\section{Introduction}
\label{sec:intro}

\subsection{Purpose}

The Real-Time Streaming Protocol (RTSP) establishes and controls either
a single or several time-synchronized streams of continuous media such
as audio and video.  It does not typically deliver the continuous
streams itself, although interleaving of the continuous media stream
with the control stream is possible (see Section \ref{sec:binary}).  In
other words, RTSP acts as a ``network remote control'' for multimedia
servers.

The set of streams to be controlled is defined by a presentation
description.  This memorandum does not define a format for a
presentation description.

There is no notion of an RTSP connection; instead, a server maintains a
session labeled by an identifier.  An RTSP session is in no way tied to
a transport-level connection such as a TCP connection.  During an RTSP
session, an RTSP client may open and close many reliable transport
connections to the server to issue RTSP requests.  Alternatively, it may
use a connectionless transport protocol such as UDP.

The streams controlled by RTSP may use RTP \cite{rfc1890}, but the
operation of RTSP does not depend on the transport mechanism used to
carry continuous media.

The protocol is intentionally similar in syntax and operation to
HTTP/1.1 \cite{rfc2068} so that extension mechanisms to HTTP can in most 
cases also be
added to RTSP.  However, RTSP differs in a number of important aspects
from HTTP:

\begin{itemize}
\item RTSP introduces a number of new methods and has a different
protocol identifier.

\item An RTSP server needs to maintain state by default in almost all
cases, as opposed to the stateless nature of HTTP.  

\item Both an RTSP server and client can issue requests.

\item Data is carried out-of-band by a different protocol.  (There is
an exception to this.)

\item RTSP is defined to use ISO 10646 (UTF-8) rather than ISO 8859-1,
consistent with current HTML internationalization efforts
\cite{rfc2070}.

\item The Request-URI always contains the absolute URI.  Because of
backward compatibility with a historical blunder, HTTP/1.1
\cite{rfc2068} carries only the absolute path in the request and puts
the host name in a separate header field.

\motivation{This makes ``virtual hosting'' easier, where a single host
with one IP address hosts several document trees.}

\end{itemize}

The protocol supports the following operations:

\begin{description}
\item[Retrieval of media from media server:] The client can request a
presentation description via HTTP or some other method.  If the
presentation is being multicast, the presentation description contains
the multicast addresses and ports to be used for the continuous media. 
If the presentation is to be sent only to the client via unicast, the
client provides the destination for security reasons.

\item[Invitation of a media server to a conference:] A media server can
be ``invited'' to join an existing conference, either to play back media
into the presentation or to record all or a subset of the media in a
presentation.  This mode is useful for distributed teaching
applications.  Several parties in the conference may take turns
``pushing the remote control buttons''. 

\item[Addition of media to an existing presentation:] Particularly for
live presentations, it is useful if the server can tell the client about
additional media becoming available.

\end{description}

RTSP requests may be handled by proxies, tunnels and caches as in
HTTP/1.1 \cite{rfc2068}.

\subsection{Requirements}

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 \cite{rfc2119}.

\subsection{Terminology}

Some of the terminology has been adopted from HTTP/1.1 \cite{rfc2068}.
Terms not listed here are defined as in HTTP/1.1.

\begin{description}
\item[Aggregate control:] The control of the multiple streams using a
single timeline by the server.  For audio/video feeds, this means that the
client may issue a single play or pause message to control both the audio
and video feeds.

\item[Conference:] a multiparty, multimedia presentation, where
``multi'' implies greater than or equal to one.

\item[Client:] The client requests continuous media data from the media
server.

\item[Connection:] A transport layer virtual circuit established between
two programs for the purpose of communication.

\item[Container file:] A file which may contain multiple media streams
which often comprise a presentation when played together.  RTSP servers may
offer aggregate control on these files, though the concept of a container
file is not embedded in the protocol.

\item[Continuous media:] Data where there is a timing relationship
between source and sink; that is, the sink must reproduce the timing
relationship that existed at the source.  The most common examples of
continuous media are audio and motion video.  Continuous media can be
{\em real-time (interactive)}, where there is a ``tight'' timing
relationship between source and sink, or {\em streaming (playback)},
where the relationship is less strict.

\item[Entity:] The information transferred as the payload of a request
or response.  An entity consists of metainformation in the form of
entity-header fields and content in the form of an entity-body, as
described in Section~\ref{sec:entity}.

\item[Media initialization:] Datatype/codec specific initialization.
This includes such things as clockrates, color tables, etc.  Any
transport-independent information which is required by a client for
playback of a media stream occurs in the media initialization phase of
stream setup.

\item[Media parameter:] Parameter specific to a media type that may
be changed before or during stream playback.

\item[Media server:] The server providing playback or recording services
for one or more media streams.  Different media streams within a
presentation may originate from different media servers.  A media server
may reside on the same or a different host as the web server the
presentation is invoked from.

\item[Media server indirection:] Redirection of a media client to a
different media server.

\item[(Media) stream:] A single media instance, e.g., an audio stream or
a video stream as well as a single whiteboard or shared application
group.  When using RTP, a stream consists of all RTP and RTCP packets
created by a source within an RTP session. This is equivalent to the
definition of a DSM-CC stream(\cite{ISO9511:13818-6}). 

\item[Message:] The basic unit of RTSP communication, consisting of a
structured sequence of octets matching the syntax defined in
Section~\ref{sec:syntax} and transmitted via a connection or a
connectionless protocol.

\item[Participant:] Member of a conference.  A participant may be a
machine, e.g., a media record or playback server.

\item[Presentation:] A set of one or more streams presented to the client
as a complete media feed, using a presentation description as defined
below.  In most cases in the RTSP context, this implies aggregate control
of those streams, but does not have to.

\item[Presentation description:] A presentation description contains
information about one or more media streams within a presentation, such
as the set of encodings, network addresses and information about the
content.  Other IETF protocols such as SDP (RFC 2327 \cite{rfc2327}) use
the term ``session'' for a live presentation.  The presentation
description may take several different formats, including but not
limited to the session description format SDP.

\item[Response:] An RTSP response. If an HTTP response is meant, that is
indicated explicitly.

\item[Request:] An RTSP request. If an HTTP request is meant, that is
indicated explicitly.

\item[RTSP session:] A complete RTSP ``transaction'', e.g., the viewing
of a movie.  A session typically consists of a client setting up a
transport mechanism for the continuous media stream (\header{SETUP}),
starting the stream with \header{PLAY} or \header{RECORD}, and closing the
stream with \header{TEARDOWN}.

\item[Transport initialization:] The negotiation of transport
information (e.g., port numbers, transport protocols) between the
client and the server.

\end{description}

\subsection{Protocol Properties}

RTSP has the following properties:

\begin{description}
\item[Extendable:] New methods and parameters can be easily added to
RTSP.

\item[Easy to parse:] RTSP can be parsed by standard HTTP or MIME
parsers.

\item[Secure:] RTSP re-uses web security mechanisms, either at the
transport level (TLS, RFC 2246 \cite{rfc2246}) or within the protocol
itself.  All HTTP authentication mechanisms such as basic (RFC 2068
\cite[Section~11.1]{rfc2068}) and digest authentication (RFC 2069
\cite{rfc2069}) are directly applicable.

\item[Transport-independent:] RTSP may use either an unreliable datagram
protocol (UDP) (RFC 768 \cite{rfc768}), a reliable datagram protocol
(RDP, RFC 1151, not widely used \cite{rfc1151}) or a reliable stream
protocol such as TCP (RFC 793 \cite{rfc793}) as it implements
application-level reliability. 

\item[Multi-server capable:] Each media stream within a presentation can
reside on a different server.  The client automatically establishes
several concurrent control sessions with the different media servers. 
Media synchronization is performed at the transport level.

\item[Control of recording devices:] The protocol can control both
recording and playback devices, as well as devices that can alternate
between the two modes (``VCR'').

\item[Separation of stream control and conference initiation:] Stream
control is divorced from inviting a media server to a conference.  The
only requirement is that the conference initiation protocol either
provides or can be used to create a unique conference identifier.  In
particular, SIP \cite{rfc2543} or H.323 \cite{H.323} may be used to
invite a server to a conference.

\item[Suitable for professional applications:] RTSP supports frame-level
accuracy through SMPTE time stamps to allow remote digital editing.

\item[Presentation description neutral:] The protocol does not impose a
particular presentation description or metafile format and can convey
the type of format to be used.  However, the presentation description
must contain at least one RTSP URI.

\item[Proxy and firewall friendly:] The protocol should be readily
handled by both application and transport-layer (SOCKS \cite{rfc1961})
firewalls.  A firewall may need to understand the \header{SETUP} method
to open a ``hole'' for the UDP media stream.

\item[HTTP-friendly:] Where sensible, RTSP reuses HTTP concepts, so that
the existing infrastructure can be reused.  This infrastructure includes
PICS (Platform for Internet Content Selection
\cite{Mill9610:Rating,Mill9610:PICS}) for associating labels with
content.  However, RTSP does not just add methods to HTTP since the
controlling continuous media requires server state in most cases.

\item[Appropriate server control:] If a client can start a stream, it
must be able to stop a stream.  Servers should not start streaming to
clients in such a way that clients cannot stop the stream.

\item[Transport negotiation:] The client can negotiate the transport
method prior to actually needing to process a continuous media stream.

\item[Capability negotiation:] If basic features are disabled, there
must be some clean mechanism for the client to determine which methods
are not going to be implemented.  This allows clients to present the
appropriate user interface.  For example, if seeking is not allowed, the
user interface must be able to disallow moving a sliding position
indicator.

\end{description}

\motivation{An earlier requirement in RTSP was multi-client capability. 
However, it was determined that a better approach was to make sure that
the protocol is easily extensible to the multi-client scenario.  Stream
identifiers can be used by several control streams, so that ``passing
the remote'' would be possible.  The protocol would not address how
several clients negotiate access; this is left to either a ``social
protocol'' or some other floor control mechanism.}

\subsection{Extending RTSP}

Since not all media servers have the same functionality, media servers
by necessity will support different sets of requests. For example:

\begin{itemize}
\item A server may only be capable of playback thus has no need to
support the \header{RECORD} request.

\item A server may not be capable of seeking (absolute positioning)
if it is to support live events only.

\item Some servers may not support setting stream parameters and thus
not support \header{GET\_PARAMETER} and \header{SET\_PARAMETER}. 

\end{itemize}

A server SHOULD implement all header fields described in
Section~\ref{sec:headers}.

It is up to the creators of presentation descriptions not to ask the
impossible of a server.  This situation is similar in HTTP/1.1
\cite{rfc2068}, where the methods described in [H19.6] are not likely to
be supported across all servers.

RTSP can be extended in three ways, listed here in order of the
magnitude of changes supported:

\begin{itemize}
\item Existing methods can be extended with new parameters, as long as
these parameters can be safely ignored by the recipient. (This is
equivalent to adding new parameters to an HTML tag.)  If the client needs
negative acknowledgement when a method extension is not supported,
a tag corresponding to the extension may be added in the \header{Require:}
field (see Section~\ref{sec:Require}).

\item New methods can be added.  If the recipient of the message does
not understand the request, it responds with error code 501 (Not
Implemented) and the sender should not attempt to use this method again. 
A client may also use the \header{OPTIONS} method to inquire about
methods supported by the server.  The server SHOULD list the methods it
supports using the \header{Public} response header.

\item A new version of the protocol can be defined, allowing almost all
aspects (except the position of the protocol version number) to change.

\end{itemize}

\subsection{Overall Operation}

Each presentation and media stream may be identified by an RTSP URL. 
The overall presentation and the properties of the media the
presentation is made up of are defined by a presentation description
file, the format of which is outside the scope of this specification. 
The presentation description file may be obtained by the client using
HTTP or other means such as email and may not necessarily be stored on
the media server.

For the purposes of this specification, a presentation description is
assumed to describe one or more presentations, each of which maintains a
common time axis.  For simplicity of exposition and without loss of
generality, it is assumed that the presentation description contains
exactly one such presentation.  A presentation may contain several media
streams.

The presentation description file contains a description of the media
streams making up the presentation, including their encodings, language,
and other parameters that enable the client to choose the most
appropriate combination of media.  In this presentation description,
each media stream that is individually controllable by RTSP is
identified by an RTSP URL, which points to the media server handling
that particular media stream and names the stream stored on that server. 
Several media streams can be located on different servers; for example,
audio and video streams can be split across servers for load sharing. 
The description also enumerates which transport methods the server is
capable of.

Besides the media parameters, the network destination address and port
need to be determined.  Several modes of operation can be distinguished:

\begin{description}
\item[Unicast:] The media is transmitted to the source of the RTSP
request, with the port number chosen by the client.  Alternatively, the
media is transmitted on the same reliable stream as RTSP.

\item[Multicast, server chooses address:] The media server picks the
multicast address and port.  This is the typical case for a live or
near-media-on-demand transmission.

\item[Multicast, client chooses address:] If the server is to
participate in an existing multicast conference, the multicast address,
port and encryption key are given by the conference description,
established by means outside the scope of this specification.

\end{description}

\subsection{RTSP States}
\label{sec:rtsp-states}

RTSP controls a stream which may be sent via a separate protocol,
independent of the control channel.  For example, RTSP control may occur
on a TCP connection while the data flows via UDP.  Thus, data delivery
continues even if no RTSP requests are received by the media server. 
Also, during its lifetime, a single media stream may be controlled by
RTSP requests issued sequentially on different TCP connections. 
Therefore, the server needs to maintain ``session state'' to be able to
correlate RTSP requests with a stream.  The state transitions are
described in Section~\ref{sec:machine}.

Many methods in RTSP do not contribute to state.  However, the following
play a central role in defining the allocation and usage of stream
resources on the server:  \header{SETUP}, \header{PLAY},
\header{RECORD}, \header{PAUSE}, and \header{TEARDOWN}.

\begin{description}
\item[\header{SETUP}:] Causes the server to allocate resources for a
stream and start an RTSP session.

\item[\header{PLAY} and \header{RECORD}:] Starts data transmission on a
stream allocated via \header{SETUP}.

\item[\header{PAUSE}:] Temporarily halts a stream without freeing server
resources.

\item[\header{TEARDOWN}:] Frees resources associated with the stream. 
The RTSP session ceases to exist on the server.

RTSP methods that contribute to state use the \header{Session} header
field (Section~\ref{sec:Session}) to identify the RTSP session whose
state is being manipulated.  The server generates session identifiers in
response to \header{SETUP} requests (Section~\ref{sec:SETUP}).

\end{description}

\subsection{Relationship with Other Protocols}

RTSP has some overlap in functionality with HTTP.  It also may interact
with HTTP in that the initial contact with streaming content is often to
be made through a web page.  The current protocol specification aims to
allow different hand-off points between a web server and the media
server implementing RTSP.  For example, the presentation description can
be retrieved using HTTP or RTSP, which reduces roundtrips in
web-browser-based scenarios, yet also allows for standalone RTSP servers
and clients which do not rely on HTTP at all.

However, RTSP differs fundamentally from HTTP in that data delivery
takes place out-of-band in a different protocol.  HTTP is an asymmetric
protocol where the client issues requests and the server responds.  In
RTSP, both the media client and media server can issue requests.  RTSP
requests are also not stateless; they may set parameters and continue to
control a media stream long after the request has been acknowledged.

\motivation{Re-using HTTP functionality has advantages in at least two
areas, namely security and proxies.  The requirements are very similar,
so having the ability to adopt HTTP work on caches, proxies and
authentication is valuable.}

While most real-time media will use RTP as a transport protocol, RTSP
is not tied to RTP.

RTSP assumes the existence of a presentation description format that can
express both static and temporal properties of a presentation containing
several media streams.
