Internet DRAFT - draft-westerlund-mmusic-sdp-bwparam
draft-westerlund-mmusic-sdp-bwparam
Network Working Group Magnus Westerlund
INTERNET-DRAFT Ericsson
Expires: April 2003
A Transport Independent Bandwidth Modifier for the Session
Description Protocol (SDP).
<draft-westerlund-mmusic-sdp-bwparam-01.txt>
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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This document is an individual submission to the IETF. Comments
should be directed to the authors.
Abstract
The existing Session Description Protocol (SDP) bandwidth modifiers
and their values include the bandwidth needed also for the transport
and IP layers. When using SDP in protocols like SAP, SIP and RTSP and
the involved hosts reside in networks running different IP versions,
the interpretation of what type of lower layers that is included is
not clear. This documents defines a bandwidth modifier that does not
include transport overhead, instead additional packet rate attributes
are defined. The transport independent bitrate value together with
the packet rate can then be used to calculate the real bitrate over
the actually used transport.
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TABLE OF CONTENTS
1. Definitions.........................................................2
1.1. Glossary.......................................................2
1.2. Terminology....................................................3
2. Changes.............................................................3
3. Introduction........................................................3
3.1. The Bandwidth Attribute........................................3
3.1.1. Conference Total..........................................4
3.1.2. Application Specific Maximum..............................4
3.1.3. RTCP Report bandwidth.....................................4
3.2. IPv6 and IPv4..................................................4
4. The Bandwidth Signaling Problems....................................5
4.1. What IP version is used........................................5
4.2. Converting bandwidth values....................................6
4.3. Header Compression.............................................6
4.4. RTCP problems..................................................7
4.5. Future development.............................................7
5. Problem Scope.......................................................7
6. Requirements........................................................8
7. A Solution..........................................................8
7.1. Introduction...................................................8
7.2. The TIAS bandwidth modifier....................................8
7.2.1. Usage.....................................................8
7.2.2. Definition................................................9
7.2.3. Usage Rules...............................................9
7.2.4. Deriving RTCP bandwidth..................................10
7.3. Packet Rate parameters........................................10
7.4. Converting to Transport Dependent values......................11
7.5. ABNF definitions..............................................12
8. Protocol Interaction...............................................12
8.1. RTSP..........................................................12
8.2. SIP...........................................................12
8.3. SAP...........................................................12
9. Security Consideration.............................................13
10. IANA Consideration................................................13
11. Acknowledgments...................................................13
12. Author's Addresses................................................13
13. References........................................................14
13.1. Normative references.........................................14
13.2. Informative References.......................................14
1. Definitions
1.1. Glossary
RTSP - Real-Time Streaming Protocol
SDP - Session Description Protocol
SAP - Session Announcement Protocol
SIP - Session Initiation Protocol
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TIAS - Transport Independent Application Specific maximum, a
bandwidth modifier.
1.2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [7].
2. Changes
The following changes has been done to this version compared to
draft-westerlund-mmusic-sdp-bwparam-00.txt:
- TIAS definition has been changed to not any longer include RTP
headers in a RTP context.
- The TIAS value is now defined in bits per seconds instead of
kilobits per second.
- Rules for how to use TIAS together with "maxprate" and "avgprate"
has been defined
- The old prate attribute has been renamed "maxprate" and a new
attribute "avgprate" has been defined.
- Use cases for TIAS has been defined
- Clarifications in the algorithm to derive transport dependent
bitrates.
- The RTCP related problem is also included in the problem
description and a solution proposed.
3. Introduction
Today the Session Description Protocol (SDP) [1] is used in several
types of applications. The original application is session
information and configuration for multicast sessions announced with
Session Announcement Protocol (SAP) [2]. SDP is also a vital
component in media negotiation for the Session Initiation Protocol
(SIP) [3] by using the offer answer model [4]. The Real-Time
Streaming Protocol (RTSP) [5] also makes use of SDP to declare what
media and codec(s) a multi-media presentation consist of to the
client.
3.1. The Bandwidth Attribute
In SDP there exist a bandwidth attribute, which has a modifier used
to specify what type of bitrate the value refers to. The attribute
has the following form:
b=<modifier>:<value>
Today there are four modifiers defined which are used for different
purposes.
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3.1.1. Conference Total
The Conference Total is indicated by giving the modifier "CT". The
meaning of Conference total is to give a maximum bandwidth that a
conference session will use. Its purpose is so that it is possible to
decide if this session can co-exist with any other sessions. Defined
in RFC 2327 [1].
3.1.2. Application Specific Maximum
The Application Specific maximum bandwidth is indicated by the
modifier "AS". The interpretation of this attribute is depending on
the application's notion of maximum bandwidth. For an RTP application
this attribute is the RTP session bandwidth as defined in RFC 1889
[6]. The session bandwidth includes the bandwidth that the RTP data
traffic will result in, including the lower layers down to IP layer.
So the bandwidth is in most cases calculated over RTP payload, RTP
header, UDP and IP. Defined in RFC 2327 [1].
3.1.3. RTCP Report bandwidth
Today there is a draft [9], currently in the RFC editors queue to
become a Proposed Standard, which defines two new bandwidth
modifiers. These modifiers "RS" and "RR", define the amount of
bandwidth that is assigned for RTCP reports by active data senders
respectively RTCP reports by receivers only.
3.2. IPv6 and IPv4
Today there are two IP versions 4 [15] and 6 [14] used in parallel on
the Internet. This creates problems and there exist a number of
possible transition mechanisms.
------------------ ----------------------
| IPv4 domain | | IPv6 Domain |
| | ---------| | |
| ---------- |-| NAT-PT |-| ---------- |
| |Server A| | ---------| | |Client B| |
| ---------- | | ---------- |
------------------ ----------------------
Figure 1. Translation between IPv6 and IPv4 addresses.
- To achieve connectivity between an IPv4 only host and an IPv6
only host translation is necessary. This translator can be for
example Network Address Translation - Protocol Translation (NAT-
PT) [13], see Figure 1. However to get connectivity for large
number of protocols, Application Level Gateway (ALG) functionality
is also required at the node. To be able to locate hosts through
the translation node DNS ALG must be supported.
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- IPv6 nodes belonging to different domains running IPv6, but
lacking IPv6 connectivity between them solves this by tunneling
over the IPv4 net, see Figure 2. Basically the IPv6 packets are
put as payload in IPv4 packets between the tunneling end-points at
the edge of each IPv6 domain.
--------------- --------------- ---------------
| IPv6 domain | | IPv4 domain | | IPv6 Domain |
| | |-------------| | |
| ---------- |--||Tunnel ||--| ---------- |
| |Server A| | |-------------| | |Client B| |
| ---------- | | | | ---------- |
--------------- --------------- --------------|
Figure 2. Tunneling through a IPv4 domain
IPv4 has minimal header size of 20 bytes. While the fixed part of the
IPv6 header is 40 bytes.
The difference in header size results in that the bandwidth used for
the IP layer is different. How big the difference is, depends on the
packet rate.
4. The Bandwidth Signaling Problems
When an application wants to use SDP to signal the bandwidth required
for this application some problems becomes evident depending on the
transport layers.
4.1. What IP version is used
If one signals the bandwidth in SDP, with for example "b=AS:" for an
RTP based application, one cannot know if the overhead is calculated
for IPv4 or IPv6. An indication to which protocol has been used when
calculating the bandwidth values is given by the "c=" connection data
line. This line contains either a multicast group address or a
unicast address of the data source or sink. The "c=" lines address
type may be assumed to be of the same type as the one used in the
bandwidth calculation. There seems to exist no specification pointing
this out.
In cases of SDP transported by RTSP this is even less clear. The
normal usage for a unicast on-demand streaming session is to set the
connection data address to a null address. This null address does
have an address type, which could be used as an indication. However
this is not clarified anywhere.
Figure 1, illustrates a connection scenario between a streaming
server A and a client B over a translator here designated as a NAT-
PT. When B receives the SDP from A over RTSP it will be very
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difficult for B to know what the bandwidth values in the SDP
represent. The following possibilities exist:
1. The SDP is unchanged and "c=" null address is of type IPv4. The
bandwidth value represents the bandwidth needed in an IPv4
network.
2. The SDP has been changed by the ALG. The "c=" address is changed
to IPv6 type. The bandwidth value is unchanged.
3. The SDP is changed and both "c=" address type and bandwidth value
is changed. Unfortunately, this can seldom be done, see 4.2.
In case 1 the client can understand that the server is located in an
IPv4 network and that it uses IPv4 overhead when calculating the
bandwidth value. The client almost never convert the bandwidth value,
see section 4.2.
In case 2 the client does not know that the server is in an IPv4
network and that the bandwidth value is not calculated with IPv6
overhead. In cases where a client reserve bandwidth for this flow,
too little will be reserved, potentially resulting in bad Quality of
Service (QoS).
In case 3 everything works correctly. However this case will be very
rare. If one tries to convert the bandwidth value without further
information about the packet rate significant errors will be
introduced into the value.
4.2. Converting bandwidth values
If one would like to convert a bandwidth value calculated using IPv4
overhead to IPv6 overhead the packet rate is required. The new
bandwidth value for IPv6 is normally "IPv4 bandwidth" + "packet rate"
* 20 bytes. Where 20 bytes is the usual difference between IPv6 and
IPv4 headers. The overhead difference may be other in cases when IPv4
options [15] or IPv6 extension headers [14] are used.
As converting requires the packet rate of the stream, this is not
possible in the general case. Many codecs has many possible packet
rates. Therefore some extra information in the SDP will be required.
The "a=ptime:" parameter may be a possible candidate. However this
parameter is normally only used for audio codecs. Also its definition
[1] is that it is only a recommendation, which the sender may
disregard from. A better parameter is needed.
4.3. Header Compression
Another mechanism that alters the actual overhead over links is
header compression. Header compression uses the fact that most
network protocol headers have either static or predictable values in
their fields within a packet stream. Compression is normally only
done on per hop basis, i.e. on a single link. The normal reason for
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doing header compression is that the link has fairly limited
bandwidth and significant gain in throughput is achieved.
There exist a couple of different header compression standard. For
compressing IP headers only, there exist RFC 2507 [10]. For
compressing packets with IP/UDP/RTP headers CRTP [11] was created at
the same time. More recently the Robust Header Compression (ROHC)
working group has been developing a framework and profiles [12] for
compressing certain combinations of protocols like IP/UDP,
IP/UDP/RTP.
When using header compression the actual used overhead will be less
deterministic but in most cases an average overhead can be determined
for a certain application. If a network node knows that some type of
header compression is employed this can taken into consideration. To
be able to do this with any accuracy the application and packet rate
is needed.
4.4. RTCP problems
When RTCP is used between host in IPv4 and IPv6 networks over an NAT-
PT similar problems exist. The RTCP traffic going from the IPv4
domain will result in a higher RTCP bitrate than intended in the IPv6
domain due to the larger headers. This may result in up to 25%
increase in required bandwidth for the RTCP traffic. The largest
increase will be for small RTCP packets when the number of IPv4 hosts
are much larger than the number of IPv6 hosts. Fortunately as RTCP
has a limited bandwidth compared to RTP it will only result in a
maximum of 1.75% increase of the total session bandwidth when RTCP
bandwidth is 5% of RTP bandwidth. The increase in bandwidth will in
most cases be less.
At the same time this results in unfairness in the reporting between
an IPv4 and IPv6 node. The IPv6 node may report in the worst case
with 25% longer intervals.
4.5. Future development
Today there is work in IETF to design a new datagram transport
protocol, which will be suitable to use for real-time media. This
protocol is called the Datagram Congestion Control Protocol (DCCP).
This protocol will most probably have another header size than UDP,
which is mostly used today. This results in even further numbers of
possible transport combinations to calculate overhead for.
5. Problem Scope
The problems described in chapter 4 does effect all the protocols
that uses SDP to signal bandwidth parameters which contains transport
level information.
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In the MMUSIC WG there is work on a replacement of SDP called SDP-NG.
That work is RECOMMENDED to consider the problems outlined in this
draft when designing solutions for specifying bandwidth in SDP-NG.
6. Requirements
A solution to the problems outlined in this draft should meet the
following requirements:
- The bandwidth value SHALL be given in a way so that it can be
calculated for all possible combinations of transport overhead.
7. A Solution
7.1. Introduction
This chapter describes a solution for the problems outlined in this
document for the Application Specific (AS) bandwidth modifier.
The CT is a session level modifier and cannot easily be dealt with.
To address the problems with different overhead the CT value is
RECOMMENDED to be calculated using reasonable worst case overhead.
The RR and RS modifiers will hopefully be possible to clarify before
the publishing of their specification.
7.2. The TIAS bandwidth modifier
7.2.1. Usage
A new bandwidth modifier is defined to at least be used for the
following purposes:
- Resource reservation. A single bitrate can be enough to use for
resource reservation. Some characteristics can be derived from the
stream, codec type, etc. In some cases more information is needed,
then another SDP parameter will be required.
- Maximum media codec rate. With the definition below of "TIAS" the
given bitrate will mostly be from the media codec. Therefore it
gives a good indication on the maximum codec bitrate required to
be supported by the decoder.
- Communication bitrate required for the stream. The "TIAS" value
together with "maxprate" can be used to determine the maximum
communication bitrate the stream will require. By adding all
maximum bitrates from the streams in a session together, a
receiver can determine if its communication resources are
sufficient to handle the stream. For example a modem user can
determine if the session fits his modems capabilities and the
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established connection.
- Determine the RTP session bandwidth and derive the RTCP
bandwidth. The derived transport dependent attribute will be the
RTP session bandwidth in case of RTP based transport. The TIAS
value can also be used to determine the RTCP bandwidth to use when
using implicit allocation. RTP [6] defines that if not explicitly
stated, additional bandwidth shall be used by RTCP equal to 5% of
the RTP session bandwidth. The RTCP bandwidth can be explicitly
allocated by using what is defined in [9].
7.2.2. Definition
A new media level bandwidth modifier is defined:
b=TIAS:<bandwidth-value>
The Transport Independent Application Specific Maximum (TIAS)
bandwidth modifier has an integer bitrate value in bits per second. A
fractional bandwidth value SHALL always be rounded up to the next
integer. The bandwidth value is the maximum needed by the application
without counting IP or transport layers. For RTP based applications,
TIAS can be used to derive the RTP "session bandwidth" as defined in
section 6.2 of [6]. However in the context of RTP transport the TIAS
value is defined as:
Only the RTP payload with its media SHALL be used in the
calculation of the bitrate, i.e. the lower layers (IP/UDP) and RTP
headers are excluded. This will allow one to use the same value
for both RTP based media transport as non-RTP transport and stored
media.
Note 1: The usage of bps is not in accordance with RFC 2327 [1]. This
change has no implications on the parser, only the interpreter of the
value must be aware. The change is done to allow for better
resolution and has also been used for the RR and RS bandwidth
modifiers, see [9].
Note 2: RTCP bandwidth is not included in the bandwidth value. In
applications using RTCP, the bandwidth used by RTCP is either 5% of
the RTP session bandwidth including lower layers or as specified by
the RR and RS modifiers [9]. To assign an equal amount of RTCP
bandwidth to user of either IPv4 or IPv6, a special rule is defined
below in chapter 7.2.4 used to derive RTCP bandwidth.
7.2.3. Usage Rules
To allow for backwards compatibility towards users of SDP that does
not implement "TIAS", it is RECOMMENDED to also include the "AS"
modifier when using "TIAS". The presence of any value, even with
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problems is better than none. However an SDP user understanding TIAS
when present SHOULD ignore the "AS" value and use TIAS instead.
When using TIAS for an RTP transported stream the "maxprate"
attribute SHOULD be included. The "avgprate" MAY also be included. If
the "avgprate" is included the "maxprate" MUST be included.
7.2.4. Deriving RTCP bandwidth
To fairly assign RTCP bandwidth to host when both IPv4 and IPv6 hosts
are communicating a special method for RTCP bandwidth is here
defined. This consist of both a special method for how to calculate
the bandwidth for RTCP corresponding to the normal 5% and what to do
when calculating the deterministic RTCP interval.
The below defined methods SHALL be used for RTP based transport when
done over IP and the bandwidth of the session is signaled with
"TIAS". The actual transport layer, UDP etc., does not matter but
will taken into consideration in the calculation. It assumes that all
communicating host uses the same transport protocols with exception
for IP version.
To determine the RTCP bandwidth the transport dependent bitrate is
calculated, in accordance with chapter 7.4, using the actual
transport layers used with the exception that the IP version used
MUST be IPv6. No extension headers SHALL be taken into consideration.
So an IPv4 host sending with RTP/UDP will for the RTCP calculation
determine a transport dependent bitrate based on RTP/UDP/IPv6
headers. The RTCP bitrate is then equal to 5% of the calculated
value.
When determining the RTCP sender interval the following changes shall
be made in the calculation. The RTCP packet size used to update the
RTCP variable "avg_rtcp_size" (Section 6.3.3 in [6]) SHALL be include
the size of an IPv6 header and not IPv4 if used. This shall be done
independent of the reason why the "avg_rtcp_size" is updated. When
initializing the "avg_rtcp_size" the size of an IPv6 SHALL be used.
By calculating in the RTCP sending rules that the RTCP sender always
uses IPv6 there will be fairness between IPv4 and IPv6 hosts.
7.3. Packet Rate parameters
To be able to calculate the bandwidth value including the actually
used lower layers two packet rate parameters is also defined.
The maximum packet rate parameter is defined as:
a=maxprate:<packet-rate>
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The <packet-rate> is a floating-point value for the streams maximum
packet rate. If the number of packets is variable the given value
SHALL be the maximum the application, can produce in case of live
stream, or for on-demand streams have produced. The value is
calculated by taking the largest value a sliding window 1 second long
produces when moved over the entire media stream. In cases that this
can't be calculated, i.e. for example a live stream, a estimated
value of the maximum rate the codec can produce for the given
configuration and content SHALL be used.
An average packet rate is defined as:
a=avgprate:<packet-rate>
The <packet-rate> is a floating point value of the average packet
rate for the stream. The average value SHOULD be calculated as an
average value over the whole stream. If not possible to calculate the
value supplied SHALL be an estimate of the most likely packet rate to
be used.
The "maxprate" and "avgprate" attribute are media level SDP
attributes.
7.4. Converting to Transport Dependent values
When converting the transport independent bitrate value (bw-value)
into a transport dependent value including the lower layers the
following MUST be done:
1. Determine which lower layers that will be used and calculate the
sum of the sizes of the headers in bits (h-size). In cases of
variable header sizes the average size SHOULD be used.
2. Retrieve the packet rate to use from the SDP (prate = maxprate or
avgprate).
3. Calculate the transport overhead by multiplying the header sizes
with the packet rate (t-over = h-size * prate).
4. Round the transport overhead up to nearest integer in bits(t-over
= CEIL(t-over)).
5. Add the transport overhead to the transport independent bitrate
value (bitrate = bw-value + t-over)
When the above calculation is performed using the "maxprate" the
bitrate value should be the absolute maximum the media stream will
use over the transport used in the calculations.
When calculated using the "avgprate" the value is derived is the
maximum value that will be needed for the average packetization.
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7.5. ABNF definitions
This chapter defines in ABNF from RFC 2234 [8] the bandwidth modifier
and the packet rate attribute.
The bandwidth modifier:
TIAS-bandwidth = "b" "=" "TIAS" ":" bandwidth-value
bandwidth-value = 1*DIGIT
The maximum packet rate attribute:
max-p-rate-def = "a" "=" "maxprate" ":" packet-rate CRLF
The average packet rate attribute:
avg-p-rate-def = "a" "=" "avgprate" ":" packet-rate CRLF
packet-rate = 1*DIGIT ["." 1*DIGIT]
8. Protocol Interaction
8.1. RTSP
The "TIAS", "maxprate" and "avgprate" can today be used with RTSP. To
be able to calculate the transport dependent bandwidth, some of the
transport header parameters will be required. There should be no
problems for a client to calculate the required bandwidth(s) prior to
a RTSP SETUP. The reason is that a client supports a limited number
of transport setups when the SDP "m=" line is taken into account. The
"m=" line will signal to the client the desired transport profiles.
8.2. SIP
The usage of "TIAS" together with "maxprate" should not be different
from the handling of the "AS" modifier currently in use. The needed
transport parameters will available in the transport field in the
"m=" line. The address class can be determined from the "c=" field
and the clients connectivity.
8.3. SAP
In the case of SAP all available information to calculate the
transport dependent bitrate should be present in the SDP. The "c="
information gives the used address family for the multicast. The
transport layer, e.g. RTP/UDP, for each media is evident in the media
line ("m=") and its transport field.
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9. Security Consideration
The bandwidth value that is supplied by the parameters defined here
can, if not protected, be altered. By altering the bandwidth value
one can fool a receiver to reserve either more or less bandwidth than
actually needed. Reserving too much may result in unwanted expenses
on behalf of user and also blocking of resources that other parties
could have used. If to little bandwidth is reserved the receiving
users quality MAY be effected. Trusting a to large TIAS value may
also result in that the receiver will turn down the session due to
insufficient communication and decoding resources.
Due to these security risks it is RECOMMENDED that the SDP is
authenticated so no tampering can be performed. It is also
RECOMMENDED that any receiver of the SDP performs an analysis of the
received bandwidth values so that they are reasonable and is what can
be expected for the application. For example an AMR encoded voice
stream claiming to use 1000 kbps is not reasonable.
10. IANA Consideration
This document register two new SDP media level attribute "maxprate"
and "avgprate", see section 7.3.
A new bandwidth modifier "TIAS" is also registered in accordance with
the rules requiring a standard tracks RFC. The modifier is defined in
section 7.2.
11. Acknowledgments
The author would like to thank Gonzalo Camarillo and Hesham Soliman
for their work reviewing this document.
The author would also like to thank all persons on the MMUSIC working
group's mailing list that has commented on this specification.
12. Author's Addresses
Magnus Westerlund Tel: +46 8 4048287
Ericsson Research Email: Magnus.Westerlund@ericsson.com
Ericsson AB
Torshamnsgatan 23
SE-164 80 Stockholm, SWEDEN
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13. References
13.1. Normative references
[1] M. Handley, V. Jacobson, "Session Description Protocol (SDP)",
IETF RFC 2327, April 1998.
[2] M. Handley et al., "Session Announcement Protocol", IETF RFC
2974, October 2000.
[3] J. Rosenberg, et. al., "SIP: Session Initiation Protocol", IETF
RFC 3261, June 2002.
[4] J. Rosenberg, H. Schulzrine, "An Offer/Answer Model with Session
Description Protocol (SDP)", IETF RFC 3164, June 2002.
[5] H. Schulzrinne, et. al., "Real Time Streaming Protocol (RTSP)",
IETF RFC 2326, April 1998.
[6] H. Schulzrinne, et. al., "RTP: A Transport Protocol for Real-
Time Applications", IETF RFC 1889, January 1996.
[7] S. Bradner, "Key words for use in RFCs to Indicate Requirement
Levels", RFC 2119, March 1997.
[8] D. Crocker and P. Overell, "Augmented BNF for syntax specifica-
tions: ABNF," RFC 2234, Internet Engineering Task Force, Nov.
1997.
13.2. Informative References
[9] S. Casner, "SDP Bandwidth Modifiers for RTCP Bandwidth", IETF WG
draft, draft-ietf-avt-rtcp-bw-05.txt, November 2001, Work in
progress
[10] M. Degermark, B. Nordgren, S. Pink, "IP Header Compression",
IETF RFC 2507, February 1999.
[11] S. Casner, V. Jacobson, "Compressing IP/UDP/RTP Headers for Low-
Speed Serial Links", IETF RFC 2508, February 1999.
[12] C. Bormann, et. al., "RObust Header Compression (ROHC):
Framework and four profiles", IETF RFC 3095, July 2001.
[13] Tsirtsis, G. and Srisuresh, P., "Network Address Translation -
Protocol Translation (NAT-PT)", RFC 2766, Internet Engineering
Task Force, February 2000.
[14] S. Deering and R. Hinden, "Internet Protocol, Version 6 (IPv6)
Specification", RFC 2460, Internet Engineering Task Force,
December 1998.
[15] J. Postel, "internet protocol", RFC 791, Internet Engineering
Task Force, September 1981.
This Internet-Draft expires in April 2003.
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