| Rfc | 4410 |
|---|
| Title | Selectively Reliable Multicast Protocol (SRMP) |
|---|
| Author | M. Pullen, F. Zhao,
D. Cohen |
|---|
| Date | February 2006 |
|---|
| Format: | TXT, HTML |
|---|
| Status: | EXPERIMENTAL |
|---|
|
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.
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
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.
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.
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.
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.
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.
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.
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.
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).
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
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.
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
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
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.
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.
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.
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
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.
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.
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
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.
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.
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.
"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.
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
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.
[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.
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
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.
This document and the information contained herein are provided on an
"AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE REPRESENTS
OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY AND THE INTERNET
ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES, EXPRESS OR IMPLIED,
INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE
INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED
WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.
Intellectual Property
The IETF takes no position regarding the validity or scope of any
Intellectual Property Rights or other rights that might be claimed to
pertain to the implementation or use of the technology described in
this document or the extent to which any license under such rights
might or might not be available; nor does it represent that it has
made any independent effort to identify any such rights. Information
on the procedures with respect to rights in RFC documents can be
found in BCP 78 and BCP 79.
Copies of IPR disclosures made to the IETF Secretariat and any
assurances of licenses to be made available, or the result of an
attempt made to obtain a general license or permission for the use of
such proprietary rights by implementers or users of this
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
copyrights, patents or patent applications, or other proprietary
rights that may cover technology that may be required to implement
this standard. Please address the information to the IETF at
ietf-ipr@ietf.org.
Acknowledgement
Funding for the RFC Editor function is provided by the IETF
Administrative Support Activity (IASA).