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INTERNET-DRAFT Anup Rao, Netscape Communications
Rob Lanphier, Progressive Networks
SUBMITTED: 26th November 1996 Expires: 26th May 1997
Real Time Streaming Protocol (RTSP)
<draft-ietf-mmusic-rtsp-00.txt>
STATUS OF THIS MEMO
This is an Internet-Draft. Internet-Drafts are working documents of
the Internet Engineering Task Force (IETF), its areas, and its
working groups. Note that other groups may also distribute working
documents as Internet-Drafts. Internet-Drafts are draft documents
valid for a maximum of six months and may be updated, replaced, or
obsoleted by other documents at any time. It is inappropriate to use
Internet-Drafts as reference material or to cite them other than as
"work in progress."
To learn the current status of any Internet-Draft, please check the
"lid-abstracts.txt" listing contained in the Internet-Drafts Shadow
Directories on ftp.is.co.za (Africa), nic.nordu.net (Europe),
mannari.oz.au (Pacific Rim), ds.internic.net (Us East Coast), or
ftp.isi.edu (US West Coast).
Prior to the expiration of this draft, the list of open issues may be
found at the following URL:
http://www.prognet.com/prognet/ra/openissues.html
This contains a list of issues that have been discussed on the MMUSIC
mailing list (confctrl). Many of the points brought up have a rather
holistic effect on the whole document (such as discussions of
converting the protocol to a text-based protocol, or breaking this
draft up into several drafts), while other have rather localized
effect (Appendix C changes). The points that are localized will be
be pointed out in various parts of this draft.
ABSTRACT
The Real Time Streaming Protocol, or RTSP, is an application-level
protocol for control over the delivery of data with real-time
properties. RTSP provides an extensible framework to enable
controlled, on-demand delivery of real- time data, such as audio and
video. Sources of data can include both live data feeds and stored
clips. This protocol is intended to control multiple data delivery
sessions, provide a means for choosing delivery channels such as UDP,
multicast UDP and TCP, and delivery mechanisms based upon RTP (RFC
1889). RTSP uses the Session Control Protocol (SCP) (see appendix) to
allow the use of a single TCP connection between the client and
server for controlling delivery of one or more streams of data.
Rao, Lanphier Page 1
INTERNET-DRAFT RTSP November 26, 1996
[Note: see "Use Of UDP For Delivery Of Control Messages" in the "Open
Issues" document for discussion of alternative control delivery
(http://www.prognet.com/prognet/rt/openissues.html)]
TABLE OF CONTENTS
1.0 INTRODUCTION
2.0 DOCUMENT CONVENTIONS
3.0 PROTOCOL DESCRIPTION
3.1 Connection Messages
3.2 Object Messages
3.3 Custom Messages/Events
3.4 Media specific options and Extension mechanism
4.0 EXAMPLES/APPLICATION NOTE
5.0 REFERENCES
6.0 APPENDIXES
6.1 Appendix A - Data Packets
6.2 Appendix B - Session Control Protocol (SCP)
6.3 Appendix C - RTSP Audio Format
6.4 Appendix D - RTSP Audio Annotations
6.5 Appendix E - Authors
1.0 INTRODUCTION
This document describes the Real Time Streaming Protocol, or RTSP.
The RTSP provides mechanisms to:
-Request delivery of real-time data
-Request a specific transport type and destination for
the delivery of the data
-Request information about the data in a format-
specific fashion
-Start, stop, and pause the delivery of the data
-Provide random access to various portions of the data
(where applicable)
RTSP uses TCP for delivery of control messages, and allows for a
variety of delivery options for the data including both multicast and
unicast UDP based on RTP(RFC 1889), and TCP where applicable. RTSP
uses the Session Control Protocol (see appendix) to allow the use of
a single TCP connection between the client and server for controlling
delivery of one or more streams of data. The RTSP specifically uses
one SCP session to deliver control messages and in addition, a new
SCP session might be opened for each real-time object delivered.
Rao, Lanphier Page 2
INTERNET-DRAFT RTSP November 26, 1996
While the protocol is designed for the streaming of real- time data
regardless of its format, it also provides mechanisms to inquire
about format specific information (compression type, data rate,
compression-type specific headers and frame boundaries).
Considerations for each peer arising from these features are
discussed later in the document. Three typical categories of data
whose delivery could be controlled with RTSP include:
1. Real-time stored clips
This category comprises all real-time recordings (primarily
audio/video). Examples include web sites with audio narration,
stored audio recordings, and video recordings. A typical end-user
application would be an audio/video store site providing excerpts
from its collection on-demand.
2. Real-time live feeds
This category comprises real-time data which is fed in live at the
server site rather than being pre-recorded. Examples of this usage
include a press conference, a live internet radio station, or a
televised shuttle launch.
3. Non real-time stored data
This category comprises non-real-time data of any MIME type, similar
to data served by HTTP servers. This data is provided over a new SCP
session. While HTTP servers would serve this kind of data in most
cases, it might be advanta- geous to serve non-real-time data with
RTSP. This is typically true of non-real-time data pertaining to
real- time data being served; for example, a text track playing along
with audio/video. This category is meant to provide the capability of
serving non-real-time data through RTSP, and it is strongly
recommended that it be used only when advantageous over existing
mechanisms.
[Note: Other scenarios that don't fall into the above categories may
be discussed in the "Open Issues" document
(http://www.prognet.com/prognet/rt/openissues.html)]
2.0 DOCUMENT CONVENTIONS
The following conventions are used throughout this document:
Byte Order and Alignment
All integer fields are carried in network byte order, where the most
significant byte is sent first. Integers are aligned on their natural
length: 16 bit integers on even byte boundary, 32 bit integers on 4
byte boundary and so on.
Rao, Lanphier Page 3
INTERNET-DRAFT RTSP November 26, 1996
Definitions
Stream
A stream is data of a distinct type that is sent from server to
client at a rate defined by the server, even if more bandwidth is
available. For a typical video broadcast, for example, one stream
could consist of the video signal, one stream could consist of the
audio data, and one stream could contain the closed caption
information. In most cases, each stream is served at the rate that it
should be rendered by the client.
Global Control Session
The SCP session with a well-known session ID, used for exchange of
connection messages described later in this document.
Stream Control Session
An SCP session established for each stream, used for
exchange of object messages described later in this
document.
URL
A Uniform Resource Locator or URL is a unique identifier
that describes a resource on the Internet. The syntax for
URLs is defined by RFC 1738.
3.0 PROTOCOL DESCRIPTION
Two classes of messages exist in RTSP; those sent on the
Global control session, termed connection messages and
those sent on the Stream control session, termed object
messages. Some messages might fall in both categories.
3.1 Connection messages
Connection messages are used to negotiate protocol version and
possibly client identity. This is common to all streams
requested on the connection.
Rao, Lanphier Page 4
INTERNET-DRAFT RTSP November 26, 1996
HELLO
Upon connection, the client sends a HELLO message to the server
informing it of its version and desire to establish a RTSP
session. The server replies with a HELLO message. The server
need not wait for the clients HELLO message before sending its
own. The message includes protocol version, and optional
character string denoting the user- agent(to be specified). The
communication continues in the lower of the exchanged
versions. If the two sides cannot speak this common version, the
connection is closed.
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| HELLO |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Major | Minor | Sub Protocol | Reserved |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
/ User Agent (Optional) /
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Major Version
Specifies the major RTSP version.
Minor Version
Specifies the minor RTSP version within the major version.
Sub Protocol
Specifies the sub-protocol spoken by the sender. For normal
client-server communication, this should be 0.
Reserved
For future use.
User Agent
Is an optional null-terminated character string sent by the
client to provide information like the client name, machine type,
OS etc.
Rao, Lanphier Page 5
INTERNET-DRAFT RTSP November 26, 1996
ID
This message is sent by either a server or a client. The
recipient of this message is asked to prove its identity. A
number of different schemes are possible, the simplest of which
uses a password. More complicated schemes can cause a one time
challenge and response mechanism. Multiple challenge response
steps are possible and implemenation dependent.
A separate ID message allows a server to service a number of
different classes of users. The ID message can be sent at any
time during the session or may be sent once per URL, depending on
how a service is configured.
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| tag(ID) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| key type | (reserved) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
/ authentication key /
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Key type
Specifies the authentication key type. Authentication of
different strengths might be required in different
situations. This field allows a choice of authentication. It is
up to the client and server to negotiate what level of
authentication is used. Note that different services might be
available at different authentication levels.
Authentication key
Specifies the key that authenticates the sender. The standard key
types and proper responses will be specified in an extension to
this document.
ID_RESP
This message is sent in response to an ID message. The entity
issuing the ID message closes the TCP connection if the response
is not valid.
Rao, Lanphier Page 6
INTERNET-DRAFT RTSP November 26, 1996
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| tag(ID_RESP) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| key type | response code |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
/ response data /
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Key type
Specifies the authentication key type. Authentication of
different strength may be required in different situations. This
field allows a choice of authentication. It is up to the client
and server to negotiate what level of authentication is used.
Note that different services might be available at different
authentication levels.
Response code
Specifies one of the two possible responses to an ID message:
OK
Authentication type recognized, response follows.
UNKNOWN
Unknown type, ordered list of authentication types follows
(can be a null list).
Response data
Specifies the response, depending on the response code. If OK,
then this is the key that authenticates the sender, and the
format is dictated by the authentication type. If UNKNOWN, then
this is a list of 16-bit response types that are acceptable to
the recipient.
The standard key types and proper responses will be specified in
an extension to this document.
REDIRECT
A redirect message informs the client that it must connect to
another server location. When received on the Global control
session, it should be made effective as specified by offset. The
message can also be sent on a Stream control session, in which
case it it relative to the stream.
Rao, Lanphier Page 7
INTERNET-DRAFT RTSP November 26, 1996
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
+-------------------------------+-------------------------------+
| tag (REDIRECT) |
+-------------------------------+-------------------------------+
| Offset |
+-------------------------------+-------------------------------+
| |
/ URL /
| |
+-------------------------------+-------------------------------+
Offset
On the Stream control session, specifies the position (in
milliseconds) in the media stream at which time the
redirection is effective. If received on the global control
session, it means that the server is not available after
this time.
URL
Is a null-terminated string that identifies the new location for
the object. This URL is similar to the one specified in the
original FETCH request, described later in this document.
OPTIONS
This message is used to negotiate additional functionality. This
can be sent by the client or the server. This is provided as an
extension mechanism to extend the protocol without changing the
version number. Set 0 refers to the global set of options, a
universally-agreed upon set. Implementors can use this to
negotiate replacements for messages in the current specification;
however, all implementations must be able to support the core
messages defined here, and be prepared not to use an option
should the other implementation not be prepared to support the
given option. The maximum size of an option set is 64.
This is defined in the spirit of the TELNET protocol (RFC 854),
which provides a further explanation of how different
implementations may interoperate using this mechanism.
Rao, Lanphier Page 8
INTERNET-DRAFT RTSP November 26, 1996
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| tag (OPTIONS) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| option set |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| highest option number |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|s.d.s.w|s.d.s.w|s.d.s.w|s.d.s.w|s.d.s.w|s.d.s.w|s.d.s.w|s.d.s.w|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
/ /
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|s.d.s.w|s.d.s.w|s.d.s.w|s.d.s.w|s.d.s.w|s.d.s.w|s.d.s.w|s.d.s.w|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Option set
Implementors can specify an option set to use for adding custom
extensions. In addition, as the protocol matures, certain
options may be accepted as universal to all clients,which will be
placed in the reserved range of option sets. The following ranges
of options are reserved:
Option set Purpose
----------- ------------------------------------
0 This is a universally accepted set
of options
0x1-0xFFFF Reserved for later allocation
0x10000-0xFFFFFFFF Custom extensions (i.e. vendor or
organization specific)
Highest option number
Specifies the highest option number that is set or unset. It is
used to calculate the length of the bit vectors that follow. This
list is zero indexed.
Rao, Lanphier Page 9
INTERNET-DRAFT RTSP November 26, 1996
s.d.s.w
s.d.s.w
^ ^ ^ ^
| | | |
select do/don't ----------+ | | |
do/don't -------------------+ | |
select will/won't ------------+ |
will/won't ---------------------+
select do/don't
If set, activates the do/don't flag
do/don't
If set to 1, means DO and if set to 0, means DON'T.
DO indicates the request that the other party perform, or
confirmation that you are expecting the other party to perform,
the indicated option.
DON'T indicates the demand that the other party stops performing,
or confirmation that you are no longer expecting the other party
to perform, the indicated option.
select will/won't
If set, activates the will/won't flag
will/won't
1 means WILL, 0 means WON'T.
WILL indicates the desire to begin performing, or confirmation
that the indicated options are being used.
WON'T indicates the refusal to perform, or continue performing,
the indicated option.
At a minimum, the response to a OPTION is to reply with the
desire to perform NONE of the requested options.
The Working Group or IETF needs to define and regulate the
registration process for OPTIONS sets. Currently, no sets are
defined.
GOODBYE
Sent only by the client to politely close the RTSP session.
Rao, Lanphier Page 10
INTERNET-DRAFT RTSP November 26, 1996
The client sends this message to the server to indicate that it
is done with its requests and wants to terminate the session.
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
+-------------------------------+-------------------------------+
| tag (GOODBYE) |
+-------------------------------+-------------------------------+
3.2 OBJECT MESSAGES
These messages are sent on the stream control session (one of
which exists for each stream). They a used to request objects,
set transport mechanism, and control data delivery.
FETCH
This is the first message sent on a new stream control session.
It requests an object of a given name. If the object exists, and
the client is permitted to access it, the server accepts the
request and sends a STREAM_HEADER message to the client. If the
object is non-existent or access is not allowed, the appro-
priate error message is sent to the client.
The client uses this message to request an object (for example,
an audio clip) from the server. The object is specified as an URL.
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
+-------------------------------+-------------------------------+
| tag (FETCH) |
+-------------------------------+-------------------------------+
| Bandwidth (Bits per second) |
+-------------------------------+-------------------------------+
| Max Number of Hops | Request Flags |F|M|N|
+-------------------------------+-------------------------------+
| |
/ URL (null-terminated) /
| |
+-------------------------------+-------------------------------+
Bandwidth
Specifies the client's estimate of the bandwidth (in bits per
sec) it has available for a particular stream. The exact bit rate
of the stream is provided by the STREAM_HEADER message from the
server. If the server refuses the request based on insufficient
bandwidth, then the server sends the appropriate error message.
Rao, Lanphier Page 11
INTERNET-DRAFT RTSP November 26, 1996
Max Number of Hops
Specifies the maximum number of hops. A hop is a connection
between Client, Server or Proxy. For example, the are three hops
for passing through two proxies and connecting to a server. If
set to 0xFFFF, then the number of hops is unlimited.
Request Flags
F - Fresh bit, used to override Proxy Cache
M - Multicast capable
N - No Rate (rate unlimited file transfer, used only for TCP.
This could be used for transfering non-real time data, such as
metafiles)
URL - Universal Resource Locator
Specifies a null-terminated string which identifies the
location for the object (i.e. RTSP://host-name:port/filename)
STREAM_HEADER
This message is sent in response to a successful FETCH message.
It provides the basic information about the stream such as rate,
generic type, and last modification time.
Rao, Lanphier Page 12
INTERNET-DRAFT RTSP November 26, 1996
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
+-------------------------------+-------------------------------+
| tag (STREAM_HEADER) |
+-------------------------------+-------------------------------+
| Generic Type (family) |
+-------------------------------+-------------------------------+
| Bit Rate (Bits per second) |
+-------------------------------+-------------------------------+
| Last Modification Time |
+-------------------------------+-------------------------------+
| Reserved |
+-------------------------------+-------------------------------+
| Length (milliseconds) |
+-------------------------------+-------------------------------+
| Max Packet size (bytes) |
+---------------+---------------+---------------+---------------+
| Flags x|L|C|H|A|U|M|
+---------------+---------------+---------------+---------------+
Generic type (family)
Specifies the type of family, such as RTSP_Audio, etc. Currently,
RTSP_Audio is defined in the appendix.
Bandwidth
Specifies the rate or bandwidth of the stream in bits per second.
It is set to 0 if non-realtime.
Last Modification Time
Specifies the last time the object was modified. The specified
time is seconds since Jan 01 1970(UTC).
Reserved
Reserved for future use.
Length
Specifies the stream length in miliseconds. It is 0 if the object
is live or if it is non-realtime.
Max Packet Size
Specifies the maximum packet size in bytes that the server
supports for the stream.
Rao, Lanphier Page 13
INTERNET-DRAFT RTSP November 26, 1996
Flags
x: Unicast available
L: Live : if set, the stream is live and there is no
concept of position.
C: Copy : if set, means that the client can save it.
H: Cached : if set, the object is cached on a proxy.
A: if set, stream can be cached by the client
U: UDP Okay : if set, the stream can be sent via a
unreliable channel.
M: Stream is available via Multicast, and a
SET_TRANSPORT message follows providing this
information
[Note: There has been discussion of providing this
information in a session description that is loaded via
HTTP or some other mechanism prior to the RTSP connection
being established. See "Metafiles/Session Descriptions" in
http://www.prognet.com/prognet/rt/openissues.html]
SET_TRANSPORT
This message is sent to set the delivery mechanism for a stream.
The client uses this message to request the transport used for
the data transfer.
The server can use this message to inform the client about multicast
availability (this is the only instance the server sends this
message).
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| tag (SET_TRANSPORT) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|C| | ChannelID |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| synchronization source (SSRC) identifier |
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
/ Transport specific fields /
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
C flag : This flag specifies the type of header/framing.
C = 0 : Standard RTP header
C = 1 : Compressed/Extensible RTP header
Rao, Lanphier Page 14
INTERNET-DRAFT RTSP November 26, 1996
SSRC: 32 bits
Identifies the synchronization source to be associated with the
data. This can be used for de-multiplexing by the client of data
received on the same port.
Channel ID
Transport channel Fixed RTP header Compressed RTP header Channel ID
Unicast UDP Yes Yes 0
Multicast UDP Yes Yes 1
SCP Yes Yes 2
SCP - compressed Yes Yes 3
Note : The client may not send a channel ID of 1. This is
sent by the server only.
Transport Spcific Fields
Channel ID 0:
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
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
| port |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Port
Specifies the port number on which the stream should be sent.
Channel ID 2,3 :
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
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
| session ID | (reserved) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Session ID
Specifies the SCP session ID that should be associated with this
stream.
Rao, Lanphier Page 15
INTERNET-DRAFT RTSP November 26, 1996
Channel ID 1
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
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
| port |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| address |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Port
Specifies the port number on which the stream is sent.
Address
Specifies the multicast group address where the data is
available.
[Note: There is some debate over the best way to address
RTP header compression in the "Compressed RTP section" of
the "RTSP Open Issues" document.
(see http://www.prognet.com/prognet/rt/openissues.html)]
[Note: There has also been debate about what level of RTP
support to provide. It has been suggested that a server
should be able to stream directly into an RTP based
conference which may contain clients that are not
necessarily RTSP-aware, so long as the requesting
conference client provides control connection proxying.
See the "RTSP Open Issues" document section titled "Joining
Into Existing Conferences"
(http://www.prognet.com/prognet/rt/openissues.html)]
SET_SPEED
This advisory message sets the speed at which the server delivers
data to the client, contingent on the server's ability and desire
to serve the file at the given speed. Implementation by the
server is optional. The default is the bit rate of the stream.
Rao, Lanphier Page 16
INTERNET-DRAFT RTSP November 26, 1996
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| tag(SET_SPEED ) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| (reserved) |F|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| speed |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Speed
Send speed as a ratio. The speed is relative to the normal
speed for the stream, as expressed as a ratio of speed
divided by 65536 (i.e. 2^16). For example, speed is 65536
for real-time delivery, 32768 (2^15) for half-speed
delivery, and 131072 (2^17) for double-speed delivery. A
speed of zero is invalid.
F
If F is set, sends the data as fast as possible. This feature
only works on a TCP data connection.
PLAY_RANGE
This message tells the server to start sending data via the
previously set transport mechanism. The two fields, From and To,
specify the range in milliseconds. If the the "To" field is zero,
it signifies the end of the stream. If the "To" field is non-
zero, it should be greater than the value specified in
the "From" field.
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
+-------------------------------+-------------------------------+
| tag (PLAY_RANGE) |
+-------------------------------+-------------------------------+
| From (in ms) |
+-------------------------------+-------------------------------+
| To (in ms) |
+-------------------------------+-------------------------------+
Rao, Lanphier Page 17
INTERNET-DRAFT RTSP November 26, 1996
STREAM_SYNC
This message is sent by the server to the client in response to
the PLAY_RANGE message. It informs the client of the first
expected sequence number, and in case of some delivery types, the
first expected timestamp.
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
+-------------------------------+-------------------------------+
| tag (STREAM_SYNC) |
+-------------------------------+-------------------------------+
| Sequence Number | Reserved |
+-------------------------------+-------------------------------+
| Start Time |
+---------------------------------------------------------------+
Sequence Number
Specifies where the new data starts. This number should be far
enough away from the last sequence number sent so that old
packets can be detected and rejected.
Start Time
Specifies the offset (in milliseconds) relative to the origin of
the first packet that will be sent.
SET_BLOCK_SIZE
This is an advisory message sent from the client to the server
setting the transport packet size. The server truncates it to the
closest multiple of the minimum media-specific block size or
overrides it with the media specific size if necessary.
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
+-------------------------------+-------------------------------+
| tag (SET_BLOCKSIZE) |
+-------------------------------+-------------------------------+
| Block Size |
+-------------------------------+-------------------------------+
Block Size
Specifies the size of the block in bytes.
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STOP
This messages tells the server to stop sending. This is used for
`pause' operation, and the server is to keep track of the stream
position in order to `resume' play when a resume command is given.
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
+-------------------------------+-------------------------------+
| tag (STOP) |
+-------------------------------+-------------------------------+
RESUME
This message tells the server to resume sending from the position
it was stopped at with the STOP message. If no STOP message was
sent prior to RESUME, the messages is ignored.
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
+-------------------------------+-------------------------------+
| tag (RESUME) |
+-------------------------------+-------------------------------+
SEND_REPORT
This message is sent by the server to the client requesting
statistics on data reception. At a minimum, the client is
required to reply to the message of type 1 (PING) indicating that
it is receiving data.
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
+-------------------------------+-------------------------------+
| tag (SEND_REPORT) |
+-------------------------------+-------------------------------+
| Report Type |
+-------------------------------+-------------------------------+
Report Type:
1 - PING,
2 - Text Message
3 - Reception Report
REPORT
Sent by the client to the server in response to the SEND_REPORT
message. or otherwise.
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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
+-------------------------------+-------------------------------+
| tag (REPORT) |
+-------------------------------+-------------------------------+
| Report Type |
+=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=+
| Report Specific Fields |
+-------------------------------+-------------------------------+
Report Type:
1 - PING
2 - Text Message,
3 - Reception Report
PING Report (1) Report Specific Fields (NONE, just respond with
Report type 1)
Text Message (2) Report Specific Fields
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
+=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=+
| Text String (Null Terminated) |
+-------------------------------+-------------------------------+
Text String (Null terminated).
Reception Report (3) - Report specific fields
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
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
| Packets Received | Packets Lost |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Lagging | Buffer occupancy |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Packets Received
Specifies the number of packets received since last report
Packets Lost
Specifies the number of packets lost since the last report.
Lagging
Specifies that the buffer level is below (1), or above the
desired level (0).
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Buffer occupancy
Specifies the number of milliseconds below or above the desired
level.
ERROR
This message is sent by the server to the client in case of an
abnormal condition. It includes a numeric error code which is
defined later in this document and an optional null-terminated
error string.
Error is sent by the server to report an error to the client.
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
+-------------------------------+-------------------------------+
| tag (ERROR) |
+-------------------------------+-------------------------------+
|F| error code |
+-------------------------------+-------------------------------+
| |
/ Error String (null-terminated) /
| |
+-------------------------------+-------------------------------+
F:(one bit) Fatal Error Flag
Is used to tell the client it must terminate the session.
error codes:
200 - Success
400 - FETCH failure; no such object exists
401 - FETCH failure; fetch refused
402 - SEND_RANGE out of bounds
403 - Incompatible protocol version
404 - SET_TRANSPORT failure; bad port specified
405 - Feed gone - live feed closed
406 - Can't do that on non-live feed
407 - Request not supported
408 - Request refused due to bandwidth mismatch
(e.g. Insufficient Client Bandwidth)
500 - Unspecified server-side problem
501 - Server running low on resources and cannot
satisfy request
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UDP_RESEND
This messages is sent by the client to request retransmission
packets of specific sequence numbers. The service of the request
at the server end is optional.
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
+-------------------------------+-------------------------------+
| tag (UDP_RESEND) |
+-------------------------------+-------------------------------+
| Sequence Number | Number of Packets |
+-------------------------------+-------------------------------+
Sequence Number:
Specifies the first packet that needs to be resent.
Number of Packets:
Specifies the number of packets that need to be resent, beginning
with the above sequence number.
3.3 CUSTOM MESSAGES/EVENTS
These messages are intended to provide a general notification
system from the server to the client or from the client to the
server. The message could be sent on the control or object
session. An example of using the custom message is when the
server announces the availability of a new stream, or drives the
client's web browser to a particular location.
The message includes payload type and payload. Currently, defined
payload types are NewURL(1) and GotoURL(2). Implementation of
this custom messaging system is optional for both client and
server, and vendor specific payload types could be added as
necessary.
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EVENT_NOTIFY
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| tag(EVENT_NOTIFY) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Payload Type |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
/ Payload /
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Payload Type
Specifies one of the currently defined payload types. For
example, NewURL(1) and GotoURL(2).
Payload
For the previous two payload types, specifies the URL.
3.4 MEDIA SPECIFIC OPTIONS AND EXTENSION MECHANISM
RTSP provides an extensible way for the client to query and set
media-specific parameters and to set media-specific parameters
for the stream. This is done via the GET_PARAM and PARAM_REPLY
messages. The GET_PARAM message includes a request ID, family
identifier and a parameter identifier. The PARAM_REPLY mes-
sage includes the corresponding request ID, and a variable length
reply data payload.
GET_PARAM
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| tag (GET_PARAM) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| REQUEST_ID |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| FAMILY_ID |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| PARAMETER_ID |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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REQUEST_ID
Is a 32 bit number used to reference this request, must be non-
zero.
FAMILY_ID
Currently, RTSP_Audio (Family 1) is defined in Appendix C.
PARAMETER_ID
Specified the parameter ID within the family. These are defined
for the in Appendix C.
PARAM_REPLY
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| tag(PARAM_REPLY) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| REQUEST_ID |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| FAMILY_ID |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| PARAMETER_ID |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| DATA_TYPE |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
/ DATA /
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
For a list of currently defined family and parameter identifiers,
see Appendix C. An example of use of these messages is available
in the next section. Data types are String(0), Integer(1), and
Opaque data(2).
This message can be sent by the server to the client, even if the
client has not sent a request. The client must be prepared to
handle this. In this case, the REQUEST_ID is 0x0.
REQUEST_ID
Is a 32 bit number used to reference this request, must be non-
zero.
FAMILY_ID
Currently, RTSP_Audio (Family 1) is defined in Appendix C.
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PARAMETER_ID
Specified the parameter ID within the family. These are defined
for the in Appendix C.
PARAM_REFUSE
If the server does not support the requested parameter or family,
it sends a ParamRefuse, whose format is as below:
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| tag(PARAM_REFUSE) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| REQUEST_ID |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| FAMILY_ID |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| PARAMETER_ID |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ERROR_CODE |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
REQUEST_ID
Is a 32 bit number used to reference this request, must be non-
zero.
FAMILY_ID
Currently, RTSP_Audio (FAMILY_ID1) is defined in Appendix C.
PARAMETER_ID
Specified the parameter ID within the family. Currently, the
RTSP_Audio is defined in Appendix C.
ERROR_CODE: As defined earlier.
SET_PARAM
Using the same principle as GET_PARAM, a media-specific option
for the stream can be set using the SET_PARAM message. Examples
of an option include encryption, custom encoding or quality
levels. The server replies with a SET_PARAM_REPLY or
PARAM_REFUSE.
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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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| tag (SET_PARAM) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| REQUEST_ID |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| FAMILY_ID |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| PARAMETER_ID |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
/ DATA /
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
REQUEST_ID
Is a 32 bit number used to reference this request, must be non-
zero.
FAMILY_ID
Currently, RTSP_Audio (Family 1) is defined in Appendix C.
PARAMETER_ID
Specified the parameter ID within the family. These are defined
for the in Appendix C.
DATA
Specifies the payload specific to the family and parameters.
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4.0 EXAMPLES/APPLICATION NOTE
Example of use of GetParam to request and play an audio file:
CLIENT SERVER
CONTROL_SCP: HELLO--------------------------->
<---------------------------CONTROL_SCP:HELLO
<--------------------------- CONTROL_SCP:ID
(OPTIONAL) CONTROL_SCP:ID_RESP--------------------->
STREAM_SCP_0: FETCH----------------------------->
STREAM_SCP_0: GETPARAM-------------------------->
STREAM_SCP_0: GETPARAM-------------------------->
<------------------------- STREAM_SCP_0:STREAM_HEADER
<------------------------- STREAM_SCP_0:PARAM_REPLY
<------------------------- STREAM_SCP_0:PARAM_REPLY
STREAM_SCP_0: SETPARAM(eg.encryption) ---------------->
<-------------------------- STREAM_SCP_0:PARAM_REPLY
STREAM_SCP_0: SET_TRANSPORT(RTP)---------------->
STREAM_SCP_0: PLAY_RANGE(a,b)------------------->
<--------------------------- STREAM_SCP_0:STREAM_SYNC
<--------------------------- Data on UDP channel
Example of use of custom message:
CLIENT SERVER
CONTROL_SCP: HELLO---------------------------->
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<----------------------------CONTROL_SCP:HELLO
STREAM_SCP_0: FETCH ------------------->
<-------------------------- STREAM_SCP_0:STREAM_HEADER
STREAM_OBJECT_SCP_0: SET_TRANSPORT(RTP)-------->
STREAM_OBJECT_SCP_0: PLAY_RANGE(a,b)----------------->
<--------------------------- STREAM_SCP_0:STREAMSYNC
<--------------------------- Data on UDP channel
<--------------------------- CONTROL_SCP: EVENT_NOTIFY(NewURL)
STREAM_OBJECT_SCP_1: FETCH--------------------->
<------------------------- STREAM_SCP_1:STREAM_HEADER
STREAM_SCP_1: SET_TRANSPORT(RTP)---------------->
STREAM_SCP_1: SEEK(a,b)------------------------->
<--------------------------------- STREAM_SCP_1:STREAM_SYNC
<------------------------------- Data on UDP channel
5.0 REFERENCES:
[1] H. Shulzrinne, S. Casner, R. Frederick, and S. McCanne, "RTP:
A Transport Protocol for real-time applications." RFC 1889
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[2] Simon Spero, Session Control Protocol(SCP).
[3] J. Postel, J. Reynolds, "Telnet Protocol Spefication.", RFC 854
6.0 APPENDIXES
6.1 APPENDIX A - Data packets
[Note: see "Compartmentalize the Protocol" in the "RTSP
Open Issues" document for discussion of the possible split
of this appendix into a separate document
(http://www.prognet.com/prognet/rt/openissues.html)]
The following are the types of data packet formats.
RTP: The format of the UDP packet is a standard RTP packet, as
per RFC 1889. This is a description of how the RTP packet is used
by RTSP. Complete generalized descriptions of these fields can be
found in RFC 1889.
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|V=2|P|X| CC |M| PT | sequence number |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| timestamp |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| synchronization source (SSRC) identifier |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| contributing source (CSRC) identifiers |
| .... |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The first twelve octets are present in every RTP packet, while
the list of CSRC identifiers usually will not be present (though
may be used when broadcasting an RTP conferencing session, for
instance). The fields have the following meaning:
Version (V): 2 bits
This field identifies the version of RTP. The version used by
this specification is two (2).
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Padding (P): 1 bit
If the padding bit is set, the packet contains one or more
additional padding octets at the end which are not part of the
payload. The last octet of the padding contains a count of how
many padding octets should be ignored.
Extension (X): 1 bit
Not used. Set to zero for RTP compatibility. CSRC count (CC): 4
bits. The CSRC count contains the number of CSRC identifiers that
follow the fixed header.
Marker (M): 1 bit
In RTSP, this designates the block boundary. This is a point at
which a client can start decoding the stream.
Payload type (PT): 7 bits
Each stream uses a payload type allocated dynamically between 96
and 127, to remain consistent with the dynamic mappings in the
audio-video profile (RFC 1890).
Sequence number: 16 bits
The sequence number increments by one for each RTP data packet
sent, and may be used by the receiver to detect packet loss and
to restore packet sequence. The initial value of the sequence
number is random (unpredictable) to make known plaintext attacks
on encryption more difficult, even if the source itself does
not encrypt, because the packets may flow through a translator
that does.
Timestamp: 32 bits
The timestamp is a tick mark which reflects the sampling instant
of the first octet in the RTP data packet. The clock frequency is
dependent on the format of data carried as payload.
SSRC: 32 bits
The SSRC field identifies the synchronization source associated
with the data. This identifier is chosen with the intent that no
two synchronization sources within the same RTSP session will
have the same SSRC.
CSRC list: 0 to 15 items, 32 bits each
The CSRC list identifies the contributing sources for the payload
contained in this packet. The number of identifiers is given by
the CC field. If there are more than 15 contributing sources,
only 15 may be identified.
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Compressed RTP
[Note: There has been some questions as to the
appropriateness of this name, since it is really an
application-level compression that relies on RTSP to carry
some of the "compressed" information out of band. We will
probably change this name to reflect its distance from RTP.]
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|v|M|t|s| SSRC | sequence number |timestamp (opt)...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
... |
+-+-+-+-+-+-+-+-+
version (v): 1 bit
If this is 1, the rest of this packet is standard RTP, and we can
treat it as such. If it is 0, the rest of the packet is defined
as follows.
marker (M): 1 bit
In RTSP, this designates the block boundary. This is a point at
which a client can start decoding the stream.
timestamp lower bits compression bit (t): 1 bit
This number specifies the existence of the least-significant two
bytes of the timestamps. If this bit is not set, the value of
these bytes are calculated based some media-specific method
(probably using the sequence number). If this bit is set, the
value of these bytes are given in this packet.
The upper bits are determined based on the probable value given
the sequence number. This compression scheme does not work for
packets that are spaced out at intervals greater than roughly 64
seconds.
sequence number inclusion bit(s)
This bit is for compatibility with reliable, sequenced delivery
mechanisms, such as SCP. This should always be set to 1.
SSRC: 4 bits or 36 bits
The SSRC field identifies the synchronization source associated
with the data. This identifier is chosen with the intent that no
two synchronization sources within the same RTSP session will
have the same SSRC.
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This is 4 bits unless set to 15 (1111), in which case it expands
to 36 bits, and the lower 32 bits are the actual SSRC.
sequence number: 16 bits
The sequence number increments by one for each RTP data packet
sent, and may be used by the receiver to detect packet loss and
to restore packet sequence. The initial value of the sequence
number is random (unpredictable) to make known-plaintext attacks
on encryption more difficult, even if the source itself does
not encrypt, because the packets may flow through a translator
that does.
6.2 APPENDIX B - Session Control Protocol (SCP)
[Note: see "Compartmentalize the Protocol" in the "RTSP
Open Issues" document for discussion of the possible split
of this appendix into a separate document
(http://www.prognet.com/prognet/rt/openissues.html)]
[Note: other methods of multiplexing are currently being
discussed. Simon Spero has a new draft of SCP at
http://sunsite.unc.edu/ses/scp.html. SMUX, from Jim
Gettys and the W3C at http://www.w3.org is another candidate.]
This document was originally written by Simon Spero for the HTTP-
NG project. It has been slightly editied and clarified.
Several heavily used Internet applications such as FTP, GOPHER,
and HTTP use a protocol model in which every transaction
requires a separate TCP connection. Since clients normally issue
multiple requests to the same server, this model is quite
inefficient, as it incurs all the connection start up costs for
every single request.
SCP is a simple protocol which lets a server and client have
multiple conversations over a single TCP connection.
Definitions
A session is a virtual stream of data carried within the single
TCP connection. It is independent of any other session within the
same connection.
A session identifier is a unique number which is used to specify
which session a particular block of data is associated with.
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Session flag names are borrowed from TCP. The name SYN means
"open" in SCP. The name FIN means "finish" or "close" in SCP. The
name RST means "reset" in SCP. PUSH indicates the end of a
segment.
Services
SCP's main service is dialogue control. This service allows
either end of the connection to establish multiple virtual
sessions over a single transport connection. SCP also allows a
sender to indicate message boundaries, and allows a receiver to
reject an incoming session.
Design goals
SCP allows data to be sent with the session establishment; the
recepient does not confirm successful connection establishment,
but may reject unsuccessful attempts. This simplifies the design
of the protocol, and removes the latency required for a con-
firmed operation.
Low overhead
SCP has a fixed overhead of 8 bytes per segment. This overhead is
only incurred once per segment, instead of once per packet.
Simple design
The session protocol should be simple enough to implement for a
single application.
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Protocol Description
Header Format:
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|0|0|0|0| flags | session ID |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| session length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
/ data /
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
flags:
0x1 : SYN
0x2 : FIN
0x4 : RST
0x8 : PUSH
Protocol Operation
Session ID allocation
Each session is allocated a session identifier. Session
Identifiers below 1024 are reserved. Session IDs allocated by
clients are even; those allocated byservers, odd.
Session establishment
A new session is established by setting the SYN bit in the first
message sent on that channel.
Graceful release
A session is ended by sending a message with the FIN bit set.
Each end of a connection may be closed independently.
Disgraceful release
A session may be terminated by sending a message with the RST bit
set. All pending data for that session should be discarded
Message boundaries
A message boundary is marked by sending a message with the PUSH
bit set. The boundary is set at the final octet in this message,
including that octet.
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Compressed Header SCP
+-+-+-+-+-+-+-+-+-----//------+-------//-------+---//---+
|1|P| IDL | LNL | session ID | segment length | data |
+-+-+-+-+-+-+-+-+-----//------+-------//-------+---//---+
P: 1 bit
Push bit. This denotes a message boundry, which is set at the
final octet in this message.
IDL: 3 bit
Length of the session ID field (as described below).
LNL: 3 bits
Length of the session length field (as described below).
session ID: Variable number of bits, depending on IDL field
Identifier for this session. The session ID value is relative to
1024, which is the minimum non-reserved value.
segment length: Variable number of bits, depending on LNL field.
Length of the payload in bytes.
IDL and LNL values: 3 bits
value Length of Session ID or Segment Length field
0x0 use previous value
0x1 4 bits
0x2 8 bits
0x3 12 bits
0x4 16 bits
0x5 24 bits
0x6 32 bits
0x7 64 bits
IDL and LNL values should be chosen so that byte alignment of the
packet is guaranteed.
6.3 APPENDIX C - RTSP -Audio (Family ID =1 )
Format descriptor (Parameter=1)
[Note: see "Compartmentalize the Protocol" in the "RTSP
Open Issues" document for discussion of the possible split
of this appendix into a separate document
(http://www.prognet.com/prognet/rt/openissues.html)]
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[Note: see "Change Appendix C (families of stream types)"
in the "RTSP Open Issues" document for discussion of a
series of proposed changes to this Appendix, such as
changing CODEC IDs to UUIDs, modifying the audio family,
and adding a video family
(http://www.prognet.com/prognet/rt/openissues.html)]
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| CODEC ID |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| NumChannels | Bits/Sample |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Samples Per Second |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Average Frame Size | Maximum Frame Size |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Samples Per Frame |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|P| Frames Per Packet | Extra Bytes |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Number Of Frames |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| CODEC-specific Data |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: : :
: : :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The above structure describes a generic compressed audio stream
for the RTSP Audio Family.
CODEC ID: The CODEC ID uniquely identifies the codec which was
used to generate this audio data. CODEC ID's in the range 0x00
through 0xffff are reserved for ACM (Audio Compression Manager
native to Microsoft Windows Operating systems) compatible/
interoperable codecs.
NumChannels: Use 1 for mono, 2 for stereo
Bits/Sample: Number of bits per sample as specified by the codec.
Codecs which do not need or do not have a well defined value for
this parameter may set this to zero.
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Samples Per Second: Playback rate for each channel in
samples per second (hertz).
Average Frame Size: Frame is the smallest unit of encoded audio
data that can be streamed from the server. Average Frame Size
represents the typical size (in bytes) of a frame. This family
does not support non byte-aligned frames.
Maximum Frame Size: Size in bytes of the largest frame. For
constant bit rate codecs, this value is the same as the Average
Frame Size.
Samples Per Frame: Number of samples of audio data encoded in
each frame. This family does not support variable duration frames.
P: If P is not set, then the audio data is planar and the client
has the option to request data packets of size which are
multiples of the average frame size. If P is set, the stream is
packetized (usually for variable bit rate data and live feeds),
then the size of the data packet is defined by frame size and
frames per packet (described below).
Frames Per Packet: For packetized formats, this represents the
number of frames in every data packet. For planar formats, this
is a hint to the client regarding the server preference.
Extra Bytes: This specifies the number of bytes of codec specific
data that follow the fixed header.
Number of Frames: Total number of frames in the audio stream. For
non-seekable streams (e.g. live feeds), this should be set to
zero.
6.4 APPENDIX D - RTSP - Audio (Family ID =1 ) : Annotations
(Parameter=2)
[Note: see "Compartmentalize the Protocol" in the "RTSP
Open Issues" document for discussion of the possible split
of this appendix into a separate document
(http://www.prognet.com/prognet/rt/openissues.html)]
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This payload data enables the server to communicate additional
information about the stream to the client. Examples of such
information include the authorof the clip, the clip name,
copyright information, creation date etc. The payload data is
organized as a list of information chunks. Each such chunk has an
identifier, length of the chunk and the actual data itself.
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Number of chunks |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Chunk Identifier (Chunk 0) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Number of Bytes (Chunk 0) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Chunk Data (Chunk 0) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: : :
: : :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Chunk Identifier (Chunk n-1) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Number of Bytes (Chunk n-1) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Chunk Data (Chunk n-1) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: : :
: : :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Chunk Identifier
This is a four character code identifying the chunk. For example,
`I''C''O''P' is used for copyright information which is
stored as a null terminated string. `I''N''A''M' is used for the
name of the audio clip. `I''U''R''L' serves to pass the URL for
the CODEC installer. The client should use this URL if it
encounters a clip compressed with an unknown CODEC.
Any chunk with an unrecognized four character code should be
ignored by the client.
Number of Bytes
Represents the length of the chunk data. The chunk data should be
16 bit aligned.
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6.5 APPENDIX E - Authors
Contacts:
At Netscape: Anup Rao <anup@netscape.com>
At Progressive Networks: Rob Lanphier <robla@prognet.com>
[Note: this list will need to be updated for the final
release of the document]
The following authors contributed to this document:
Martin Dunsmuir <martind@prognet.com>
General Manager Server Products,
Progressive Networks
1111 Third Avenue, Suite 2900
Tel. 206 674 2700 (main #)
John K. Ho <johnkho@netscape.com>
Project Manager
Netscape Communications Corp.
685 E. Middlefield Rd.
Mountain View, CA, 94043
Tel. 415 254 1900 (main #)
Rob Lanphier <robla@prognet.com>
Program Manager
Progressive Networks
Eduardo F. Llach <eduardo@netscape.com>
Manager, Media Server Team,
Netscape Communications Corp.
Rob McCool <robm@netscape.com>
Technical Team Leader, Media Server
Netscape Communications Corp.
Igor Plotnikov <igor@netscape.com>
Technical Team Leader, Media Server
Netscape Communications Corp.
Anup Rao
Member, Technical Staff
Netscape Communications Corp.
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Pinaki Shah
Member, Technical Staff
Netscape Communications Corp.
Alexander Sokolsky
Member, Technical Staff
Netscape Communications Corp.
Dale Stammen
Lead Saxophone
Progressive Networks
John Francis Stracke
Senior LiveMedia Architect
Netscape Communications Corp.
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