Internet Engineering Task Force AVT WG
Internet Draft Julian Chesterfield
draft-chesterfield-avt-rtcpssm-02.txt AT&T Internet Research
Joerg Ott
Tellique Kommunikationstechnik GmbH
November, 2001
Expires: May, 2002
RTCP Extension for Single Source Multicast Sessions with
Unicast RTCP feedback
Status of this Memo
This document is an Internet-Draft and is in full conformance with
all provisions of Section 10 of RFC2026.
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Abstract
This document specifies a modification to the Real-time Transport
Control Protocol to enable the operation of RTP/RTCP using unicast
RTCP feedback for Single Source multicast sessions such as Source
Specific Multicast (SSM) Communication where the traditional model
of Any Source Multicast (ASM) group communication of many to
many is either not possible or not preferred. This draft can be
applied to any group communication which might benefit from a sender
controlled summarised reporting mechanism. It extends [1], section 6
which defines the RTP session group control channel.
1. Conventions and Acronyms
The keywords MUST, MUST NOT, REQUIRED, SHALL, SHALL NOT, SHOULD,
SHOULD NOT, RECOMMENDED, MAY, and OPTIONAL, when they appear in this
document, are to be interpreted as described in RFC 2119.
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2. Introduction
RTP [1] provides a real-time transport mechanism suitable for unicast
or Internet Standard Multicast communication between multimedia
applications. Typical uses are for real-time or near real-time
group communication via audio and video data streams. An important
component of the RTP protocol is the control channel, defined
as the Real-Time Control Protocol (RTCP). RTCP involves the
periodic transmission of control packets between group members in a
session, enabling the distribution or calculation of session specific
information such as packet loss, and round trip time calculation to
other hosts. An additional advantage of providing a control channel
for a session is that a third-party session monitor can listen to the
traffic and establish network conditions and diagnose faults based on
receiver locations.
RTP was designed to operate in a unicast mode or in the traditional
mode of Any Source Multicast (ASM) Group communication which
encompasses a network which supports both one to many and many to
many communication via a common group address in the range 224.0.0.0
through 239.255.255.255. Typical routing protocols that enable such
communication are Distance Vector Multicast Routing Protocol (DVMRP)
[2] or Protocol Independent Multicast (PIM) [3][4] Sparse/Dense Mode
in combination with an Inter-domain routing protocol such as
Multicast Border Gateway Protocol (MBGP) [5] with Multicast Source
Discovery (MSDP) [6]. Such routing protocols enable a host to join a
single Multicast group address and send to or receive traffic from
all members in the group with no prior knowledge of membership. In
order to enable such a service in the network, however there is a
great deal of complexity involved at the routing level. The Source
Specific Multicast (SSM) [7] Model has the advantage of removing a
great deal of the routing complexity involved in multicast group
creation and source information distribution. The disadvantage of SSM
with respect to Real-time traffic using RTP is that the
simplification to the routing protocols removes the ability for any
member of the group to communicate with any other member of the group
without an explicit SSM (Source, Group) or unicast join to that host.
The solution proposed in this draft defines a new method for
distributing control information amongst all members in a multicast
session and is designed to operate under any multicast group
communication scenario. It is, however, of particular benefit to SSM
sessions in the absence of receivers being able to communicate with
each other. The RTP data stream protocol itself is unaffected.
The basic architectural models to which this feedback method could
apply are:
a) SSM groups with a single sender. This is the main motivation
behind the unicast RTCP feedback mechanism and allows SSM groups
which do not have many to many communication capability,
traditionally available in ASM multicast groups to still receive RTP
data streams and operate on them in the usual manner. SSM adopts the
notion of a sender data channel which provides a one to many
communication facility from the source to all the receivers in the
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group. The feedback is unicast to the source on the standard RTCP
port.
b) One to many broadcast networks such as satellite communication
typically using a terrestrial link low bandwidth return channel or a
broadband cable link. This architecture differs very little from the
SSM channel concept, but most likely will require a translator of
some kind to render the RTP data stream onto the satellite or cable
distribution channel.
c) ASM with a single sender. An SDP session announcement type
identifies a session as having a single sender receiving unicast RTCP
feedback. Receivers join the multicast group address and receive RTP
and RTCP data on the specified address/port combinations. The RTCP
feedback is directed to the source on the RTCP port. This model is
not considered to be more efficient than a standard multicast group
RTP communication scenario, and is therefore not recommended to
replace the traditional mechanism, however it might be useful in
helping to prevent overtaxing multicast routing infrastructure that
does not scale as efficiently.
SSM sessions are typically assigned a value in the group address
range 232.0.0.0 through 232.255.255.255, although this is not a
requirement. A session may be assigned any valid multicast address,
as long as the local network is configured to allow source specific
joins outside the suggested SSM range. In order for a host to receive
traffic from an SSM capable source, it must support the IGMPv3
multicast group membership reporting protocol which enables the host
to explicitly request traffic from a source,group pair. In this case,
the host is aware of the significance of the address range and is
therefore capable of identifying the unicast RTCP feedback session
requirements based on this knowledge. For sessions which take
advantage of the unicast feedback model but do not inherently need to
use it, it is anticipated that an SDP syntax will be defined.
The modifications proposed in this document are intended to provide
an optional replacement to the method of RTCP operation for sessions
which either require or may benefit from a new reporting structure.
For certain distribution networks, such as SSM networks, this may be
a requirement, however in other cases this is an optional feature
which may be used.
3. Basic Operation
This draft proposes two methods for enabling receiver feedback to
all members in a session. Each involves the unicasting of RTCP
packets to a source whose job it is to distribute the information
to the members of the group. The source must always be able to
communicate with all the other members in order for either mechanism
to work.
The first method, the 'Simple Feedback Model' is a basic mechanism
whereby all receiver reports are unicast to the source and
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subsequently forwarded by the source to all receivers on the
multicast feedback channel. The advantage of using this method is
that an existing receiver implementation requires little modification
in order to operate in this new state. Instead of forwarding Receiver
Reports to a multicast address, it uses a unicast address and still
receives RTCP traffic in the usual manner. This method also has the
advantage of being backwards compatible with RTP/RTCP implementations
which do not support unicast feedback to the source and operate using
the standard multicast group communication model, ASM. In a session
that is using ASM, such a receiver would multicast Receiver Reports
to the group address and port+1 as stated in [1]. This would still be
received by all receivers. In a session using an SSM distribution
network, the network would prevent any data from the receiver being
distributed further than the first hop router. Additionally, any data
heard from this receiver by other hosts on the same subnet should be
filtered out by the host IP stack and will therefore not cause any
problems with respect to the calculation of Receiver RTCP bandwidth
since this receiver will not be heard by any other members.
The second method, the 'Sender Feedback Summary Model' is a
summarised reporting scheme that provides savings in bandwidth by
consolidating all the receiver reports into one summary packet which
is then distributed to all the receivers. The advantage of this
scheme is apparent for large group sessions where the basic
forwarding mechanism outlined above would create a large amount of
packet replication in order to forward all the information to all the
receivers. The basic operation of the scheme is the same as the first
method, however it requires that all the members in the session
understand the new summarised packet format outlined in section 7.1.
To differentiate between the two reporting mechanisms, a new SDP
identifier is created and discussed in section 10. The method of
reporting must be decided prior to the start of the session, a
distribution source may not change the method during a session.
4. Definitions
Distribution Source: In order for unicast feedback to work, there
must only be one session distribution source for any subset of
receivers to which RTCP feedback is directed. Heterogeneous
networks comprised of ASM multiple sender groups, unicast
only clients and/or SSM single sender/receiver groups may be
connected via translators or mixers (see section 9 for details on
this) to create a single source group. However, in order for
unicast feedback to work, only one source must be responsible for
distributing the RTP stream and forwarding RTCP information to all
receivers.
RTP and RTCP Channels: The data distribution from the source to the
receivers whether via an SSM {source,group} identifier, a standard
ASM multicast group or a unicast reflector, is referred to as the
RTP and RTCP channels. These channels are differentiated via the
port numbers as [port] and [port + 1] for RTP and RTCP
respectively. See [1] for further explanation of the port
numbering.
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Unicast RTCP Feedback Target: For a session defined as having a
distribution source A, on ports n and n+1, the unicast feedback
target is the IP address of Source A on port n+1.
SSRC: Synchronization source. A 32-bit value that uniquely identifies
each member in a session. See [1] for further information.
Report blocks: In the RTCP design [1] it is encouraged to stack
multiple report blocks in Sender and Receiver report packets. In
this way, a variable size packet is created which can include
information from one source pertaining to multiple sources in the
group. The concept of report blocks is extended in this draft to
encompass Loss Jitter Summary packets in which a source can
optionally stack multiple reports into one packet in order to
provide additional feedback on the RTCP traffic received from the
group.
5. Packet types
The RTCP packet types defined in [1] are:
type description Payload number
SR sender report 200
RR receiver report 201
SDES source description 202
BYE goodbye 203
APP application-defined 204
These remain unmodified. In addition to the exisiting types, two new
packet types are introduced. Further information on each of these is
provided in this draft. The packet types are:
type description Payload number
RSI Receiver Summary Information [see section 12]
LJS Loss and Jitter Summary [see section 12]
6. Simple feedback model
6.1 Packet Formats
For this mechanism, the packet types used remain the same as for
standard RTCP feedback in [1]. Receivers generate Receiver Reports
with information on the quality of the stream received from the
source. The source must create Sender Reports which include timestamp
information for stream synchronisation and round trip time
calculation. Both senders and receivers are required to send SDES
packets as outlined in [1]. The usual rules for BYE and APP packets
also apply.
6.2 Distribution Source behaviour
For the simple feedback model, the source provides a simple packet
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reflection mechanism. It is the default behaviour for any
distribution source and is the minimum requirement for acting as a
source to a group of receivers using unicast RTCP feedback.
The source may not stack report blocks received from different SSRCs
into one packet for retransmission to the group. Every RTCP packet
from each receiver must be reflected individually.
The source must listen for unicast RTCP data sent to the RTCP port.
All unicast data received on this port must be forwarded to the
group on the multicast RTP channel. Any multicast data received on
this port must not be forwarded but processed as defined in [1].
The reflected traffic should not be included in the transmission
interval calculation by the source. In other words the source should
not consider reflected packets as part of it's own control data
bandwidth allowance. The algorithm for computing the allowance is
explained in section 9. The control bandwidth traffic included in the
calculation includes any Sender reports to the group, along with any
additional SDES and APP packets.
If an application wishes to use APP packets, it is recommended that
the 'simple feedback model' be used since it is likely that all
receivers in the session will need to hear the APP specific packets.
This decision must be made in advance of the session and indicated in
the SDP announcement.
6.3 Receiver behaviour
Receivers listen on the RTP and RTCP channels for data. Each receiver
calculates it's share of the receiver bandwidth based on the
standard rules i.e. 75% of the RTCP bandwidth is divided equally
between all unique SSRCs in the session. See section 9 for further
information on this. When a receiver is eligible to transmit, it
sends a unicast Receiver Report packet to the RTCP port of the
distribution source.
7. Sender feedback summary model
In the sender feedback summary mode, the sender is required to
summarise the information received from all the Receiver Reports
generated by the receivers and place the information into summary
reports. The sender must send at least 1 Receiver Summary Information
packet for each reporting interval. The sender can additionally stack
Loss Jitter Summary (LJS) reports after the RSI packet. Each LJS
packet corresponds to the initial RSI packet and acts as an
enhancement to the basic summary information required by the
receivers to calculate their reporting time interval. For this
reason LJS packets are not required but recommended. RSI and LJS
packets are sent in addition to the standard Sender Reports and SDES
packets outlined in [1].
7.1 Packet Formats
The Sender feedback summary model introduces 2 new packet formats.
The Receiver Summary Information packet (RSI) which must be sent by a
source if the summarised feedback mechanism is selected and the
optional Loss and Jitter Summary report packet (LJS) that may be
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appended to the RSI packet to provide more detailed information on
the overall session characteristics reported by all receivers.
7.1.1 RSI: Receiver Summary Information RTCP Packet
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|V=2|P| SC | PT | length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC of Sender |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| group size |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| AFL | HCNL |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Highest interarrival jitter |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Receiver RTCP Bandwidth |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| collision SSRC #1 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| . . . |
The RSI packet consists of a main report block modeled along the same
lines as a receiver report with optional LJS blocks appended. The
first 4 bytes of header extension follow the standard RTP header
outline. This ensures backwards compatibility with older versions
which may not understand the RSI packet format but can read the
length field indicating the end of the report block. The following
fields are included:
The fields "V", "P", and "length" have the same meaning as per [1].
SC: 5 bits
The number of collision SSRC entries towards the end of the report
block. A value of 0 is allowed.
SSRC: 32 bits
The synchronisation source identifier for the originator of the
summary report packet.
group size: 32 bits
This field provides the sender's view of the number of receivers
in a session. This should include the sender itself and any other
senders potentially connected to the session e.g. via a
mixer/translator gateway. The group size is calculated according
to the rules outlined in [1].
Average fraction lost (AFL): 8 bits
The average fraction lost indicated by receiver reports forwarded
to this source, expressed as a fixed point number with the binary
point at the left edge of the field.
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Highest cumulative number of packets lost (HCNL): 24 bits
Highest 'cumulative number of packets lost' value out of all RTCP
RR packets since the last RSI from any of the receivers.
Highest interarrival jitter: 32 bits
Highest 'interarrival jitter' value out of all RTCP RR packets
since the last RSI from any of the receivers.
receiver bandwidth: 32 bits
indicates the maximum bandwidth allocated to any single receiver
for sending RTCP data relating to the session. This is a fraction
value indicating a percentage of the session bandwidth, expressed
as a fixed point number with the binary point at the left edge of
the field.
collision SSRC: n x 32 bits
the final fields in the packet are used to identify any SSRCs
that are duplicated within the group. Usually this is handled by
the hosts upon detection of the same SSRC, however since receivers
no longer have a global view of the session, the collision
algorithm is handled by the source. SSRCs that collide are listed
in the packet and it is the responsibility of the receiver(s) to
detect the collision and select another ID.
7.1.2 LJS: Loss Jitter Summary RTCP Packet
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|V=2|P| LJSC | PT | Length | header
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC of Sender |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| NLB | LF | MIL | MAL | NJB | JF |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Minimum Jitter Value |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Maximum Jitter Value | report
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ block
| Loss Buckets | 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Loss Buckets cont... |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Jitter Buckets |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Jitter Buckets cont... |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| ... | report
| ... | block
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ 2
Loss Jitter Summary Count (LJSC): 5 bits
The number of Loss Jitter Summary report blocks contained in this
packet
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Number of Loss Buckets (NLB): 4 bits
The number of Loss Buckets over the reserved 64 bit space.
Possible values are 0, 1, 2, 4, 8, 16
Loss Factor (LF): 4 bits
Indicates the multiplicative factor to be applied to the Loss
Bucket values. Possible values are 1 - 15.
Minimum Loss Value (MIL): 8 bits
Minimum loss value. In combination with the Maximum Loss value
indicates the range covered by the Loss Bucket values. Possible
values are 0 - 99. The Minimum Loss Value must always be less than
the maximum, expressed as a fixed point number with the binary
point at the left edge of the field.
Maximum Loss Value (MAL): 8 bits
Maximum loss value. In combination with the Minimum Loss value
indicates the range covered by the Loss Bucket values. Possible
values are 1 - 100. The maximum Loss Value must always be greater
than the minimum, expressed as a fixed point number with the
binary point at the left edge of the field.
Number of Jitter Buckets (NJB): 4 bits
The number of Jitter Buckets over the reserved 64 bit space.
Possible values are 0, 1, 2, 4, 8, 16
Jitter Factor (JF): 4 bits
Indicates the multiplicative factor to be applied to the Jitter
Bucket values. Possible values are 1 - 15.
Minimum Jitter Value (MIJ): 32 bits
Minimum jitter value. In combination with the Maximum jitter value
indicates the range covered by the jitter Bucket values. The
Minimum jitter Value must always be less than the maximum.
Maximum Jitter Value (MAJ): 32 bits
Maximum jitter value. In combination with the Minimum jitter value
indicates the range covered by the jitter Bucket values. The
Maximum jitter Value must always be greater than the minimum.
Loss Buckets: 16*4 bits - 8*8 bits - 4*16 bits - 2*32 bits - 1*64
bits
Loss Bucket. The size and number of buckets depends upon the value
of NLB. This indicates the division of the 64 bit space. Depending
upon whether NLB is 16, 8, 4, 2 or 1, the size of each LB will be
4, 8, 16, 32 or 64 bits respectively. Each value must be
multiplied by the Loss Factor.
Jitter Buckets: 16*4 bits - 8*8 bits - 4*16 bits - 2*32 bits - 1*64
bits
Jitter Bucket. The size of the bucket depends upon the value of
JLB. This indicates the division of the 64 bit space. Depending
upon whether JLB is 16, 8, 4, 2 or 1, the size of each JB will be
4, 8, 16, 32 or 64 bits respectively. Each value must be
multiplied by the Jitter Factor.
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7.2 Distribution Source behaviour
The length field of the RSI packet must be calculated over the length
of the whole packet, using the
method defined in [1]. The group size must be included in the RSI
packet. The source should also calculate the Receiver RTCP bandwidth
field. Typically this value will be calculated as outlined in [1]
using the group size and session bandwidth as variables. This field
does however provide the source with the capability to control the
amount of feedback from the receivers and can be increased or
decreased based on the requirements of the source. Regardless of the
value selected by the source for the RTCP bandwidth field, the source
must continue to forward Sender reports and RSI packets at the rate
allowed by its bandwidth allocation. See section 9 for further
details.
In order to identify SSRC collisions, the source is responsible for
maintaining a record of each SSRC and the correpsonding IP address
within at least one reporting interval in order to differentiate
between clients. It is recommended that an updated list of more than
one interval be maintained to increase accuracy. This mechanism
does not prevent the possibility of collisions since IP addresses may
not be unique e.g. due to NAT gateways, however it greatly increases
the capability to detect collisions. In the event that collisions are
not detected, the effect will be an innaccurate impression of the
group size on the part of the source. Since the statistical
probablility that collisions will both occur and be undetectable is
very low, the clients would have to randomly select the same SSRC and
be located behind the same NAT gateway, this should not be a
significant concern.
For the LJS packet, the source must decide which are the most
significant values to convey. The packet format provides flexibility
in the amount of detail conveyed by the data points. There is a
trade-off between the granularity of the data and the accuracy based
on the factorisation values, the number of buckets and the min and
max values. In order to focus on a particular region of the
distribution, the source can adjust the minimum and maximum values
and either increase the number of buckets and possibly the
factorisation, or decrease the number of buckets and provide more
accurate values. See Appendix B for detailed examples on how to
convey RTCP reports as LJS information.
The results should correspond as near as possible to the values
received during the interval since the last report.
The source may stack as many report blocks as required in order to
convey loss and jitter information.
7.3 Receiver behaviour
The receiver must process RSI packets and adapt session parameters
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such as the RTCP bandwidth based on the information received. The
receiver no longer has a global view of the session, and will
therefore be unable to receive information from individual receivers
aside from itself. However, the information portrayed by the source
can be extremely detailed, providing the receiver with an accurate
view of the session quality overall, without the processing overhead
associated with listening to and analysing all the receiver reports.
The SSRC collision list must be checked against the SSRC selected by
the receiver to ensure there are no collisions. The group size value
provides the receiver with the data necessary to calculate it's share
of the RTCP bandwidth. This share of the bandwidth may be overridden
by the 'Receiver RTCP Bandwidth' field. This field provides the
source with the capability to control the amount of feedback from the
receivers.
The receiver can handle the LJS data as desired. This data is most
useful in providing the receiver with a more global view of the
conditions experienced by other receivers, and enables the client to
place itself within the distribution and establish the extent to
which it's reported conditions correspond to the group reports as a
whole. Appendix A provides further information and examples of
data processing at the receiver.
The receiver should assume that any report blocks in the same packet
correspond to the same data set received by the source during the
last reporting time interval. This applies to packets with multiple
blocks, where each block conveys a different range of values.
8. Mixer/Translator issues
The original RTP specification allows for the use of mixers and
translators in an RTP session which help to connect heterogeneous
networks into one session. There are a number of issues, however
which are raised by the unicast feedback model proposed in this
document. The term 'mixer' refers to devices that provide data stream
multiplexing where multiple sources are combined into one stream.
Conversely, a translator does not multiplex streams, but simply
acts as a bridge between two distribution mechanisms, e.g. a unicast
to multicast network translator. Since the issues raised by this
draft apply equally to either a mixer or translator, they are
referred to from this point onwards generically as a gateway.
A gateway between distribution networks in a session must ensure that
all members in the session receive all the relevant traffic to enable
the usual operation by the clients. A typical use may be to connect
an older implementation of an RTP client with an SSM distribution
network, where the client is not capable of unicasting feedback to
the source. In this instance the gateway must join the session on
behalf of the client and send and receive traffic from the session to
the client. Certain hybrid scenarios may have different requirements.
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8.1 Use of a mixer-translator
The gateway must adhere to the SDP descriptor for the single source
session and use the feedback mechanism indicated. Receivers should be
aware that by introducing a gateway into the session, more than one
source may potentially be active in a session since the gateway may
be forwarding traffic from either multiple unicast sources or from an
ASM session to the SSM receivers. Receivers should still forward
unicast RTCP reports in the usual manner to the distribution source,
which in this case would be the gateway itself. It is recommended
that the simple packet reflection mechanism be used under these
circumstances since attempting to coordinate RSI + LJS reporting
between more than one source may be complicated unless the gateway is
capable of undertaking the summarisation itself.
8.2 Encryption and Authentication issues
Encryption and security issues are discussed in detail in section 11.
A gateway must be able to follow the same security policy as the
client in order to unicast forward RTCP data to the source, and it
therefore must be able to apply the same authentication and/or
encryption policy required for the session. Transparent bridging,
where the gateway is not acting as the distribution source, and
subsequent unicast feedback to the source is only allowed if the
gateway can conduct the same source authentication as required by the
receivers.
9. Transmission interval calculation
The Control Traffic Bandwidth referred to in [1] is an arbitrary
amount which is intended to be supplied by a session management
application (e.g. [9]) or decided based upon the bandwidth of a
single sender in a session. A receiver must calculate the number of
other members in a session based upon either it's own SSRC count
determined by the forwarded Receiver Reports, or from the RSI report
from a sender.
The RTCP transmission Interval calculation remains the same as in the
original RTP specification [1]. In the original specification, the
senders are allocated 1/4 of the control traffic bandwidth if
they number 25% or less than the group size. Otherwise the allocation
for senders is the percentage of senders to group size. The remaining
bandwidth is allocated to the receivers to be divided evenly amongst
the group. The source
should calculate the transmission interval for RSI + LJS packets out
of it's 1/4 of the control traffic bandwidth with a minimum
transmission interval of 5 seconds.
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10. SDP Extensions
The Session Description Protocol (SDP) is used as a means to describe
media sessions in terms of their transport addresses, codecs, and
further attributes. Providing RTCP feedback via unicast as specified
in this document constitutes another session parameter. To make
receivers aware that they are supposed to provide their feedback
via unicast, this needs to be indicated in the session description.
Similarly, parameters of SSM RTCP feedback -- such as the mode of
summarizing information at the sender and the target unicast address
to send feedback information to -- needs to be provided. This section
defines the necessary SDP parameters (that also need to be registered
with IANA).
10.1 SSM RTCP Session Identification
A new session level attributes MUST be used to indicate the use of
unicast instead of multicast feedback: "rtcp:unicast".
This attribute uses one further parameter to specify the mode of
operation.
rtcp:unicast reflection -- MUST be used to indicate packet reflection
by the RTCP target (without further
processing).
rtcp:unicast ljs -- MUST be used to indicate the "Loss Jitter
Summary" mode of operation
rtcp:unicast rsi -- MUST be used to indicate the "Receiver
Summary Information" mode of operation.
10.2 SSM Source Specification
In addition, in an SSM RTCP session, the sender(s) need to be
indicated for both source-specific joins to the multicast group
as well as for addressing RTCP packets to.
This is done following the proposal for SDP source filters
documented in draft-ietf-mmusic-sdp-srcfilter-00.txt [15].
From this specification, only the inclusion mode ("a=incl:")
MUST be used for SSM RTCP.
There SHOULD be exactly one "a=incl:" attribute listing the
address of the sender. The RTCP port MUST be derived from the
m= line of the media description.
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11. Security Considerations
Packet bombing of unsuspecting victims via a fake SDP or SSM address
is a real concern for this architecture. For this reason it is
required that a security policy be applied to any session which
involves unicast feedback of data to a single IP address. At a
minimum, it is recommended that source authentication be conducted by
every receiver prior to unicasting data.
An additional concern is the problem of fake RSI + LJS packets which
could increase the RTCP bandwidth sent to the source. Any security
policy must address this as a minimum requirement.
Receiver authentication would also be beneficial, since Denial of
Service attacks by generating false Receiver Reports is also
possible. The consequences of this are not as drastic, affecting only
the group size and transmission interval calculation and therefore
the integrity and frequency of the reported data.
The issue of source feedback implosion should not occur in the event
that receivers practice the standard RTP/RTCP guidelines for starting
sessions and for implementing the scaling algorithm based on the
number of hosts. An additional issue which should be addressed, but
is beyond the scope of this document is the potential for host
anonymity which is facilitated by Source Specific Multicast and adds
additional security measures into group communication. By explicitly
controlling receiver feedback, a source could solicit feedback from
the receivers in a scalable way without the need to inform all
members in a session of the group membership.
11.1 Security Requirements Outline
In order to overcome the issues outlined above, there are some
minimal and recommended policies which must be addressed:
- The information providing the unicast feedback address
needs to be authenticated as being from a trusted source.
- Data integrity of the RTCP traffic from the source, particularly
RSI + LJS packets is also required.
- Receiver authentication is recommended in order to ensure
integrity of RTCP traffic and group size.
- Data encryption of both the RTCP and RTP streams are optional
but recommended for this draft.
Ideally, a public key infrastructure would provide the mechanism
necessary to ensure the trusted authentication of distributed SDP
announcements. Since this is not generally available, the following
precautions are highly recommended.
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The primary danger of the use of a fake session announcement must be
addressed by the distribution media itself since SDP remains
independent of the underlying mechanism and provides no facility to
combat authentication and/or message integrity. The most common
methods of distributing SDP messages are the Session Announcement
Protocol (SAP)[11], a web page or an email message. All of these
mechanisms provide the capability to authenticate the source of the
announcement:
- SAP has the option to include an authentication header in
the message which assures the integrity of the message contents
and identifies the source of the message via public key
encryption.
- A secure web server can be used to provide Secure Sockets Layer
(SSL) [12] authentication of the web site containing the SDP
message.
- An email message can be signed using a public key mechanism to
ensure data integrity.
All of these methods rely on a level of trust in order to validate
the public key of the originator of the message. The establishment of
trust is beyond the scope of this document, however it is recommended
that receivers should only trust an originating source if a digital
certificate signed by a trusted third party such as a Signing
Authority is available or if the key has been received prior to the
session via some secure out-of-band method.
A number of options are available to address the issue of session
data integrity, the most obvious being the use of Secure RTP (SRTP)
[14] or a more general security framework such as TESLA [13].
Adopting such a scheme to ensure that both source traffic and
receiver messages are encrypted would prevent the generation of fake
RTCP traffic to the group or from any unsolicited receivers.
12. IANA Considerations
Based on the guidelines suggested in [10], this document proposes 2
new RTCP data payload types for consideration by IANA.
Furthermore, four new SDP media-level attributes are defined in
section 10.
13. References
[1] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson,
"RTP - A Transport Protocol for Real-time Applications," Internet
Draft, draft-ietf-avt-rtp-new-10.txt, Work in Progress, July 2001.
[2] Pusateri, T, "Distance Vector Multicast Routing Protocol",
draft-ietf-idmr-dvmrp-v3-10, August 2000
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[3] Fenner, B, Handley, M, Holbrook, H, Kouvelas, I, "Protocol
Independent Multicast - Sparse Mode (PIM-SM): Protocol Specification
(Revised)", draft-ietf-pim-sm-v2-new-02.txt, March 2001
[4] Farinacci, D, Kouvelas, I, Windisch, K, "State Refresh in PIM-DM"
draft-ietf-pim-refresh-02.txt, November, 2000
[5] Thaler, D, Cain, B, "BGP Attributes for Multicast Tree
Construction", draft-ietf-idmr-bgp-mcast-attr-00.txt, February 1999
[6] Farinacci, D, Rekhter, Y, Meyer, D, Lothberg, P, Kilmer, H,
Hall, J, "Multicast Source Discovery Protocol (MSDP)",
draft-ietf-msdp-spec-06.txt, July 2000
[7] Shepherd, G, Luczycki, E, Rockell, R, "Source-Specific Protocol
Independent Multicast in 232/8", draft-shepherd-ssm232-00.txt, March
2000.
[8] Holbrook, H, Cain, B, "Using IGMPv3 For Source-Specific
Multicast", draft-holbrook-idmr-igmpv3-ssm-00.txt, July 2000.
[9] Session Directory Rendez-vous (SDR), developed at University
College London by Mark Handley and the Multimedia Research Group.
[10] Alvestrand, H. and T. Narten, "Guidelines for Writing an IANA
Considerations Section in RFCs", BCP 26, RFC 2434, October 1998.
[11] Handley, M, Perkins, C, Whelan, E, "Session Announcement
Protocol", (SAP), RFC 2974, October 2000.
[12] A. Frier, P. Karlton, and P. Kocher, "The SSL 3.0 Protocol",
Netscape Communications Corp., Nov 18, 1996.
[13] Perrig, Canetti, Briscoe, Tygar, Song, "TESLA: Multicast Source
Authentication Transform", draft-irtf-smug-tesla-00.txt.
[14] E. Carrara, D. McGrew, M. Naslund, K. Norrman, D. Oran, "The
Secure Real Time Transport Protocol", draft-ietf-avt-srtp-01.txt.
[15] B. Quinn, "SDP Source-Filters", Internet Draft
draft-ietf-mmusic-sdp-srcfilter-00.txt, Work in Progress, May 2000.
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13. Appendix
A LJS packet processing at the receiver
A.1 Algorithm
X values represent the loss percentage.
Y values represent the number of receivers.
Number of x values is the NLB value
xrange = MAL - MIL
First data point = MIL,first ydata
then
Foreach ydata => xdata += (MIL + (xrange / NLB))
A.2 Pseudo-code
Packet Variables -> factor,NLB,MIL,MAL
Code variables -> xrange, ydata[NLB],x,y
xrange = MAL - MIL
x = MIL;
for(i=0;i<NLB;i++) {
y = ydata[i] * factor;
/*OUTPUT x and y values*/
x += (xrange / NLB)
}
B LJS packet creation at the source
See Postscript version.
C AUTHORS ADDRESSES
Julian Chesterfield
AT&T Labs - Research
75 Willow Road
Menlo Park, CA 94025
julian@research.att.com
Joerg Ott
Tellique Kommunikationstechnik GmbH
Berliner Str. 26
D-13507 Berlin
GERMANY
Phone: +49.30.43095-560 (sip:jo@tzi.org)
Fax: +49.30.43095-579
Email: jo@tellique.com
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D FULL COPYRIGHT STATEMENT
Copyright (C) The Internet Society (2000). All Rights Reserved.
This document and translations of it may be copied and furnished to
others, and derivative works that comment on or otherwise explain it
or assist in its implementation may be prepared, copied, published
and distributed, in whole or in part, without restriction of any
kind, provided that the above copyright notice and this paragraph are
included on all such copies and derivative works. However, this doc-
ument itself may not be modified in any way, such as by removing the
copyright notice or references to the Internet Society or other
Internet organizations, except as needed for the purpose of develop-
ing Internet standards in which case the procedures for copyrights
defined in the Internet languages other than English.
The limited permissions granted above are perpetual and will not be
revoked by the Internet Society or its successors or assigns.
This document and the information contained herein is provided on an
"AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING
TASK FORCE DISCLAIMS 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 MER-
CHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE."
Chesterfield, Ott [Page 18]