT messages and sends these to the SRT
   sublayer.

4.10.  Heartbeat Bundle

   SRMP implementations may support a user-defined Heartbeat_Interval,
   which SHALL NOT be less than one second.  At the end of each
   heartbeat interval, if the sender has not sent any bundle, an empty
   bundle will be sent in order to trigger Mode 1 loss detection.

5.  SRT Operation

   SRMP operates in three distinct transmission modes in order to
   deliver varying levels of reliability: Mode 0 for multicast data that
   does not require reliable transmission, Mode 1 for data that must be
   received reliably by all members of a multicast group, and Mode 2 for
   data that must be received reliably by a single dynamically
   determined member of a multicast group.

   Mode 0 operates as a pure best-effort service.  Mode 1 operates with
   negative acknowledgements only, triggered by bundle arrivals that
   indicate loss of a Mode 1 message.  Mode 2 uses a positive
   acknowledgement for each message to provide reliability and low
   latency.  Mode 2 is used where a transaction between two members of a
   multicast group is needed.  Because there can be many members in such
   a group, use of a transaction protocol, with reliability achieved by
   SRMP retransmission, avoids the potentially large amount of
   connection setup and associated state that would be required if each
   pair of hosts in the group established a separate TCP connection.

   Use of SRMP anticipates that only a small fraction of messages will
   require reliable multicast, and a comparably small fraction will
   require reliable unicast.  This is due to a property of distributed
   virtual simulation: the preponderance of messages consist of state
   update streams for object attributes such as position and
   orientation.  SRMP is unlikely to provide effective reliable
   multicast if the traffic does not have this property.

   In SRMP, "dataID" is used to associate related messages with each
   other.  Typically, all messages with the same dataID are associated
   with the same application entity.  All the messages with the same
   dataID must be transmitted in the same mode.  Among all the messages
   with the same dataID, the latest version  will obsolete all older
   messages.




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5.1.  Mode 0 Operation

   Mode 0 is for multicast messages that do not require reliable
   transmission because they are part of a real-time stream of data that
   is periodically updated with high frequency.  Any such message is
   very likely to have been superceded by a more recent update before
   retransmission could be completed.

5.1.1.  Sending Mode 0 Messages

   When an application requests transmission of Mode 0 data, a
   destination multicast group must be provided to SRMP along with the
   data to be sent.  After verifying the data length and multicast
   group, the following steps MUST be performed by the SRT sublayer:

   1. An SRT message MUST be generated with the following
      characteristics:

      the version is set to the current version, the message type is set
      to 0x0, the mode is set to 0x0.  User data is included after the
      message header.  If the message cannot be generated as described
      above, the user data is discarded and the error MUST be reported
      to the application.

   2. If step 1 was completed without error, the newly generated message
      MUST be sent to the bundling sublayer.  The implementation MUST
      report to the application whether the message was ultimately
      accepted by UDP.

5.1.2.  Receiving Mode 0 Messages

   When a Mode 0 message is received by SRMP, it MUST be processed as
   follows: after verifying the version, message type, and destination
   multicast address fields, the user data MUST be delivered to all
   applications that are associated with the multicast group in the
   message.  If the SRMP receiver has never received any Mode 1 messages
   before the Mode 0 message is received, the Mode 0 message should be
   silently discarded.

   It is RECOMMENDED that the following information be provided to the
   receiving applications: message body, multicast address.

5.2.  Mode 1 Operation

   Mode 1 is for multicast data that requires reliable transmission.  A
   Mode 1 message can be either a data message or a NACK.  Mode 1 data
   messages are expected to be part of a data stream.  This data stream
   is likely to contain Mode 0 messages as well (see section 5.1.1), but



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   it is possible for a data stream to be comprised solely of Mode 1
   messages.

5.2.1.  Sending Mode 1 Data Messages

   After the data length, dataID, and destination multicast group are
   verified, SRT MUST take the following steps:

   1. If the message will not fit in an empty bundle with DSN_Max DSN in
      the header, the message MUST be segmented.  The remaining steps
      pertain to each segment of the message.  Each segment receives a
      unique SegNo, starting with 0 and ending with (NoSegs-1).

   2. An SRT message is generated with the following characteristics:
      the version is set to 0x02, the message type is set to 0x0, the
      transmission mode is set to 0x01, the SN is set equal to the SN of
      the most recently sent Mode 1 complete message of the same dataID,
      incremented by 1 modulo 512.  If no such Mode 1 message exists,
      the SN is set to 0x0.

   3. The newly generated message (all segments) must then be buffered,
      replacing any formerly buffered Mode 1 message of the same dataID,
      destination multicast address.  If the message cannot be buffered,
      the user data is discarded and the error is reported to the
      application.

   4. If step 2 was completed without error, the newly generated message
      is sent to the TFMCC sublayer.

5.2.2.  Receiving Mode 1 Data Messages

   When a Mode 1 data message is received by SRT, it will be processed
   as follows (assuming that the version field has already been verified
   to be 0x02):

   1. The destination address MUST be verified to be a valid IP
      multicast address on which this instance of SRMP is a member.  If
      this is not the case, the message should be silently discarded.

   2. The destination address MUST be verified to be one for which some
      application has indicated interest.  Otherwise, the message should
      be silently discarded.

   3. The SN, SegNo, source_ip_address, and the body of the received
      message MUST be buffered, and the user data MUST then be delivered
      to all applications that have indicated interest in the multicast
      group of the received message.




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   4. When a new DSN value is received with NoSegs greater than zero, a
      timer should be set for Segment_Timeout, after which a NACK should
      be sent to the bundling sublayer and the timer should be restarted
      for Segment_Timeout.

   5. If NoSegs in the received message is not 0, a reassembly process
      MUST be started.  Each segment MUST be buffered.  If receipt of
      the current message completes the segment, the reassembled message
      MUST be released to the application and the Segment_Timeout timer
      cancelled.

   6. If a new DSN is received before all segments of the previous DSN
      are received, the segments that have been received should be
      dropped silently.

   7. It is RECOMMENDED that the following information be provided to
      the receiving applications: message body, dataID,
      source_ip_address, multicast_group address.

   8. When a client signs on to a new multicast group, all locally
      buffered Mode 1 messages related to that multicast group should be
      delivered to the client immediately.

5.2.3.  Sending a Negative Acknowledgement

   Whenever a bundle is received, the bundling sublayer will forward the
   DSN list from the bundle header to the SRT sublayer.  The SRT
   sublayer will examine buffered values of <SenderID,dataID,SN,SegNo>
   to determine whether a NACK is required.  If so, it will generate a
   NACK message and send it to the bundling sublayer.  The NACK message
   will have version set to 0x2, message type set to 0x2, and
   transmission mode set to 0x7.  dataID, SN, and destination address
   are set to that of the Mode 1 message for which the NACK is being
   sent.  If a NACK has been received from any member of the destination
   multicast group for the Mode 1 message in question within the NACK
   threshold, no NACK is generated.

   For segmented messages, there are two possible types of NACKs:

   o  Based on the DSN list in the bundle header, the SRT implementation
      may determine that an entire segmented Mode 1 message was lost.
      In this case, the NACK MUST carry SegNo=0x7F (all in one field).

   o  Based on the Segment Timeout, the SRT implementation may determine
      that one or more segments of a message have not been delivered.
      In this case, a NACK will be sent for each missing segment.





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5.2.4.  Receiving a Negative Acknowledgement

   When a NACK is received by SRT, it MUST be processed as follows,
   after verifying the multicast address, dataID, source IP address, and
   transmission mode:

   1. If this instance of SRT's most recent Mode 1 message of the dataID
      indicated in the NACK has an SN newer than the SN in the NACK,
      that message (which is buffered) should be immediately
      retransmitted to the multicast address indicated in the received
      NACK.  If the most recent Mode 1 message has an SN equal to the SN
      indicated in the NACK, and if the SegNo field in the NACK contains
      0x7F, all segments of the buffered Mode 1 message MUST be
      retransmitted; if the SegNo has some other value, only the
      indicated segment should be retransmitted.

   2. Whether or not step 1 results in the retransmission of a message,
      the event of receiving the NACK and the (local machine) time at
      which the NACK was received should be buffered.  Each instance of
      SRT MUST buffer the number of NACKs that have been received for
      each dataID-multicast address pair, since the most recent Mode 1
      message of the same pair was received and the time at which the
      most recent of these NACKs was received.

5.3.  Mode 2 Operation

   Mode 2 is for infrequent reliable transaction-oriented communication
   between two dynamically determined members of a multicast group.  TCP
   could be used for such communication, but there would be unnecessary
   overhead and delay in establishing a stream-oriented connection for a
   single exchange of data, whereas there is already an ongoing stream
   of best-effort data between the hosts that require Mode 2
   transmission.  An example is a Distributed Interactive Simulation
   (DIS) collision PDU.

5.3.1.  Sending Mode 2 Data Messages

   When an application requests transmission of Mode 2 data, a dataID
   and a destination unicast IP address MUST be provided to SRT along
   with the data to be sent.  After verifying the data length, dataID,
   and destination address, SRT MUST perform the following steps:

   1. An SRT message is generated with the following characteristics:
      the version is set to 0x02, the message type is set to 0x02, the
      transmission mode is set to 0x2, the dataID is set to the
      application-provided value, and the destination address is set to
      the application-provided IP address.  The SN is set equal to the
      SN of the most recently sent Mode 2 message of the same dataID



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      incremented by 1 modulo 65536.  If no such Mode 1 message exists,
      it is set to 0x0.

   2. The newly generated message is buffered.  This new message does
      not replace any formerly buffered Mode 2 messages.  An
      implementation MUST provide a Mode 2 message buffer that can hold
      one or more Mode 2 messages. Mode 2 messages are expected to be
      infrequent (less than 1 percent of total traffic), but it is still
      strongly RECOMMENDED that an implementation provide a buffer of
      user-configurable size Mode2_Max that can hold more than a single
      Mode 2 message.  If the message cannot be buffered, the user data
      is discarded and the error MUST be reported to the application.
      If the message can be buffered, it should be sent to UDP
      immediately after being buffered.

   3. If step 2 was completed without error, the newly generated message
      MUST be sent to the IP address contained in its destination
      address field, encapsulated within a UDP datagram.  If the UDP
      interface on the sending system reports an error to SRT when the
      attempt to send the SRT message is made, an implementation may
      attempt to resend the message any finite number of times.
      However, every implementation MUST provide a mode in which no
      retries are attempted.  Implementations should default to this
      latter mode of operation.  The implementation MUST report to the
      application whether the message was ultimately accepted by UDP.

   4. If some user-configurable "ACK_Threshold" (which should be greater
      than the worst-case round-trip time for the multicast group)
      elapses without receipt of an ACK for the Mode 2 message, it is
      retransmitted.  An implementation may define a maximum number of
      retransmissions to be attempted before the Mode 2 message is
      removed from the buffer.

5.3.2.  Receiving Mode 2 Data Messages

   When a Mode 2 data message is received by SRT, it should be processed
   as follows after verifying version, dataID, sender address, and SN:

   1. For Mode 2 messages, the sequence number field is used to
      associate the required positive acknowledgement with a specific
      Mode 2 message.  If the message passes verification, the
      encapsulated user data is delivered to all applications that have
      indicated interest in the dataID and multicast address of the
      received message, regardless of the value of the SN field.

   2. Additionally, an ACK MUST be sent to the host from which the Mode
      2 data message originated.  See section 5.3.3. below for details.




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5.3.3.  Sending a Positive Acknowledgement

   A positive acknowledgement (ACK) is triggered by the receipt of a
   Mode 2 data message.  To send an ACK, a new SRT message is generated
   with version set to 0x02, message type set to 0x2, and transmission
   mode set to 0x2.  The dataID and SN are those of the Mode 2 data
   message being acknowledged.  The destination address field is set to
   the source IP address from which the data message was received.
   Since Mode 2 data messages are unicast, there is little concern about
   an ACK implosion causing excessive congestion at the original sender,
   so no suppression mechanism is necessary.

5.3.4.  Receiving a Positive Acknowledgement

   When an ACK is received by SRT, after verifying the transmission
   mode, dataID, and source IP address against outstanding Mode 2
   transmission, SRT MUST remove the pending transmission from its
   buffer.

6.  RFC 2357 Analysis

   This section provides answers to the questions posed by RFC 2357 for
   reliable multicast protocols, which are quoted.

6.1.  Scalability

   "How scalable is the protocol to the number of senders or receivers
   in a group, the number of groups, and wide dispersion of group
   members?"

   SRMP is intended to scale at least to hundreds of group members.  It
   has been designed not to impose limitations on the scalability of the
   underlying multicast network.  No problems have been identified in
   its mechanisms that would preclude this on uncongested networks.

   "Identify the mechanisms which limit scalability and estimate those
   limits."

   There is a practical concern with use of TFMCC, in that the receiver
   with the most congested path constrains delivery to the entire group.
   Distributed virtual simulation requires data delivery at rates
   perceived as continuous by humans.  Therefore, it may prove necessary
   to assign such receivers to different, lower-fidelity groups as a
   practical means of sustaining performance to the majority of
   participating hosts.  SRMP does not have a mechanism to support such
   pruning at this time.





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6.2.  Congestion

   "How does the protocol protect the Internet from congestion?  How
   well does it perform?  When does it fail?  Under what circumstances
   will the protocol fail to perform the functions needed by the
   applications it serves?  Is there a congestion control mechanism?
   How well does it perform?  When does it fail?"

   Both simulations and tests indicate that SRMP with TFMCC displays
   backoff comparable to that of TCP under conditions of significant
   packet loss.  The mechanism fails in a network-friendly way, in that
   under severe congestion, it reduces sending of the best-effort
   traffic to a very small rate that typically is unsatisfactory to
   support a virtual simulation.  This is possible because the reliable
   traffic typically is a small percentage of the overall traffic and
   SRMP is NACK oriented, with NACK suppression, so that reliable
   traffic loss adds little traffic to the total.  If the traffic mix
   assumption is not met, the reliable traffic (which does not back off
   under increased RTT) could produce a higher level of traffic than a
   comparable TCP connection.  However, levels of reliable traffic this
   large are not in the intended application domain of SRMP.

   "Include a description of trials and/or simulations which support the
   development of the protocol and the answers to the above questions."

   SRMP has been simulated using a discrete event simulator developed
   for academic use [8].  The design assumptions were validated by the
   results.  It also has been emulated in a LAN-based cluster and
   application-tested in a wide-area testbed under its intended traffic
   mix (distributed virtual simulation) and using a traffic generator
   with losses emulated by random dropping of packets [9].

   "Include an analysis of whether the protocol has congestion avoidance
   mechanisms strong enough to cope with deployment in the Global
   Internet, and if not, clearly document the circumstances in which
   congestion harm can occur.  How are these circumstances to be
   prevented?"

   Because it provides sending backoff comparable to TCP, SRMP is able
   to function as well as TCP for congestion avoidance, even in the
   Global Internet.  The only way an SRMP sender can generate congestion
   is to use the protocol for unintended purposes, for example, reliable
   transmission of a large fraction of the traffic.  Doing this would
   produce unsatisfactory results for the application, as SRMP's
   mechanism for providing reliability will not function well if the
   best-effort traffic does not constitute the majority of the total
   traffic.




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   "Include a description of any mechanisms which contain the traffic
   within limited network environments."

   SRMP has no such mechanisms, as it is intended for use over the open
   Internet.

   "Reliable multicast protocols must include an analysis of how they
   address a number of security and privacy concerns."

   See section 7 below.

7.  Security Considerations

   As a transport protocol, SRMP is subject to denial of service by
   hostile third parties sending conflicting values of its parameters on
   the multicast address.  SRMP could attempt to protect itself from
   this sort of behavior.  However, it can be shielded from such attacks
   by traffic authentication at the network layer, as described below.
   A comparable level of authentication also could be obtained by a
   message using MD5, or a similar message hash in each bundle, and
   using the SRMP bundle header to detect duplicate transmissions from a
   given host.  However, this would duplicate the function of existing
   network layer authentication protocols.

   Specific threats that can be eliminated by packet-level
   authentication are as follows:

   a. Amplification attack: SRMP receivers could be manipulated into
      sending large amounts of NACK traffic, which could cause network
      congestion or overwhelm the processing capabilities of a sender.
      This could be done by sending them faked traffic indicating that a
      reliable transmission has been lost.  SRMP's NACK suppression
      limits the effect of such manipulation.  However, true protection
      requires authentication of each bundle.

   b. Denial-of-service attack: If an SRMP sender accepts a large number
      of forged NACKs, it will flood the multicast group with repair
      messages.  This attack also is stopped by per-bundle
      authentication.

   c. Replay attack: The attacker could copy a valid, authenticated
      bundle containing a NACK and send it repeatedly to the original
      sender of the NACKed data.  Protection against this attack
      requires a sequence number per transmission per source host.  The
      SRMP bundle header sequence number would satisfy this need.
      However, the SN also can be applied at a lower layer.





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   d. Reverse path forwarding attack (spoofing): If checks are not
      enabled in all network routers and switches along the path from
      each sender to all receivers, forged packets can be injected into
      the multicast tree data path to manipulate the protocol into
      sending a large volume of repairs.  Packet-level authentication
      can eliminate this possibility.

   e. Inadvertent errors: A receiver with an incorrect or corrupted
      implementation of TFMCC could respond with values of RTT that
      might stimulate a TFMCC sender to create or increase congestion in
      the path to that sender.  It is therefore RECOMMENDED that
      receivers be required to identify themselves as legitimate before
      they receive the Session Description needed to join the session.
      How receivers identify themselves as legitimate is outside the
      scope of this document.

   The required authentication could become part of SRMP or could be
   accomplished by a lower layer protocol.  In any case, it needs to be
   (1) scalable and (2) not very computationally demanding so it can be
   performed with minimal delay on a real-time virtual simulation
   stream.  Public-key encryption meets the first requirement but not
   the second.  Using the IPsec Authentication Header (AH) (RFC 4302
   [3]) meets the second requirement using symmetric-key cryptography.
   See RFC 4302 [3] for guidance on multicast per-packet authentication.
   In practice, users of distributed simulation are likely to work over
   a (possibly virtual) private network and thus will not need special
   authentication for SRMP.

8.  List of Acronyms Used

   ACK   - positive acknowledgement
   AH    - Authentication Header
   CLR   - current limiting receiver
   IPSEC - Internet Protocol Security
   MTU   - maximum transmission unit
   NACK  - negative acknowledgement
   RTT   - round-trip time
   SA    - security association
   SRMP  - Selectively Reliable Multicast Protocol
   SRT   - Selectively Reliable Transport
   TFMCC - TCP-Friendly Multicast Congestion Control










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9.  Contributions

   We gratefully acknowledge the significant contributions of two
   people without whom this RFC would not have been developed.
   Vincent Laviano created the first specification and implementation
   of SRMP (at that time called SRTP).  Babu Shanmugam employed SRMP
   in a sizable distributed virtual simulation environment, where he
   revised the implementation and helped revise the design to support
   distributed virtual simulation workload effectively.

10.  References

10.1.  Normative References

   [1]  Bradner, S., "Key words for use in RFCs to Indicate Requirement
        Levels", BCP 14, RFC 2119, March 1997.

   [2]  J. Widmer, M. Handley, Extending Equation-Based Congestion
        Control to Multicast Applications, ACM SIGCOMM Conference, San
        Diego, August 2001.  <http://www.sigcomm.org/sigcomm2001/p22-
        widmer.pdf>

   [3]  Kent, S., "IP Authentication Header", RFC 4302, December 2005.

10.2.  Informative References

   [4]  Pullen, M., Myjak, M., and C. Bouwens, "Limitations of Internet
        Protocol Suite for Distributed Simulation the Large Multicast
        Environment", RFC 2502, February 1999.

   [5]  J. Padhye, V. Firoiu, D. Towsley and J. Kurose, "Modeling TCP
        Throughput: A Simple Model and its Empirical Validation",
        Proceedings of ACM SIGCOMM 1998.

   [6]  Mankin, A., Romanow, A., Bradner, S., and V. Paxson, "IETF
        Criteria for Evaluating Reliable Multicast Transport and
        Application Protocols", RFC 2357, June 1998.

   [7]  Floyd, S., "Congestion Control Principles", BCP 41, RFC 2914,
        September 2000.

   [8]  J. M. Pullen, "The Network Workbench: Network Simulation
        Software for Academic Investigation of Internet Concepts,"
        Computer Networks Vol 32 No 3 pp 365-378, March 2000.







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   [9]  J. M. Pullen, R. Simon, F. Zhao and W. Chang, "NGI-FOM over
        RTI-NG and SRMP: Lessons Learned," Proceedings of the IEEE Fall
        Simulation Interoperability Workshop, paper 03F-SIW-111,
        Orlando, FL, September 2003.

   [10] D. Cohen, "NG-DIS-PDU: The Next Generation of DIS-PDU (IEEE-
        P1278)", 10th Workshop on Standards for Interoperability of
        Distributed Simulations, March 1994.

   [11] Handley, M., Floyd, S., Whetten, B., Kermode, R., Vicisano, L.,
        and M. Luby, "The Reliable Multicast Design Space for Bulk Data
        Transfer", RFC 2887, August 2000.

   [12] Luby, M., Gemmell, J., Vicisano, L., Rizzo, L., and J.
        Crowcroft, "Asynchronous Layered Coding (ALC) Protocol
        Instantiation", RFC 3450, December 2002.

   [13] Luby, M., Gemmell, J., Vicisano, L., Rizzo, L., Handley, M., and
        J. Crowcroft, "Layered Coding Transport (LCT) Building Block",
        RFC 3451, December 2002.

   [14] Luby, M., Vicisano, L., Gemmell, J., Rizzo, L., Handley, M., and
        J. Crowcroft, "Forward Error Correction (FEC) Building Block",
        RFC 3452, December 2002.

   [15] Luby, M., Vicisano, L., Gemmell, J., Rizzo, L., Handley, M., and
        J. Crowcroft, "The Use of Forward Error Correction (FEC) in
        Reliable Multicast", RFC 3453, December 2002.























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Authors' Addresses

   J. Mark Pullen
   C4I Center
   George Mason University
   Fairfax, VA 22030
   USA

   EMail: mpullen@gmu.edu


   Fei Zhao
   C4I Center
   George Mason University
   Fairfax, VA 22030
   USA

   EMail: fzhao@netlab.gmu.edu


   Danny Cohen
   Sun Microsystems
   M/S UMPK16-160
   16 Network Circle
   Menlo Park, CA 94025
   USA

   EMail: danny.cohen@sun.com























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Full Copyright Statement

   Copyright (C) The Internet Society (2006).

   This document is subject to the rights, licenses and restrictions
   contained in BCP 78, and except as set forth therein, the authors
   retain all their rights.

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   "AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE REPRESENTS
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   ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES, EXPRESS OR IMPLIED,
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   specification can be obtained from the IETF on-line IPR repository at
   http://www.ietf.org/ipr.

   The IETF invites any interested party to bring to its attention any
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   rights that may cover technology that may be required to implement
   this standard.  Please address the information to the IETF at
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Acknowledgement

   Funding for the RFC Editor function is provided by the IETF
   Administrative Support Activity (IASA).







Pullen, et al.                Experimental                     [Page 30]

