Internet DRAFT - draft-ietf-dccp-rtp
draft-ietf-dccp-rtp
Network Working Group C. Perkins
Internet-Draft University of Glasgow
Intended status: Standards Track June 20, 2007
Expires: December 22, 2007
RTP and the Datagram Congestion Control Protocol (DCCP)
draft-ietf-dccp-rtp-07.txt
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Copyright Notice
Copyright (C) The IETF Trust (2007).
Abstract
The Real-time Transport Protocol (RTP) is a widely used transport for
real-time multimedia on IP networks. The Datagram Congestion Control
Protocol (DCCP) is a newly defined transport protocol that provides
desirable services for real-time applications. This memo specifies a
mapping of RTP onto DCCP, along with associated signalling, such that
real-time applications can make use of the services provided by DCCP.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Rationale . . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Conventions Used in this Memo . . . . . . . . . . . . . . . . 4
4. RTP over DCCP: Framing . . . . . . . . . . . . . . . . . . . . 4
4.1. RTP Data Packets . . . . . . . . . . . . . . . . . . . . . 4
4.2. RTP Control Packets . . . . . . . . . . . . . . . . . . . 5
4.3. Multiplexing Data and Control . . . . . . . . . . . . . . 6
4.4. RTP Sessions and DCCP Connections . . . . . . . . . . . . 7
4.5. RTP Profiles . . . . . . . . . . . . . . . . . . . . . . . 7
5. RTP over DCCP: Signalling using SDP . . . . . . . . . . . . . 8
5.1. Protocol Identification . . . . . . . . . . . . . . . . . 8
5.2. Service Codes . . . . . . . . . . . . . . . . . . . . . . 9
5.3. Connection Management . . . . . . . . . . . . . . . . . . 11
5.4. Multiplexing Data and Control . . . . . . . . . . . . . . 11
5.5. Example . . . . . . . . . . . . . . . . . . . . . . . . . 11
6. Security Considerations . . . . . . . . . . . . . . . . . . . 12
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 13
8. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 14
9. References . . . . . . . . . . . . . . . . . . . . . . . . . . 14
9.1. Normative References . . . . . . . . . . . . . . . . . . . 14
9.2. Informative References . . . . . . . . . . . . . . . . . . 15
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 16
Intellectual Property and Copyright Statements . . . . . . . . . . 17
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1. Introduction
The Real-time Transport Protocol (RTP) [1] is widely used in video
streaming, telephony, and other real-time networked applications.
RTP can run over a range of lower-layer transport protocols, and the
performance of an application using RTP is heavily influenced by the
choice of lower-layer transport. The Datagram Congestion Control
Protocol (DCCP) [2] is a newly specified transport protocol that
provides desirable properties for real-time applications running on
unmanaged best-effort IP networks. This memo describes how RTP can
be framed for transport using DCCP, and discusses some of the
implications of such a framing. It also describes how the Session
Description Protocol (SDP) [3] can be used to signal such sessions.
The remainder of this memo is structured as follows: it begins with a
rationale for the work in Section 2, describing why a mapping of RTP
onto DCCP is needed. Following a description of the conventions used
in this memo in Section 3, the specification begins in Section 4 with
the definition of how RTP packets are framed within DCCP. Associated
signalling is described in Section 5. Security considerations are
discussed in Section 6, and IANA considerations in Section 7.
2. Rationale
With the widespread adoption of RTP have come concerns that many real
time applications do not implement congestion control, leading to the
potential for congestion collapse of the network [15]. The designers
of RTP recognised this issue, stating that [4]:
If best-effort service is being used, RTP receivers SHOULD monitor
packet loss to ensure that the packet loss rate is within
acceptable parameters. Packet loss is considered acceptable if a
TCP flow across the same network path and experiencing the same
network conditions would achieve an average throughput, measured
on a reasonable time-scale, that is not less than the RTP flow is
achieving. This condition can be satisfied by implementing
congestion control mechanisms to adapt the transmission rate (or
the number of layers subscribed for a layered multicast session),
or by arranging for a receiver to leave the session if the loss
rate is unacceptably high.
While the goals are clear, the development of TCP friendly congestion
control that can be used with RTP and real-time media applications is
an open research question with many proposals for new algorithms, but
little deployment experience.
Two approaches have been used to provide congestion control for RTP:
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1) develop RTP extensions that incorporate congestion control; and 2)
provide mechanisms for running RTP over congestion controlled
transport protocols. An example of the first approach can be found
in [16], extending RTP to incorporate feedback information such that
TFRC congestion control [17] can be implemented at the application
level. This will allow congestion control to be added to existing
applications without operating system or network support, and it
offers the flexibility to experiment with new congestion control
algorithms as they are developed. Unfortunately, it also passes the
complexity of implementing congestion control onto application
authors, a burden which many would prefer to avoid.
The second approach is to run RTP on a lower-layer transport protocol
that provides congestion control. One possibility is to run RTP over
TCP, as defined in [5], but the reliable nature of TCP and the
dynamics of its congestion control algorithm make this inappropriate
for most interactive real time applications (the Stream Control
Transmission Protocol (SCTP) is inappropriate for similar reasons).
A better fit for such applications may be to run RTP over DCCP, since
DCCP offers unreliable packet delivery and a choice of congestion
control. This gives applications the ability to tailor the transport
to their needs, taking advantage of better congestion control
algorithms as they come available, while passing complexity of
implementation to the operating system. If DCCP should come to be
widely available, it is believed these will be compelling advantages.
Accordingly, this memo defines a mapping of RTP onto DCCP.
3. Conventions Used in this Memo
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 [6].
4. RTP over DCCP: Framing
The following section defines how RTP and RTCP packets can be framed
for transport using DCCP. It also describes the differences between
RTP sessions and DCCP connections, and the impact these have on the
design of applications.
4.1. RTP Data Packets
Each RTP data packet MUST be conveyed in a single DCCP datagram.
Fields in the RTP header MUST be interpreted according to the RTP
specification, and any applicable RTP Profile and Payload Format.
Header processing is not affected by DCCP framing (in particular,
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note that the semantics of the RTP sequence number and the DCCP
sequence number are not compatible, and the value of one cannot be
inferred from the other).
A DCCP connection is opened when an end system joins an RTP session,
and it remains open for the duration of the session. To ensure NAT
bindings are kept open, an end system SHOULD send a zero length DCCP-
Data packet once every 15 seconds during periods when it has no other
data to send. This removes the need for RTP no-op packets [18], and
similar application level keep-alives, when using RTP over DCCP.
This application level keepalive does not need to be sent if it is
known that the DCCP CCID in use provides a transport level keepalive,
or if the application can determine that there are no NAT devices on
the path.
RTP data packets MUST obey the dictates of DCCP congestion control.
In some cases, the congestion control will require a sender to send
at a rate below that which the payload format would otherwise use.
To support this, an application could use either a rate adaptive
payload format, or a range of payload formats (allowing it to switch
to a lower rate format if necessary). Details of the rate adaptation
policy for particular payload formats are outside the scope of this
memo (but see [19] and [20] for guidance).
RTP extensions that provide application-level congestion control
(e.g. [16]) will conflict with DCCP congestion control, and MUST NOT
be used.
DCCP allows an application to choose the checksum coverage, using a
partial checksum to allow an application to receive packets with
corrupt payloads. Some RTP Payload Formats (e.g. [21]) can make use
of this feature in conjunction with payload-specific mechanisms to
improve performance when operating in environments with frequent non-
congestive packet corruption. If such a payload format is used, an
RTP end system MAY enable partial checksums at the DCCP layer, in
which case the checksum MUST cover at least the DCCP and RTP headers
to ensure packets are correctly delivered. Partial checksums MUST
NOT be used unless supported by mechanisms in the RTP payload format.
4.2. RTP Control Packets
The RTP Control Protocol (RTCP) is used in the standard manner with
DCCP. RTCP packets are grouped into compound packets, as described
in Section 6.1 of [1], and each compound RTCP packet is transported
in a single DCCP datagram.
The usual RTCP timing rules apply, with the additional constraint
that RTCP packets MUST obey the DCCP congestion control algorithm
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negotiated for the connection. This can prevent a participant from
sending an RTCP packet at the expiration of the RTCP transmission
timer if there is insufficient network capacity available. In such
cases the RTCP packet is delayed and sent at the earliest possible
instant when capacity becomes available. The actual time the RTCP
packet was sent is then used as the basis for calculating the next
RTCP transmission time.
RTCP packets comprise only a small fraction of the total traffic in
an RTP session. Accordingly, it is expected that delays in their
transmission due to congestion control will not be common, provided
the configured nominal "session bandwidth" (see Section 6.2 of [1])
is in line with the bandwidth achievable on the DCCP connection. If,
however, the capacity of the DCCP connection is significantly below
the nominal session bandwidth, RTCP packets may be delayed enough for
participants to time out due to apparent inactivity. In such cases,
the session parameters SHOULD be re-negotiated to more closely match
the available capacity, for example by performing a re-invite with an
updated "b=" line when using the Session Initiation Protocol [22] for
signalling.
Note: Since the nominal session bandwidth is chosen based on media
codec capabilities, a session where the nominal bandwidth is much
larger than the available bandwidth will likely become unusable
due to constraints on the media channel, and so require
negotiation of a lower bandwidth codec, before it becomes unusable
due to constraints on the RTCP channel.
As noted in Section 17.1 of [2], there is the potential for overlap
between information conveyed in RTCP packets and that conveyed in
DCCP acknowledgement options. In general this is not an issue since
RTCP packets contain media-specific data that is not present in DCCP
acknowledgement options, and DCCP options contain network-level data
that is not present in RTCP. Indeed, there is no overlap between the
five RTCP packet types defined in the RTP specification [1] and the
standard DCCP options [2]. There are, however, cases where overlap
does occur: most clearly between the optional RTCP Extended Reports
Loss RLE Blocks [23] and the DCCP Ack Vector option. If there is
overlap between RTCP report packets and DCCP acknowledgements, an
application SHOULD use either RTCP feedback or DCCP acknowledgements,
but not both (use of both types of feedback will waste available
network capacity, but is not otherwise harmful).
4.3. Multiplexing Data and Control
The obvious mapping of RTP onto DCCP creates two DCCP connections for
each RTP flow: one for RTP data packets, one for RTP control packets.
A frequent criticism of RTP relates to the number of ports it uses,
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since large telephony gateways can support more than 32768 RTP flows
between pairs of gateways, and so run out of UDP ports. In addition,
use of multiple ports complicates NAT traversal. For these reasons,
it is RECOMMENDED that the RTP and RTCP traffic for a single RTP
session is multiplexed onto a single DCCP connection following the
guidelines in [7], where possible (it may not be possible in all
circumstances, for example when translating from an RTP stream over a
non-DCCP transport that uses conflicting RTP payload types and RTCP
packet types).
4.4. RTP Sessions and DCCP Connections
An end system SHOULD NOT assume that it will observe only a single
RTP synchronisation source (SSRC) because it is using DCCP framing.
An RTP session can span any number of transport connections, and can
include RTP mixers or translators bringing other participants into
the session. The use of a unicast DCCP connection does not imply
that the RTP session will have only two participants, and RTP end
systems SHOULD assume that multiple synchronisation sources may be
observed when using RTP over DCCP, unless otherwise signalled.
An RTP translator bridging multiple DCCP connections to form a single
RTP session needs to be aware of the congestion state of each DCCP
connection, and must adapt the media to the available capacity of
each. The Codec Control Messages defined in [24] may be used to
signal congestion state to the media senders, allowing them to adapt
their transmission. Alternatively, media transcoding may be used to
perform adaptation: this is computationally expensive, induces delay,
and generally gives poor quality results. Depending on the payload,
it might be possible to use some form of scalable coding. Scalable
media coding formats are an active research area, and are not in
widespread use at the time of this writing.
A single RTP session may also span a DCCP connection and some other
type of transport connection. An example might be an RTP over DCCP
connection from an RTP end system to an RTP translator, with an RTP
over UDP/IP multicast group on the other side of the translator. A
second example might be an RTP over DCCP connection that links PSTN
gateways. The issues for such an RTP translator are similar to those
when linking two DCCP connections, except that the congestion control
algorithms on either side of the translator may not be compatible.
Implementation of effective translators for such an environment is
non-trivial.
4.5. RTP Profiles
In general, there is no conflict between new RTP Profiles and DCCP
framing, and most RTP profiles can be negotiated for use over DCCP
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with the following exceptions:
o An RTP profile that is intolerant of packet corruption may
conflict with the DCCP partial checksum feature. An example of
this is the integrity protection provided by the RTP/SAVP profile,
which cannot be used in conjunction with DCCP partial checksums.
o An RTP profile that mandates a particular non-DCCP lower layer
transport will conflict with DCCP.
RTP profiles which fall under these exceptions SHOULD NOT be used
with DCCP unless the conflicting features can be disabled.
Of the profiles currently defined, the RTP Profile for Audio and
Video Conferences with Minimal Control [4], the Secure Real-time
Transport Protocol [8], the Extended RTP Profile for RTCP-based
Feedback [9], and the Extended Secure RTP Profile for RTCP-based
Feedback [10] MAY be used with DCCP (noting the potential conflict
between DCCP partial checksums and the integrity protection provided
by the secure RTP variants -- see Section 6).
5. RTP over DCCP: Signalling using SDP
The Session Description Protocol (SDP) [3] and the offer/answer model
[11] are widely used to negotiate RTP sessions (for example, using
the Session Initiation Protocol [22]). This section describes how
SDP is used to signal RTP sessions running over DCCP.
5.1. Protocol Identification
SDP uses a media ("m=") line to convey details of the media format
and transport protocol used. The ABNF syntax of a media line is as
follows (from [3]):
media-field = %x6d "=" media SP port ["/" integer] SP proto
1*(SP fmt) CRLF
The proto field denotes the transport protocol used for the media,
while the port indicates the transport port to which the media is
sent. Following [5] and [12] this memo defines the following five
values of the proto field to indicate media transported using DCCP:
DCCP
DCCP/RTP/AVP
DCCP/RTP/SAVP
DCCP/RTP/AVPF
DCCP/RTP/SAVPF
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The "DCCP" protocol identifier is similar to the "UDP" and "TCP"
protocol identifiers and denotes the DCCP transport protocol [2], but
not its upper-layer protocol. An SDP "m=" line that specifies the
"DCCP" protocol MUST further qualify the application layer protocol
using a "fmt" identifier (the "fmt" namespace is managed in the same
manner as for the "UDP" protocol identifier). A single DCCP port is
used, as denoted by the port field in the media line. The "DCCP"
protocol identifier MUST NOT be used to signal RTP sessions running
over DCCP; those sessions MUST use a protocol identifier of the form
"DCCP/RTP/..." as described below.
The "DCCP/RTP/AVP" protocol identifier refers to RTP using the RTP
Profile for Audio and Video Conferences with Minimal Control [4]
running over DCCP.
The "DCCP/RTP/SAVP" protocol identifier refers to RTP using the
Secure Real-time Transport Protocol [8] running over DCCP.
The "DCCP/RTP/AVPF" protocol identifier refers to RTP using the
Extended RTP Profile for RTCP-based Feedback [9] running over DCCP.
The "DCCP/RTP/SAVPF" protocol identifier refers to RTP using the
Extended Secure RTP Profile for RTCP-based Feedback [10] running over
DCCP.
RTP payload formats used with the "DCCP/RTP/AVP", "DCCP/RTP/SAVP",
"DCCP/RTP/AVPF" and "DCCP/RTP/SAVPF" protocol identifiers MUST use
the payload type number as their "fmt" value. If the payload type
number is dynamically assigned, an additional "rtpmap" attribute MUST
be included to specify the format name and parameters as defined by
the media type registration for the payload format.
DCCP port 5004 is registered for use by the RTP profiles listed
above, and SHOULD be the default port chosen by applications using
those profiles. If multiple RTP sessions are active from a host,
even numbered ports in the dynamic range SHOULD be used for the other
sessions. If RTCP is to be sent on a separate DCCP connection to
RTP, the RTCP connection SHOULD use the next higher destination port
number, unless an alternative DCCP port is signalled using the
"a=rtcp:" attribute [13]. For improved interoperability, "a=rtcp:"
SHOULD be used whenever an alternate DCCP port is used.
5.2. Service Codes
In addition to the port number, specified on the SDP "m=" line, a
DCCP connection has an associated service code. A single new SDP
attribute ("dccp-service-code") is defined to signal the DCCP service
code according to the following ABNF [14]:
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dccp-service-attr = %x61 "=dccp-service-code:" service-code
service-code = hex-sc / decimal-sc / ascii-sc
hex-sc = %x53 %x43 "=" %x78 *HEXDIG
decimal-sc = %x53 %x43 "=" *DIGIT
ascii-sc = %x53 %x43 ":" *sc-char
sc-char = %d42-43 / %d45-47 / %d63-90 / %d95 / %d97-122
where DIGIT and HEXDIG are as defined in [14]. The service code is
interpreted as defined in Section 8.1.2 of [2] and may be specified
using either the hexadecimal, decimal, or ASCII formats. A parser
MUST interpret service codes according to their numeric value,
indpendent of the format used to represent them in SDP.
The following DCCP service codes are registered for use with RTP:
o SC:RTPA (equivalently SC=1381257281 or SC=x52545041): an RTP
session conveying audio data (and OPTIONAL multiplexed RTCP)
o SC:RTPV (equivalently SC=1381257302 or SC=x52545056): an RTP
session conveying video data (and OPTIONAL multiplexed RTCP)
o SC:RTPT (equivalently SC=1381257300 or SC=x52545054): an RTP
session conveying text media (and OPTIONAL multiplexed RTCP)
o SC:RTPO (equivalently SC=1381257295 or SC=x5254504f): an RTP
session conveying any other type of media (and OPTIONAL
multiplexed RTCP)
o SC:RTCP (equivalently SC=1381253968 or SC=x52544350): an RTCP
connection, separate from the corresponding RTP
To ease the job of middleboxes, applications SHOULD use these service
codes to identify RTP sessions running within DCCP. The service code
SHOULD match the top-level media type signalled for the session (i.e.
the SDP "m=" line), with the exception connections using media types
other than audio, video, or text which use SC:RTPO, and connections
that transport only RTCP packets, which use SC:RTCP.
The "a=dccp-service-code:" attribute is a media level attribute which
is not subject to the charset attribute.
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5.3. Connection Management
The "a=setup:" attribute indicates which of the end points should
initiate the DCCP connection establishment (i.e. send the initial
DCCP-Request packet). The "a=setup:" attribute MUST be used in a
manner comparable with [12], except that DCCP connections are being
initiated rather than TCP connections.
After the initial offer/answer exchange, the end points may decide to
re-negotiate various parameters. The "a=connection:" attribute MUST
be used in a manner compatible with [12] to decide whether a new DCCP
connection needs to be established as a result of subsequent offer/
answer exchanges, or if the existing connection should still be used.
5.4. Multiplexing Data and Control
A single DCCP connection can be used to transport multiplexed RTP and
RTCP packets. Such multiplexing MUST be signalled using an "a=rtcp-
mux" attribute according to [7]. If multiplexed RTP and RTCP is not
to be used, then the "a=rtcp-mux" attribute MUST NOT be present in
the SDP offer, and a separate DCCP connection MUST be opened to
transport the RTCP data on a different DCCP port.
5.5. Example
An offerer at 192.0.2.47 signals its availability for an H.261 video
session, using RTP/AVP over DCCP with service code "RTPV" (using the
hexadecimal encoding of the service code in the SDP). RTP and RTCP
packets are multiplexed onto a single DCCP connection:
v=0
o=alice 1129377363 1 IN IP4 192.0.2.47
s=-
c=IN IP4 192.0.2.47
t=0 0
m=video 5004 DCCP/RTP/AVP 99
a=rtcp-mux
a=rtpmap:99 h261/90000
a=dccp-service-code:SC=x52545056
a=setup:passive
a=connection:new
An answerer at 192.0.2.128 receives this offer and responds with the
following answer:
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v=0
o=bob 1129377364 1 IN IP4 192.0.2.128
s=-
c=IN IP4 192.0.2.128
t=0 0
m=video 9 DCCP/RTP/AVP 99
a=rtcp-mux
a=rtpmap:99 h261/90000
a=dccp-service-code:SC:RTPV
a=setup:active
a=connection:new
The end point at 192.0.2.128 then initiates a DCCP connection to port
5004 at 192.0.2.47. DCCP port 5004 is used for both the RTP and RTCP
data, and port 5005 is unused. The textual encoding of the service
code is used in the answer, and represents the same service code as
in the offer.
6. Security Considerations
The security considerations in the RTP specification [1] and any
applicable RTP profile (e.g. [4], [8], [9], or [10]) or payload
format apply when transporting RTP over DCCP.
The security considerations in the DCCP specification [2] apply.
The SDP signalling described in Section 5 is subject to the security
considerations of [3], [11], [12], [5], and [7].
The provision of effective congestion control for RTP through use of
DCCP is expected to help reduce the potential for denial-of-service
present when RTP flows ignore the advice in [1] to monitor packet
loss and reduce their sending rate in the face of persistent
congestion.
There is a potential conflict between the Secure RTP Profiles [8],
[10] and the DCCP partial checksum option, since these profiles
introduce, and recommend the use of, message authentication for RTP
and RTCP packets. Message authentication codes of the type used by
these profiles cannot be used with partial checksums, since any bit-
error in the DCCP packet payload will cause the authentication check
to fail. Accordingly, DCCP partial checksums SHOULD NOT be used in
conjunction with SRTP authentication. The confidentiality features
of the basic RTP specification cannot be used with DCCP partial
checksums, since bit errors propagate. Also, despite the fact that
bit errors do not propagate when using AES in counter mode, the
Secure RTP profiles SHOULD NOT be used with DCCP partial checksums,
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since it requires authentication for security, and authentication is
incompatible with partial checksums.
7. IANA Considerations
[Note to RFC Editor: please replace "RFC xxxx" below with the RFC
number of this memo, and then remove this note].
The following SDP "proto" field identifiers are to be registered (see
Section 5.1):
Type SDP Name Reference
---- -------- ---------
proto DCCP [RFC xxxx]
DCCP/RTP/AVP [RFC xxxx]
DCCP/RTP/SAVP [RFC xxxx]
DCCP/RTP/AVPF [RFC xxxx]
DCCP/RTP/SAVPF [RFC xxxx]
The following new SDP attribute ("att-field") is to be registered:
Contact name: Colin Perkins <csp@csperkins.org>
Attribute name: dccp-service-code
Long-form attribute name in English: DCCP service code
Type of attribute: Media level.
Subject to the charset attribute? No.
Purpose of the attribute: see RFC xxxx Section 5.2
Allowed attribute values: see RFC xxxx Section 5.2
The following DCCP service code values are to be registered (see
Section 5.2):
1381257281 RTPA RTP audio [RFC xxxx]
1381257302 RTPV RTP video [RFC xxxx]
1381257300 RTPT RTP text [RFC xxxx]
1381257295 RTPO RTP (unspecified media type) [RFC xxxx]
1381253968 RTCP RTP control protocol (RTCP) [RFC xxxx]
The following DCCP ports are to be registered (see Section 5.1):
avt-profile-1 5004/dccp RTP media data [RFC 3551, RFC xxxx]
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avt-profile-2 5005/dccp RTP control protocol [RFC 3551, RFC xxxx]
Note: ports 5004/tcp, 5004/udp, 5005/tcp, and 5005/udp have existing
registrations, but incorrect description and reference. The IANA is
requested to update the existing registrations as follows:
avt-profile-1 5004/tcp RTP media data [RFC 3551, RFC 4571]
avt-profile-1 5004/udp RTP media data [RFC 3551]
avt-profile-2 5005/tcp RTP control protocol [RFC 3551, RFC 4571]
avt-profile-2 5005/udp RTP control protocol [RFC 3551]
8. Acknowledgements
This work was supported in part by the UK Engineering and Physical
Sciences Research Council. Thanks are due to to Philippe Gentric,
Magnus Westerlund, Sally Floyd, Dan Wing, Gorry Fairhurst, Stephane
Bortzmeyer, Arjuna Sathiaseelan, Tom Phelan, Lars Eggert, Eddie
Kohler, Miguel Garcia, and the other members of the DCCP working
group for their comments.
9. References
9.1. Normative References
[1] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson,
"RTP: A Transport Protocol for Real-Time Applications", STD 64,
RFC 3550, July 2003.
[2] Kohler, E., Handley, M., and S. Floyd, "Datagram Congestion
Control Protocol (DCCP)", RFC 4340, March 2006.
[3] Handley, M., Jacobson, V., and CS. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.
[4] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and Video
Conferences with Minimal Control", STD 65, RFC 3551, July 2003.
[5] Lazzaro, J., "Framing RTP and RTCP Packets over Connection-
Oriented Transport", RFC 4571, June 2006.
[6] Bradner, S., "Key words for use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, March 1997.
[7] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
Control Packets on a Single Port",
draft-ietf-avt-rtp-and-rtcp-mux-05 (work in progress),
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Internet-Draft RTP over DCCP June 2007
May 2007.
[8] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004.
[9] Ott, J., Wenger, S., Sato, N., and C. Burmeister, "Extended RTP
Profile for RTCP-based Feedback(RTP/AVPF)", RFC 4585,
June 2006.
[10] Ott, J. and E. Carrara, "Extended Secure RTP Profile for RTCP-
based Feedback (RTP/SAVPF)", draft-ietf-avt-profile-savpf-10
(work in progress), February 2007.
[11] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with
Session Description Protocol (SDP)", RFC 3264, June 2002.
[12] Yon, D. and G. Camarillo, "TCP-Based Media Transport in the
Session Description Protocol (SDP)", RFC 4145, September 2005.
[13] Huitema, C., "Real Time Control Protocol (RTCP) attribute in
Session Description Protocol (SDP)", RFC 3605, October 2003.
[14] Crocker, D., Ed. and P. Overell, "Augmented BNF for Syntax
Specifications: ABNF", RFC 4234, October 2005.
9.2. Informative References
[15] Floyd, S. and J. Kempf, "IAB Concerns Regarding Congestion
Control for Voice Traffic in the Internet", RFC 3714,
March 2004.
[16] Gharai, L., "RTP with TCP Friendly Rate Control",
draft-ietf-avt-tfrc-profile-07 (work in progress), March 2007.
[17] Handley, M., Floyd, S., Padhye, J., and J. Widmer, "TCP
Friendly Rate Control (TFRC): Protocol Specification",
RFC 3448, January 2003.
[18] Andreasen, F., Oran, D., and D. Wing, "A No-Op Payload Format
for RTP", draft-ietf-avt-rtp-no-op-03 (work in progress),
April 2007.
[19] Phelan, T., "Strategies for Streaming Media Applications Using
TCP-Friendly Rate Control", draft-ietf-dccp-tfrc-media-01
(work in progress), October 2005.
[20] Phelan, T., "Datagram Congestion Control Protocol (DCCP) User
Perkins Expires December 22, 2007 [Page 15]
Internet-Draft RTP over DCCP June 2007
Guide", draft-ietf-dccp-user-guide-03 (work in progress),
April 2005.
[21] Sjoberg, J., Westerlund, M., Lakaniemi, A., and Q. Xie, "RTP
Payload Format and File Storage Format for the Adaptive Multi-
Rate (AMR) and Adaptive Multi-Rate Wideband (AMR-WB) Audio
Codecs", RFC 4867, April 2007.
[22] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A.,
Peterson, J., Sparks, R., Handley, M., and E. Schooler, "SIP:
Session Initiation Protocol", RFC 3261, June 2002.
[23] Friedman, T., Caceres, R., and A. Clark, "RTP Control Protocol
Extended Reports (RTCP XR)", RFC 3611, November 2003.
[24] Wenger, S., "Codec Control Messages in the RTP Audio-Visual
Profile with Feedback (AVPF)", draft-ietf-avt-avpf-ccm-05
(work in progress), May 2007.
Author's Address
Colin Perkins
University of Glasgow
Department of Computing Science
17 Lilybank Gardens
Glasgow G12 8QQ
UK
Email: csp@csperkins.org
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Perkins Expires December 22, 2007 [Page 17]
