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Network Working Group J. Lazzaro
Request for Comments: 4571 UC Berkeley
Category: Standards Track July 2006
Framing Real-time Transport Protocol (RTP)
and RTP Control Protocol (RTCP) Packets
over Connection-Oriented Transport
Status of This Memo
This document specifies an Internet standards track protocol for the
Internet community, and requests discussion and suggestions for
improvements. Please refer to the current edition of the "Internet
Official Protocol Standards" (STD 1) for the standardization state
and status of this protocol. Distribution of this memo is unlimited.
Copyright Notice
Copyright (C) The Internet Society (2006).
Abstract
This memo defines a method for framing Real-time Transport Protocol
(RTP) and RTP Control Protocol (RTCP) packets onto connection-
oriented transport (such as TCP). The memo also defines how session
descriptions may specify RTP streams that use the framing method.
Table of Contents
1. Introduction ....................................................2
1.1. Terminology ................................................2
2. The Framing Method ..............................................2
3. Packet Stream Properties ........................................3
4. Session Descriptions for RTP/AVP over TCP .......................3
5. Example .........................................................5
6. Congestion Control ..............................................6
7. Acknowledgements ................................................6
8. Security Considerations .........................................6
9. IANA Considerations .............................................7
10. Normative References ...........................................7
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RFC 4571 RTP & RTCP over Connection-Oriented Transport July 2006
1. Introduction
The Audio/Video Profile (AVP, [RFC3550]) for the Real-time Transport
Protocol (RTP, [RFC3551]) does not define a method for framing RTP
and RTP Control Protocol (RTCP) packets onto connection-oriented
transport protocols (such as TCP). However, earlier versions of
RTP/AVP did define a framing method, and this method is in use in
several implementations.
In this memo, we document the framing method that was defined by
earlier versions of RTP/AVP. In addition, we introduce a mechanism
for a session description [SDP] to signal the use of the framing
method. Note that session description signalling for the framing
method is new and was not defined in earlier versions of RTP/AVP.
1.1. 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 BCP 14, RFC 2119
[RFC2119].
2. The Framing Method
Figure 1 defines the framing method.
0 1 2 3
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
---------------------------------------------------------------
| LENGTH | RTP or RTCP packet ... |
---------------------------------------------------------------
Figure 1: The bit field definition of the framing method
A 16-bit unsigned integer LENGTH field, coded in network byte order
(big-endian), begins the frame. If LENGTH is non-zero, an RTP or
RTCP packet follows the LENGTH field. The value coded in the LENGTH
field MUST equal the number of octets in the RTP or RTCP packet.
Zero is a valid value for LENGTH, and it codes the null packet.
This framing method does not use frame markers (i.e., an octet of
constant value that would precede the LENGTH field). Frame markers
are useful for detecting errors in the LENGTH field. In lieu of a
frame marker, receivers SHOULD monitor the RTP and RTCP header fields
whose values are predictable (for example, the RTP version number).
See Appendix A.1 of [RFC3550] for additional guidance.
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RFC 4571 RTP & RTCP over Connection-Oriented Transport July 2006
3. Packet Stream Properties
In most respects, the framing method does not specify properties
above the level of a single packet. In particular, Section 2 does
not specify the following:
Bi-directional issues
Section 2 defines a framing method for use in one direction on a
connection. The relationship between framed packets flowing in a
defined direction and in the reverse direction is not specified.
Packet loss and reordering
The reliable nature of a connection does not imply that a framed
RTP stream has a contiguous sequence number ordering. For
example, if the connection is used to tunnel a UDP stream through
a network middlebox that only passes TCP, the sequence numbers in
the framed stream reflect any packet loss or reordering on the UDP
portion of the end-to-end flow.
Out-of-band semantics
Section 2 does not define the RTP or RTCP semantics for closing a
TCP socket, or of any other "out of band" signal for the
connection.
Memos that normatively include the framing method MAY specify these
properties. For example, Section 4 of this memo specifies these
properties for RTP/AVP sessions specified in session descriptions.
In one respect, the framing protocol does indeed specify a property
above the level of a single packet. If a direction of a connection
carries RTP packets, the streams carried in this direction MUST
support the use of multiple synchronization sources (SSRCs) in those
RTP packets. If a direction of a connection carries RTCP packets,
the streams carried in this direction MUST support the use of
multiple SSRCs in those RTCP packets.
4. Session Descriptions for RTP/AVP over TCP
Session management protocols that use the Session Description
Protocol [SDP] in conjunction with the Offer/Answer Protocol
[RFC3264] MUST use the methods described in [COMEDIA] to set up
RTP/AVP streams over TCP. In this case, the use of Offer/Answer is
REQUIRED, as the setup methods described in [COMEDIA] rely on
Offer/Answer.
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RFC 4571 RTP & RTCP over Connection-Oriented Transport July 2006
In principle, [COMEDIA] is capable of setting up RTP sessions for any
RTP profile. In practice, each profile has unique issues that must
be considered when applying [COMEDIA] to set up streams for the
profile.
In this memo, we restrict our focus to the Audio/Video Profile (AVP,
[RFC3551]). Below, we define a token value ("TCP/RTP/AVP") that
signals the use of RTP/AVP in a TCP session. We also define the
operational procedures that a TCP/RTP/AVP stream MUST follow.
We expect that other standards-track memos will appear to support the
use of the framing method with other RTP profiles. The support memo
for a new profile MUST define a token value for the profile, using
the style we used for AVP. Thus, for profile xyz, the token value
MUST be "TCP/RTP/xyz". The memo SHOULD adopt the operational
procedures we define below for AVP, unless these procedures are in
some way incompatible with the profile.
The remainder of this section describes how to setup and use an AVP
stream in a TCP session. Figure 2 shows the syntax of a media (m=)
line [SDP] of a session description:
"m=" media SP port ["/" integer] SP proto 1*(SP fmt) CRLF
Figure 2: Syntax for an SDP media (m=) line (from [SDP])
The <proto> token value "TCP/RTP/AVP" specifies an RTP/AVP [RFC3550]
[RFC3551] stream that uses the framing method over TCP.
The <fmt> tokens that follow <proto> MUST be unique unsigned integers
in the range 0 to 127. The <fmt> tokens specify an RTP payload type
associated with the stream.
In all other respects, the session description syntax for the framing
method is identical to [COMEDIA].
The TCP <port> on the media line carries RTP packets. If a media
stream uses RTCP, a second connection carries RTCP packets. The port
for the RTCP connection is chosen using the algorithms defined in
[SDP] or by the mechanism defined in [RFC3605].
The TCP connections MAY carry bi-directional traffic, following the
semantics defined in [COMEDIA]. Both directions of a connection MUST
carry the same type of packets (RTP or RTCP). The packets MUST
exclusively code the RTP or RTCP streams specified on the media
line(s) associated with the connection.
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RFC 4571 RTP & RTCP over Connection-Oriented Transport July 2006
As noted in [RFC3550], the use of RTP without RTCP is strongly
discouraged. However, if a sender does not wish to send RTCP packets
in a media session, the sender MUST add the lines "b=RS:0" AND
"b=RR:0" to the media description (from [RFC3556]).
If the session descriptions of the offer AND the answer both contain
the "b=RS:0" AND "b=RR:0" lines, an RTCP TCP flow for the media
session MUST NOT be created by either endpoint in the session. In
all other cases, endpoints MUST establish two TCP connections for an
RTP/AVP stream, one for RTP and one for RTCP.
As described in [RFC3264], the use of the "sendonly" or "sendrecv"
attribute in an offer (or answer) indicates that the offerer (or
answerer) intends to send RTP packets on the RTP TCP connection. The
use of the "recvonly" or "sendrecv" attributes in an offer (or
answer) indicates that the offerer (or answerer) wishes to receive
RTP packets on the RTP TCP connection.
5. Example
The session descriptions in Figures 3 and 4 define a TCP RTP/AVP
session.
v=0
o=first 2520644554 2838152170 IN IP4 first.example.net
s=Example
t=0 0
c=IN IP4 192.0.2.105
m=audio 9 TCP/RTP/AVP 11
a=setup:active
a=connection:new
Figure 3: TCP session description for the first participant
v=0
o=second 2520644554 2838152170 IN IP4 second.example.net
s=Example
t=0 0
c=IN IP4 192.0.2.94
m=audio 16112 TCP/RTP/AVP 10 11
a=setup:passive
a=connection:new
Figure 4: TCP session description for the second participant
The session descriptions define two parties that participate in a
connection-oriented RTP/AVP session. The first party (Figure 3) is
capable of receiving stereo L16 streams (static payload type 11).
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RFC 4571 RTP & RTCP over Connection-Oriented Transport July 2006
The second party (Figure 4) is capable of receiving mono (static
payload type 10) or stereo L16 streams.
The "setup" attribute in Figure 3 specifies that the first party is
"active" and initiates connections, and the "setup" attribute in
Figure 4 specifies that the second party is "passive" and accepts
connections [COMEDIA].
The first party connects to the network address (192.0.2.94) and port
(16112) of the second party. Once the connection is established, it
is used bi-directionally: the first party sends framed RTP packets to
the second party in one direction of the connection, and the second
party sends framed RTP packets to the first party in the other
direction of the connection.
The first party also initiates an RTCP TCP connection to port 16113
(16112 + 1, as defined in [SDP]) of the second party. Once the
connection is established, the first party sends framed RTCP packets
to the second party in one direction of the connection, and the
second party sends framed RTCP packets to the first party in the
other direction of the connection.
6. Congestion Control
The RTP congestion control requirements are defined in [RFC3550]. As
noted in [RFC3550], all transport protocols used on the Internet need
to address congestion control in some way, and RTP is not an
exception.
In addition, the congestion control requirements for the Audio/Video
Profile are defined in [RFC3551]. The basic congestion control
requirement defined in [RFC3551] is that RTP sessions should compete
fairly with TCP flows that share the network. As the framing method
uses TCP, it competes fairly with other TCP flows by definition.
7. Acknowledgements
This memo, in part, documents discussions on the AVT mailing list
about TCP and RTP. Thanks to all of the participants in these
discussions.
8. Security Considerations
Implementors should carefully read the Security Considerations
sections of the RTP [RFC3550] and RTP/AVP [RFC3551] documents, as
most of the issues discussed in these sections directly apply to RTP
streams framed over TCP.
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RFC 4571 RTP & RTCP over Connection-Oriented Transport July 2006
Session descriptions that specify connection-oriented media sessions
(such as the example session shown in Figures 3 and 4 of Section 5)
raise unique security concerns for streaming media. The Security
Considerations section of [COMEDIA] describes these issues in detail.
Below, we discuss security issues that are unique to the framing
method defined in Section 2.
Attackers may send framed packets with large LENGTH values to exploit
security holes in applications. For example, a C implementation may
declare a 1500-byte array as a stack variable, and use LENGTH as the
bound on the loop that reads the framed packet into the array. This
code would work fine for friendly applications that use Etherframe-
sized RTP packets, but may be open to exploit by an attacker. Thus,
an implementation needs to handle packets of any length, from a NULL
packet (LENGTH == 0) to the maximum-length packet holding 64K octets
(LENGTH = 0xFFFF).
9. IANA Considerations
[SDP] defines the syntax of session description media lines. We
reproduce this definition in Figure 2 of Section 4 of this memo. In
Section 4, we define a new token value for the <proto> field of media
lines: "TCP/RTP/AVP". Section 4 specifies the semantics associated
with the <proto> field token, and Section 5 shows an example of its
use in a session description.
10. Normative References
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551,
July 2003.
[COMEDIA] Yon, D. and G. Camarillo, "TCP-Based Media Transport in the
Session Description Protocol (SDP)", RFC 4145, September
2005.
[SDP] Handley, M., Jacobson, V., and C. Perkins. "SDP: Session
Description Protocol", RFC 4566, July 2006.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
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RFC 4571 RTP & RTCP over Connection-Oriented Transport July 2006
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264, June
2002.
[RFC3605] Huitema, C., "Real Time Control Protocol (RTCP) attribute
in Session Description Protocol (SDP)", RFC 3605, October
2003.
[RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth
Modifiers for RTP Control Protocol (RTCP) Bandwidth", RFC
3556, July 2003.
Author's Address
John Lazzaro
UC Berkeley
CS Division
315 Soda Hall
Berkeley CA 94720-1776
EMail: lazzaro@cs.berkeley.edu
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RFC 4571 RTP & RTCP over Connection-Oriented Transport July 2006
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