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 Benefits of using AJAX:

Bandwidth utilization

By generating the HTML locally within the browser, and only bringing down JavaScript calls and the actual data, Ajax web pages can appear to load relatively quickly since the payload coming down is much smaller in size. An example of this technique is a large result set where multiple pages of data exist. With Ajax, the HTML of the page, e.g., a table control and related TD and TR tags can be produced locally in the browser and not brought down with the first page of the document.

In addition to "load on demand" of contents, some web based applications load stubs of event handlers and then load the functions on the fly. This technique significantly cuts down the bandwidth consumption for web applications that have complex logic and functionality.

User interface

The most obvious reason for using Ajax is an improvement to the user experience. Pages using Ajax behave more like a standalone application than a typical web page. Clicking on links that cause the entire page to refresh feels like a "heavy" operation. With Ajax, the page often can be updated dynamically, allowing a faster response to the user's interaction. While the full potential of Ajax has yet to be determined, some believe it will prove to be an important technology, helping making the web even more interactive and popular than it currently is.

Separation of Data, Format, Style, and Function

A less specific benefit of the AJAX approach is that it tends to encourage programmers to clearly separate the methods and formats used for the different aspects of information delivery via the web. Although AJAX can appear to be a jumble of languages and techniques, and programmers are free to adopt and adapt whatever works for them, they are generally propelled by the development motif itself to adopt separation between:

(1) The raw data or content to be delivered - which is normally embedded in XML and sometimes derived from a server-side database.

(2) The format or structure of the webpage - which is almost always built in HTML (or better, XHTML) and is then reflected and made available to dynamic manipulation in the DOM

(3) The style elements of the webpage - everything from fonts to picture placement - are derived by reference to embedded or referenced CSS; and

(4) The functionality of the web page is provided by a combination of

(A) Javascript on the client browser (also called DHTML),

(B) Standard HTTP and XMLHttp for client-to-server communication, and

(C) server-side scripting and/or programs utilizing any suitable language preferred by the programmer to receive the client's specific requests and respond appropriately.

  Disadvantages of AJAX:

Browser integration

The dynamically created page does not register itself with the browser history engine, so triggering the "Back" function of the users' browser might not bring the desired result.

Developers have implemented various solutions to this problem. These solutions can involve using invisible IFRAMEs to invoke changes that populate the history used by a browser's back button. Google Maps, for example, performs searches in an invisible IFRAME and then pulls results back into an element on the visible web page.

Another issue is that dynamic web page updates make it difficult for a user to bookmark a particular state of the application. Solutions to this problem exist, many of which use the URL fragment identifier (the portion of a URL after the '#') to keep track of, and allow users to return to, the application in a given state. This is possible because many browsers allow JavaScript to update the fragment identifier of the URL dynamically, so that Ajax applications can maintain it as the user changes the application's state. This solution also improves back-button support. It is, however, not a complete solution.

Response-time concerns

Network latency — or the interval between user request and server response needs to be considered carefully during Ajax development. Without clear feedback to the user, smart preloading of data and proper handling of the XMLHttpRequest object, users might experience delay in the interface of the web application, something which users might not expect or understand. Additionally, when an entire page is rendered there is a brief moment of re-adjustment for the eye when the content changes. The lack of this re-adjustment with smaller portions of the screen changing makes the latency more apparent. The use of visual feedback (such as throbbers) to alert the user of background activity and/or preloading of content and data are often suggested solutions to these latency issues.

In general the potential impact of latency has not been "solved" by any of the open source Ajax toolkits and frameworks available today, such as the effect of latency variance over time.

Search Engine Optimization

Websites that use Ajax to load data which should be indexed by search engines must be careful to provide equivalent data at a public, linked URL and in a format that the search engine can read, as search engines do not generally execute the JavaScript code required for Ajax functionality. This problem is not specific to Ajax, as the same issue occurs with sites that provide dynamic data as a full-page refresh in response to, eg, a form submit (the general problem is sometimes called the hidden web).

Javascript reliability

Ajax relies on Javascript, which may be implemented differently by different browsers or versions of a particular browser. Because of this, sites that use Javascript may need to be tested in multiple browsers to check for compatibility issues. It's not uncommon to see a Javascript code written twice, a part for IE, a part for Mozilla compatibles.

AJAX is the buzz-word of the moment to web developers. It's a buzz-word that is hotly debated from many angles, but it has stuck as a buzz-word because it encapsulates something that is new from a user's perspective. The functionally that is newly available to all web users is "in-page replacement" - the ability for a web page to change using data from a web server without totally redrawing itself. This functionality has been around in Mozilla and Internet Explorer for a while, but it is only recently that Safari and Konqueror users have been able to join in.

Ajax, which stands for Asynchronous JavaScript and XML, is transforming the Web and Internet-based applications forever. The initial user interaction model of the Web is based on page-oriented HyperText Markup Language (HTML). It is designed to display text and graphic information easily in a web browser. Its simple publication and viewing model has helped fuel the explosive adoption of the World Wide Web into the global phenomenon we know today. But this growth has users wanting more; they want web applications to provide desktop-like user experiences, and HTML's page-oriented model falls short.

Ajax substantially improves the end user experience for web-based applications and provides a new user interaction model. In more technical terms, Ajax is a development technique in which Web pages act more like desktop applications because the page does not have to be reloaded on each user input. Instead of repeated page refreshes, small amounts of data are exchanged with the server behind the scenes and the application remains usable by the end-user.

Ajax technology has emerged as a critical technology for launching the Web's next generation, often referred to as Web 2.0. Ajax is increasing end user productivity and enabling new classes of collaboratory applications. In short, Ajax is redefining the Web at a rapid pace, thus requiring every organization with a Web presence or Internet-based business applications to develop a cohesive Ajax strategy.

p>An E-mail MessageThe first e-mail message was sent in 1971 by an
engineer named Ray Tomlinson. Prior to this, you could only send
messages to users on a single machine. Tomlinson's breakthrough was
the ability to send messages to other machines on the Internet, using
the @ sign to designate the receiving machine.
An e-mail message has always been nothing more than a simple text
message -- a piece of text sent to a recipient. In the beginning and
even today, e-mail messages tend to be short pieces of text, although
the ability to add attachments now makes many e-mail messages quite
long. Even with attachments, however, e-mail messages continue to be
text messages -- we'll see why when we get to the section on
attachments
E-mail Clients
You have probably already received several e-mail messages today. To
look at them, you use some sort of e-mail client. Many people use
well-known stand-alone clients like Microsoft Outlook, Outlook
Express, Eudora or Pegasus. People who subscribe to free e-mail
services like Hotmail or Yahoo use an e-mail client that appears in a
Web Page. If you are an AOL customer, you use AOL's e-mail reader. No
matter which type of client you are using, it generally does four
things:
It shows you a list of all of the messages in your mailbox by
displaying the message headers. The header shows you who sent the
mail, the subject of the mail and may also show the time and date of
the message and the message size.
It lets you select a message header and read the body of the e-mail message.
It lets you create new messages and send them. You type in the e-mail
address of the recipient and the subject for the message, and then
type the body of the message.
Most e-mail clients also let you add attachments to messages you send
and save the attachments from messages you receive.
Sophisticated e-mail clients may have all sorts of bells and whistles,
but at the core, this is all that an e-mail client does.

A Simple E-mail Server
Given that you have an e-mail client on your machine, you are ready to
send and receive e-mail. All that you need is an e-mail server for the
client to connect to. Let's imagine what the simplest possible e-mail
server would look like in order to get a basic understanding of the
process. Then we will look at the real thing.
You know that machines on the Internet can run software applications
that act as servers. There are Web servers, FTP servers, telnet
servers and e-mail servers running on millions of machines on the
Internet right now. These applications run all the time on the server
machine and they listen to specific ports, waiting for people or
programs to attach to the port . The simplest possible e-mail server
would work something like this:
It would have a list of e-mail accounts, with one account for each
person who can receive e-mail on the server. My account name might be
mbrain, John Smith's might be jsmith, and so on.
It would have a text file for each account in the list. So the server
would have a text file in its directory named MBRAIN.TXT, another
named JSMITH.TXT, and so on.
If someone wanted to send me a message, the person would compose a
text message ("Marshall, Can we have lunch Monday? John") in an e-mail
client, and indicate that the message should go to mbrain. When the
person presses the Send button, the e-mail client would connect to the
e-mail server and pass to the server the name of the recipient
(mbrain), the name of the sender (jsmith) and the body of the message.
The server would format those pieces of information and append them to
the bottom of the MBRAIN.TXT file. The entry in the file might look
like this:
From: jsmith To: mbrain Marshall, Can we have lunch Monday? John
There are several other pieces of information that the server might
save into the file, like the time and date of receipt and a subject
line; but overall, you can see that this is an extremely simple
process.
As other people sent mail to mbrain, the server would simply append
those messages to the bottom of the file in the order that they
arrived. The text file would accumulate a series of five or 10
messages, and eventually I would log in to read them. When I wanted to
look at my e-mail, my e-mail client would connect to the server
machine. In the simplest possible system, it would:
Ask the server to send a copy of the MBRAIN.TXT file
Ask the server to erase and reset the MBRAIN.TXT file
Save the MBRAIN.TXT file on my local machine
Parse the file into the separate messages (using the word "From:" as
the separator)
Show me all of the message headers in a list
When I double-clicked on a message header, it would find that message
in the text file and show me its body.
You have to admit that this is a very simple system. Surprisingly, the
real e-mail system that you use every day is not much more complicated
than this.

The Real E-mail System
For the vast majority of people right now, the real e-mail system
consists of two different servers running on a server machine. One is
called the SMTP server, where SMTP stands for Simple Mail Transfer
Protocol. The SMTP server handles outgoing mail. The other is either a
POP3 server or an IMAP server, both of which handle incoming mail. POP
stands for Post Office Protocol, and IMAP stands for Internet Mail
Access Protocol.

The SMTP server listens on well-known port number 25, POP3 listens on
port 110 and IMAP uses port 143

Because of the similarity in syntax and usage between RTSP servers and HTTP servers, the security considerations outlined in apply.
Authentication Mechanisms:
RTSP and HTTP share common authentication schemes, and thus should follow the same prescriptions with regards to authentication.
Abuse of Server Log Information:
RTSP and HTTP servers will presumably have similar logging mechanisms, and thus should be equally guarded in protecting the contents of those logs, thus protecting the privacy of the users of the servers.
Transfer of Sensitive Information:
There is no reason to believe that information transferred via RTSP may be any less sensitive than that normally transmitted via HTTP. Therefore, all of the precautions regarding the protection of data privacy and user privacy apply to implementors of RTSP clients, servers, and proxies.
Attacks Based on File and Path Names:
Though RTSP URLs are opaque handles that do not necessarily have file system semantics, it is anticipated that many implementations will translate portions of the request URLs directly to file system calls. In such cases, file systems SHOULD follow the precautions, such as checking for ".." in path components.
Personal Information:
RTSP clients are often privy to the same information that HTTP clients are (user name, location, etc.) and thus should be equally.
Privacy Issues Connected to Accept Headers:
Since may of the same "Accept" headers exist in RTSP as in HTTP with regards to their use should be followed.
DNS Spoofing:
Presumably, given the longer connection times typically associated to RTSP sessions relative to HTTP sessions, RTSP client DNS optimizations should be less prevalent. Nonetheless, the recommendations provided are still relevant to any implementation which attempts to rely on a DNS-to-IP mapping to hold beyond a single use of the mapping.
Location Headers and Spoofing:
If a single server supports multiple organizations that do not trust one another, then it must check the values of Location and Content-Location headers in responses that are generated under control of said organizations to make sure that they do not attempt to invalidate resources over which they have no authority.
The following are added considerations for RTSP implementations.
Concentrated denial-of-service attack:
The protocol offers the opportunity for a remote-controlled denial-of-service attack. The attacker may initiate traffic flows to one or more IP addresses by specifying them as the destination in SETUP requests. While the attacker's IP address may be known in this case, this is not always useful in prevention of more attacks or ascertaining the attackers identity. Thus, an RTSP server SHOULD only allow client- specified destinations for RTSP-initiated traffic flows if the server has verified the client's identity, either against a database of known users using RTSP authentication mechanisms (preferably digest authentication or stronger), or other secure means.
Session hijacking:
Since there is no relation between a transport layer connection and an RTSP session, it is possible for a malicious client to issue requests with random session identifiers which would affect unsuspecting clients. The server SHOULD use a large, random and non-sequential session identifier to minimize the possibility of this kind of attack.
Authentication:
Servers SHOULD implement both basic and digest authentication. In environments requiring tighter security for the control messages, the RTSP control stream may be encrypted.
Stream issues:
RTSP only provides for stream control. RTSP implementations will most likely rely on other protocols such as RTP, IP multicast, RSVP and IGMP, and should address security considerations brought up in those and other applicable specifications.
Persistently suspicious behavior:
RTSP servers SHOULD return error code 403 (Forbidden) upon receiving a single instance of behavior which is deemed a security risk. RTSP servers SHOULD also be aware of attempts to probe the server for weaknesses and entry points and MAY arbitrarily disconnect and ignore further requests clients which are deemed to be in violation of local security policy.

The conference participant client C asks the media server M to record the audio and video portions of a meeting. The client uses the ANNOUNCE method to provide meta-information about the recorded session to the server.
16 Syntax
The RTSP syntax is described in an augmented Backus-Naur form (BNF).
Base Syntax
OCTET =
CHAR =
UPALPHA =
LOALPHA =
ALPHA = UPALPHA | LOALPHA

DIGIT =
CTL = (octets 0 - 31) and DEL (127)>
CR =
LF =

SP =
HT =
<"> =
CRLF = CR LF
LWS = [CRLF] 1*( SP | HT )
TEXT =
tspecials = "(" | ")" | "<" | ">" | "@"
| "," | ";" | ":" | "\" | <">
| "/" | "[" | "]" | "?" | "="
| "{" | "}" | SP | HT

token = 1*
quoted-string = ( <"> *(qdtext) <"> )
qdtext = hats <">>
quoted-pair = "\" CHAR

message-header = field-name ":" [ field-value ] CRLF
field-name = token
field-value = *( field-content | LWS )
field-content = consisting of either *TEXT or
combinations of token, tspecials, and
quoted-string>

safe = "\$" | "-" | "_" | "." | "+"
extra = "!" | "*" | "$'$" | "(" | ")" | ","

hex = DIGIT | "A" | "B" | "C" | "D" | "E" | "F" |
"a" | "b" | "c" | "d" | "e" | "f"
escape = "\%" hex hex
reserved = ";" | "/" | "?" | ":" | "@" | "&" | "="

unreserved = alpha | digit | safe | extra
xchar = unreserved | reserved | escape

The following examples refer to stream description formats that are not standards, such as RTSL. The following examples are not to be used as a reference for those formats.
Media on Demand (Unicast)
Client C requests a movie from media servers a (audio.example.com) and V (video.example.com). The media description is stored on a web server W. The media description contains descriptions of the presentation and all its streams, including the codecs that are available, dynamic RTP payload types, the protocol stack, and content information such as language or copyright restrictions. It may also give an indication about the timeline of the movie.
Even though the audio and video track are on two different servers, and may start at slightly different times and may drift with respect to each other, the client can synchronize the two using standard RTP methods, in particular the time scale contained in the RTCP sender reports.
Streaming of a Container file
For purposes of this example, a container file is a storage entity in which multiple continuous media types pertaining to the same end-user presentation are present. In effect, the container file represents an RTSP presentation, with each of its components being RTSP streams. Container files are a widely used means to store such presentations. While the components are transported as independent streams, it is desirable to maintain a common context for those streams at the server end.
This enables the server to keep a single storage handle open easily. It also allows treating all the streams equally in case of any prioritization of streams by the server.
It is also possible that the presentation author may wish to prevent selective retrieval of the streams by the client in order to preserve the artistic effect of the combined media presentation. Similarly, in such a tightly bound presentation, it is desirable to be able to control all the streams via a single control message using an aggregate URL.
The following is an example of using a single RTSP session to control multiple streams. It also illustrates the use of aggregate URLs.
Client C requests a presentation from media server M . The movie is stored in a container file. The client has obtained an RTSP URL to the container file.
In the first instance of failure, the client tries to pause one stream (in this case video) of the presentation. This is disallowed for that presentation by the server. In the second instance, the aggregate URL may not be used for SETUP and one control message is required per stream to set up transport parameters.
This keeps the syntax of the Transport header simple and allows easy parsing of transport information by firewalls.
Single Stream Container Files
Some RTSP servers may treat all files as though they are "container files", yet other servers may not support such a concept. Because of this, clients SHOULD use the rules set forth in the session description for request URLs, rather than assuming that a consistent URL may always be used throughout
Live Media Presentation Using Multicast
The media server M chooses the multicast address and port. Here, we assume that the web server only contains a pointer to the full description, while the media server M maintains the full description.

In HTTP, response-request pairs are cached. RTSP differs
significantly in that respect. Responses are not cacheable, with the exception of the presentation description returned by DESCRIBE or included with ANNOUNCE. (Since the responses for anything but DESCRIBE and GET_PARAMETER do not return any data, caching is not really an issue for these requests.) However, it is desirable for the continuous media data, typically delivered out-of-band with respect to RTSP, to be cached, as well as the session description.
On receiving a SETUP or PLAY request, a proxy ascertains whether it has an up-to-date copy of the continuous media content and its description. It can determine whether the copy is up-to-date by issuing a SETUP or DESCRIBE request, respectively, and comparing the Last-Modified header with that of the cached copy. If the copy is not up-to-date, it modifies the SETUP transport parameters as appropriate and forwards the request to the origin server. Subsequent control commands such as PLAY or PAUSE then pass the proxy unmodified. The proxy delivers the continuous media data to the client, while possibly making a local copy for later reuse. The exact behavior allowed to the cache is given by the cache-response directives. A cache MUST answer any DESCRIBE requests if it is currently serving the stream to the requestor, as it is possible that low-level details of the stream description may have changed on the origin-server.
Note that an RTSP cache, unlike the HTTP cache, is of the "cut- through" variety. Rather than retrieving the whole resource from the origin server, the cache simply copies the streaming data as it passes by on its way to the client. Thus, it does not introduce additional latency.
To the client, an RTSP proxy cache appears like a regular media server, to the media origin server like a client. Just as an HTTP cache has to store the content type, content language, and so on for the objects it caches, a media cache has to store the presentation description. Typically, a cache eliminates all transport-references (that is, multicast information) from the presentation description, since these are independent of the data delivery from the cache to the client. Information on the encodings remains the same. If the cache is able to translate the cached media data, it would create a new presentation description with all the encoding possibilities it can offer.

Table summarizes the header fields used by RTSP. Type "g" designates general request headers to be found in both requests and responses, type "R" designates request headers, type "r" designates response headers, and type "e" designates entity header fields. Fields marked with "req." in the column labeled "support" MUST be implemented by the recipient for a particular method, while fields marked "opt." are optional. Note that not all fields marked "req." will be sent in every request of this type. The "req." means only that client (for response headers) and server (for request headers) MUST implement the fields. The last column lists the method for which this header field is meaningful; the designation "entity" refers to all methods that return a message body. Within this specification, DESCRIBE and GET_PARAMETER fall into this class.
Header type support methods
Accept R opt. entity
Accept-Encoding R opt. entity
Accept-Language R opt. all
Allow r opt. all
Authorization R opt. all
Bandwidth R opt. all
Blocksize R opt. all but OPTIONS, TEARDOWN
Cache-Control g opt. SETUP
Conference R opt. SETUP
Connection g req. all
Content-Base e opt. entity
Content-Encoding e req. SET_PARAMETER
Content-Encoding e req. DESCRIBE, ANNOUNCE
Content-Language e req. DESCRIBE, ANNOUNCE
Content-Length e req. SET_PARAMETER, ANNOUNCE
Content-Length e req. entity
Content-Location e opt. entity
Content-Type e req. SET_PARAMETER, ANNOUNCE
Content-Type r req. entity
CSeq g req. all
Date g opt. all
Expires e opt. DESCRIBE, ANNOUNCE
From R opt. all
If-Modified-Since R opt. DESCRIBE, SETUP
Last-Modified e opt. entity
Proxy-Authenticate
Proxy-Require R req. all
Public r opt. all
Range R opt. PLAY, PAUSE, RECORD
Range r opt. PLAY, PAUSE, RECORD
Referer R opt. all
Require R req. all
Retry-After r opt. all
RTP-Info r req. PLAY
Scale Rr opt. PLAY, RECORD
Session Rr req. all but SETUP, OPTIONS
Server r opt. all
Speed Rr opt. PLAY
Transport Rr req. SETUP
Unsupported r req. all
User-Agent R opt. all
Via g opt. all
WWW-Authenticate r opt. all
Overview of RTSP header fields
Accept
The Accept request-header field can be used to specify certain presentation description content types which are acceptable for the response.
The "level" parameter for presentation descriptions is properly defined as part of the MIME type registration, not here.
Example of use:
Accept: application/rtsl, application/sdp;level=2
Accept-Language
Note that the language specified applies to the presentation description and any reason phrases, not the media content.



Allow
The Allow response header field lists the methods supported by the resource identified by the request-URI. The purpose of this field is to strictly inform the recipient of valid methods associated with the resource. An Allow header field must be present in a 405 (Method not allowed) response.
Example of use:
Allow: SETUP, PLAY, RECORD, SET_PARAMETER
Bandwidth
The Bandwidth request header field describes the estimated bandwidth available to the client, expressed as a positive integer and measured in bits per second. The bandwidth available to the client may change during an RTSP session, e.g., due to modem retraining.
Bandwidth = "Bandwidth" ":" 1*DIGIT
Example:
Bandwidth: 4000
Block size
This request header field is sent from the client to the media server asking the server for a particular media packet size. This packet size does not include lower-layer headers such as IP, UDP, or RTP. The server is free to use a block size which is lower than the one requested. The server MAY truncate this packet size to the closest multiple of the minimum, media-specific block size, or override it with the media-specific size if necessary. The block size MUST be a positive decimal number, measured in octets. The server only returns an error (416) if the value is syntactically invalid.
Cache-Control
The Cache-Control general header field is used to specify directives that MUST be obeyed by all caching mechanisms along the
request/response chain.
Cache directives must be passed through by a proxy or gateway application, regardless of their significance to that application, since the directives may be applicable to all recipients along the request/response chain. It is not possible to specify a cache- directive for a specific cache.
Cache-Control should only be specified in a SETUP request and its response.
No-cache:
Indicates that the media stream MUST NOT is cached anywhere. This allows an origin server to prevent caching even by caches that have been configured to return stale responses to client requests.
Public:
Indicates that the media stream is cacheable by any cache.
Private:
Indicates that the media stream is intended for a single user and MUST NOT be cached by a shared cache. A private (non- shared) cache may cache the media stream.
No-transform:
An intermediate cache (proxy) may find it useful to convert the media type of a certain stream. A proxy might, for example, convert between video formats to save cache space or to reduce the amount of traffic on a slow link. Serious operational problems may occur, however, when these transformations have been applied to streams intended for certain kinds of applications. For example, applications for medical imaging, scientific data analysis and those using end-to-end authentication all depend on receiving a stream that is bit-for-bit identical to the original entity-body. Therefore, if a response includes the no-transform directive, an intermediate cache or proxy MUST NOT change the encoding of the stream. Unlike HTTP, RTSP does not provide for partial transformation at this point, e.g., allowing translation into a different language.
Only-if-cached:
In some cases, such as times of extremely poor network connectivity, a client may want a cache to return only those media streams that it currently has stored, and not to receive these from the origin server. To do this, the client may include the only-if-cached directive in a request. If it receives this directive, a cache SHOULD either respond using a cached media stream that is consistent with the other constraints of the request, or respond with a 504 (Gateway Timeout) status. However, if a group of caches is being operated as a unified system with good internal connectivity, such a request MAY be forwarded within that group of caches.
Max-stale:
Indicates that the client is willing to accept a media stream that has exceeded its expiration time. If max-stale is assigned a value, then the client is willing to accept a response that has exceeded its expiration time by no more than the specified number of seconds. If no value is assigned to max-stale, then the client is willing to accept a stale response of any age.
Min-fresh:
Indicates that the client is willing to accept a media stream whose freshness lifetime is no less than its current age plus the specified time in seconds. That is, the client wants a response that will still be fresh for at least the specified number of seconds.
Must-revalidate:
When the must-revalidate directive is present in a SETUP response received by a cache, that cache MUST NOT use the entry after it becomes stale to respond to a subsequent request without first revalidating it with the origin server. That is, the cache must do an end-to-end revalidation every time, if, based solely on the origin server's Expires, the cached response is stale.)
Conference
This request header field establishes a logical connection between a pre-established conference and an RTSP stream. The conference-id must not be changed for the same RTSP session.
Conference = "Conference" ":" conference-id Example:
Conference: 199702170042.SAA08642@obiwan.arl.wustl.edu%20Starr
A response code of 452 (452 Conference Not Found) is returned if the conference-id is not valid.
Content-Length
This field contains the length of the content of the method (i.e. after the double CRLF following the last header). Unlike HTTP, it MUST be included in all messages that carry content beyond the header portion of the message. If it is missing, a default value of zero is assumed.
Content-Type

Content types suitable for RTSP are likely to be restricted in practice to presentation descriptions and parameter-value types.

CSeq
The CSeq field specifies the sequence number for an RTSP request- response pair. This field MUST be present in all requests and responses. For every RTSP request containing the given sequence number, there will be a corresponding response having the same number. Any retransmitted request must contain the same sequence number as the original (i.e. the sequence number is not incremented for retransmissions of the same request).


Expires
The Expires entity-header field gives a date and time after which the description or media-stream should be considered stale. The interpretation depends on the method:
DESCRIBE response:
The Expires header indicates a date and time after which the description should be considered stale.
A stale cache entry may not normally be returned by a cache (either a proxy cache or a user agent cache) unless it is first validated with the origin server (or with an intermediate cache that has a fresh copy of the entity). The presence of an Expires field does not imply that the original resource will change or cease to exist at, before, or after that time.
The format is an absolute date and time as defined by HTTP-date; it MUST be in RFC1123-date format:
Expires = "Expires" ":" HTTP-date
An example of its use is
Expires: Thu, 01 Dec 1994 16:00:00 GMT
RTSP/1.0 clients and caches MUST treat other invalid date formats, especially including the value "0", as having occurred in the past (i.e., "already expired").
To mark a response as "already expired," an origin server should use an Expires date that is equal to the Date header value. To mark a response as "never expires," an origin server should use an Expires date approximately one year from the time the response is sent. RTSP/1.0 servers should not send Expires dates more than one year in the future.
The presence of an Expires header field with a date value of some time in the future on a media stream that otherwise would by default be non-cacheable indicates that the media stream is cacheable, unless indicated otherwise by a Cache-Control header field.
Host
This HTTP request header field is not needed for RTSP. It should be silently ignored if sent.


If-Match
This field is especially useful for ensuring the integrity of the presentation description, in both the case where it is fetched via means external to RTSP (such as HTTP), or in the case where the server implementation is guaranteeing the integrity of the description between the time of the DESCRIBE message and the SETUP message.
The identifier is an opaque identifier, and thus is not specific to any particular session description language.
If-Modified-Since
The If-Modified-Since request-header field is used with the DESCRIBE and SETUP methods to make them conditional. If the requested variant has not been modified since the time specified in this field, a description will not be returned from the server (DESCRIBE) or a stream will not be set up (SETUP). Instead, a 304 (not modified) response will be returned without any message-body.
If-Modified-Since = "If-Modified-Since" ":" HTTP-date

An example of the field is:
If-Modified-Since: Sat, 29 Oct 1994 19:43:31 GMT
Last-Modified
The Last-Modified entity-header field indicates the date and time at which the origin server believes the presentation description or media stream was last modified. For the methods DESCRIBE or ANNOUNCE, the header field indicates the last modification date and time of the description, for SETUP that of the media stream.
Proxy-Require
The Proxy-Require header is used to indicate proxy-sensitive features that MUST be supported by the proxy. Any Proxy-Require header features that are not supported by the proxy MUST be negatively acknowledged by the proxy to the client if not supported.
Range
This request and response header field specifies a range of time. The range can be specified in a number of units. This specification defines the smpte, npt, and clock range units. Within RTSP, byte ranges are not meaningful and MUST NOT be used. The header may also contain a time parameter in UTC, specifying the time at which the operation is to be made effective. Servers supporting the Range header MUST understand the NPT range format and SHOULD understand the SMPTE range format. The Range response header indicates what range of time is actually being played or recorded. If the Range header is given in a time format that is not understood, the recipient should return "501 Not Implemented".
Ranges are half-open intervals, including the lower point, but excluding the upper point. In other words, a range of a-b starts exactly at time a, but stops just before b. Only the start time of a media unit such as a video or audio frame is relevant. As an example, assume that video frames are generated every 40 ms. A range of 10.0- 10.1 would include a video frame starting at 10.0 or later time and would include a video frame starting at 10.08, even though it lasted beyond the interval. A range of 10.0-10.08, on the other hand, would exclude the frame at 10.08.
Range = "Range" ":" 1\#ranges-specifier
[“;" "time" "=" utc-time]
ranges-specifier = npt-range | utc-range | smpte-range
Example:
Range: clock=19960213T143205Z-; time=19970123T143720Z
The notation is similar to that used for the HTTP/1.1 byte- range header. It allows clients to select an excerpt from the media object, and to play from a given point to the end as well as from the current location to a given point. The start of playback can be scheduled for any time in the future, although a server may refuse to keep server resources for extended idle periods.
Require
The require header is used by clients to query the server about options that it may or may not support. The server MUST respond to this header by using the unsupported header to negatively acknowledge those options which are NOT supported.
This is to make sure that the client-server interaction will proceed without delay when all options are understood by both sides, and only slow down if options are not understood (as in the case above). For a well-matched client-server pair, the interaction proceeds quickly, saving a round-trip often required by negotiation mechanisms. In addition, it also removes state ambiguity when the client requires features that the server does not understand.
RTP-Info
This field is used to set RTP-specific parameters in the PLAY response.
URL:
Indicates the stream URL which for which the following RTP parameters correspond.
Seq:
Indicates the sequence number of the first packet of the stream. This allows clients to gracefully deal with packets when seeking. The client uses this value to differentiate packets that originated before the seek from packets that originated after the seek.
rtptime:
Indicates the RTP timestamp corresponding to the time value in the Range response header. (Note: For aggregate control, a particular stream may not actually generate a packet for the Range time value returned or implied. Thus, there is no guarantee that the packet with the sequence number indicated by seq actually has the timestamp indicated by rtptime.) The client uses this value to calculate the mapping of RTP time to NPT.
A mapping from RTP timestamps to NTP timestamps (wall clock) is available via RTCP. However, this information is not sufficient to generate a mapping from RTP timestamps to NPT. Furthermore, in order to ensure that this information is available at the necessary time (immediately at startup or after a seek), and that it is delivered reliably, this mapping is placed in the RTSP control channel.
In order to compensate for drift for long, uninterrupted presentations, RTSP clients should additionally map NPT to NTP, using initial RTCP sender reports to do the mapping, and later reports to check drift against the mapping.
Syntax:
RTP-Info = "RTP-Info" ":" 1#stream-url 1*parameter
Stream-url = "url" "=" URL
parameter = ";" "seq" "=" 1*DIGIT
| ";" "rtptime" "=" 1*DIGIT
Example:
RTP-Info: url=rtsp://foo.com/bar.avi/streamid=0;seq=45102, url=rtsp://foo.com/bar.avi/streamid=1;seq=30211
Scale
A scale value of 1 indicates normal play or record at the normal forward viewing rate. If not 1, the value corresponds to the rate with respect to normal viewing rate. For example, a ratio of 2 indicates twice the normal viewing rate ("fast forward") and a ratio of 0.5 indicates half the normal viewing rate. In other words, a ratio of 2 has normal play time increase at twice the wall clock rate. For every second of elapsed (wall clock) time, 2 seconds of content will be delivered. A negative value indicates reverse direction.
Unless requested otherwise by the Speed parameter, the data rate SHOULD not be changed. Implementation of scale changes depends on the server and media type. For video, a server may, for example, deliver only key frames or selected key frames. For audio, it may time-scale the audio while preserving pitch or, less desirably, deliver fragments of audio.
The server should try to approximate the viewing rate, but may restrict the range of scale values that it supports. The response MUST contain the actual scale value chosen by the server.
If the request contains a Range parameter, the new scale value will take effect at that time.
Scale = "Scale" ":" [“-”] 1*DIGIT [“." *DIGIT]
Example of playing in reverse at 3.5 times normal rate:
Scale: -3.5
Speed
This request header field’s parameter requests the server to deliver data to the client at a particular speed, contingent on the server's ability and desire to serve the media stream at the given speed. Implementation by the server is OPTIONAL. The default is the bit rate of the stream.
The parameter value is expressed as a decimal ratio, e.g., a value of 2.0 indicates that data is to be delivered twice as fast as normal. A speed of zero is invalid. If the request contains a Range parameter, the new speed value will take effect at that time.
Speed = "Speed" ":" 1*DIGIT [“." *DIGIT]
Example:
Speed: 2.5
Use of this field changes the bandwidth used for data delivery. It is meant for use in specific circumstances where preview of the presentation at a higher or lower rate is necessary. Implementers should keep in mind that bandwidth for the session may be negotiated beforehand (by means other than RTSP), and therefore re-negotiation may be necessary. When data is delivered over UDP, it is highly recommended that means such as RTCP be used to track packet loss rates.

Session
This request and response header field identifies an RTSP session started by the media server in a SETUP response and concluded by TEARDOWN on the presentation URL. The session identifier is chosen by the media server. Once a client receives a Session identifier, it MUST return it for any request related to that session. A server does not have to set up a session identifier if it has other means of identifying a session, such as dynamically generated URLs.
Session = "Session" ":" session-id [“;" "timeout" "=" delta-seconds]
The timeout parameter is only allowed in a response header. The server uses it to indicate to the client how long the server is prepared to wait between RTSP commands before closing the session due to lack of activity . The timeout is measured in seconds, with a default of 60 seconds (1 minute).
Note that a session identifier identifies a RTSP session across transport sessions or connections. Control messages for more than one RTSP URL may be sent within a single RTSP session. Hence, it is possible that clients use the same session for controlling many streams constituting a presentation, as long as all the streams come from the same server. However, multiple "user" sessions for the same URL from the same client MUST use different session identifiers.
The session identifier is needed to distinguish several delivery requests for the same URL coming from the same client.
The response 454 (Session Not found) is returned if the session identifier is invalid.
Timestamp
The timestamp general header describes when the client sent the request to the server. The value of the timestamp is of significance only to the client and may use any timescale. The server MUST echo the exact same value and MAY, if it has accurate information about this, add a floating point number indicating the number of seconds that has elapsed since it has received the request. The timestamp is used by the client to compute the round-trip time to the server so that it can adjust the timeout value for retransmissions.
Timestamp = "Timestamp" ":" *(DIGIT) [“." *(DIGIT)] [delay]
Delay = *(DIGIT) [“." *(DIGIT)]
Transport
This request header indicates which transport protocol is to be used and configures its parameters such as destination address, compression, and multicast time-to-live and destination port for a single stream. It sets those values not already determined by a presentation description. Transports are comma separated, listed in order of preference. Parameters may be added to each transport, separated by a semicolon.
The Transport header MAY also be used to change certain transport parameters. A server MAY refuse to change parameters of an existing stream.
The server MAY return a Transport response header in the response to indicate the values actually chosen. A Transport request header field may contain a list of transport options acceptable to the client. In that case, the server MUST return a single option which was actually chosen.
The syntax for the transport specifier is
Transport/profile/lower-transport.
The default value for the "lower-transport" parameters is specific to the profile. For RTP/AVP, the default is UDP.
Below are the configuration parameters associated with transport:
General parameters:
unicast | multicast:
mutually exclusive indication of whether unicast or multicast delivery will be attempted. Default value is multicast. Clients that are capable of handling both unicast and multicast transmission MUST indicate such capability by including two full transport-specs with separate parameters for each.
Destination:
The address to which a stream will be sent. The client may specify the multicast address with the destination parameter. To avoid becoming the unwitting perpetrator of a remote- controlled denial-of-service attack, a server SHOULD authenticate the client and SHOULD log such attempts before allowing the client to direct a media stream to an address not chosen by the server. This is particularly important if RTSP commands are issued via UDP, but implementations cannot rely on TCP as reliable means of client identification by itself. A server SHOULD not allow a client to direct media streams to an address that differs from the address commands are coming from.
Source:
If the source address for the stream is different than can be derived from the RTSP endpoint address (the server in playback or the client in recording), the source MAY be specified.
This information may also be available through SDP. However, since this is more a feature of transport than media initialization, the authoritative source for this information should be in the SETUP response.
Layers:
The number of multicast layers to be used for this media stream. The layers are sent to consecutive addresses starting at the destination address.
Mode:
The mode parameter indicates the methods to be supported for this session. Valid values are PLAY and RECORD. If not provided, the default is PLAY.
Append:
If the mode parameter includes RECORD, the append parameter indicates that the media data should append to the existing resource rather than overwrite it. If appending is requested and the server does not support this, it MUST refuse the request rather than overwrite the resource identified by the URI. The append parameter is ignored if the mode parameter does not contain RECORD.
Interleaved:
The interleaved parameter implies mixing the media stream with the control stream in whatever protocol is being used by the control stream. The argument provides the channel number to be used in the $ statement. This parameter may be specified as a range, e.g., interleaved=4-5 in cases where the transport choice for the media stream requires it.
This allows RTP/RTCP to be handled similarly to the way that it is done with UDP, i.e., one channel for RTP and the other for RTCP.
Multicast specific:
ttl:
multicast time-to-live
RTP Specific:
Port:
This parameter provides the RTP/RTCP port pair for a multicast session. It is specified as a range, e.g., port=3456-3457.
client_port:
This parameter provides the unicast RTP/RTCP port pair on which the client has chosen to receive media data and control information. It is specified as a range, e.g.,
client_port=3456-3457.
server_port:
This parameter provides the unicast RTP/RTCP port pair on which the server has chosen to receive media data and control information. It is specified as a range, e.g.,
server_port=3456-3457.
Ssrc:
The ssrc parameter indicates the RTP SSRC value that should be (request) or will be (response) used by the media server. This parameter is only valid for unicast transmission. It identifies the synchronization source to be associated with the media stream.
Example:
Transport: RTP/AVP; multicast; ttl=127; mode="PLAY",
RTP/AVP; unicast; client_port=3456-3457; mode="PLAY"
The Transport header is restricted to describing a single RTP stream. (RTSP can also control multiple streams as a single entity.) Making it part of RTSP rather than relying on a multitude of session description formats greatly simplifies designs of firewalls.
Unsupported

The Unsupported response header lists the features not supported by the server. In the case where the feature was specified via the Proxy-Require field, if there is a proxy on the path between the client and the server, the proxy MUST insert a message reply with an error message "551 Option Not Supported".

The method token indicates the method to be performed on the resource identified by the Request-URI. The method is case-sensitive. New methods may be defined in the future. Method names may not start with a $ character (decimal 24) and must be a token.
OPTIONS
An OPTIONS request may be issued at any time, e.g., if the client is about to try a nonstandard request. It does not influence server state.
DESCRIBE
The DESCRIBE method retrieves the description of a presentation or media object identified by the request URL from a server. It may use the Accept header to specify the description formats that the client understands. The server responds with a description of the requested resource. The DESCRIBE reply-response pair constitutes the media initialization phase of RTSP.

The DESCRIBE response MUST contain all media initialization information for the resource(s) that it describes. If a media client obtains a presentation description from a source other than DESCRIBE and that description contains a complete set of media initialization parameters, the client SHOULD use those parameters and not then request a description for the same media via RTSP.
Additionally, servers SHOULD NOT use the DESCRIBE response as a means of media indirection.
Clear ground rules need to be established so that clients have an unambiguous means of knowing when to request media initialization information via DESCRIBE, and when not to. By forcing a DESCRIBE response to contain all media initialization for the set of streams that it describes, and discouraging use of DESCRIBE for media indirection, we avoid looping problems that might result from other approaches.
Media initialization is a requirement for any RTSP-based system, but the RTSP specification does not dictate that this must be done via the DESCRIBE method. There are three ways that an RTSP client may receive initialization information:
* Via RTSP's DESCRIBE method;
* Via some other protocol (HTTP, email attachment, etc.);
* Via the command line or standard input (thus working as a browser
helper application launched with an SDP file or other media initialization format).

ANNOUNCE
The ANNOUNCE method serves two purposes:
When sent from client to server, ANNOUNCE posts the description of a presentation or media object identified by the request URL to a server. When sent from server to client, ANNOUNCE updates the session description in real-time.
If a new media stream is added to a presentation (e.g., during a live presentation), the whole presentation description should be sent again, rather than just the additional components, so that components can be deleted.
SETUP
The SETUP request for a URI specifies the transport mechanism to be used for the streamed media. A client can issue a SETUP request for a stream that is already playing to change transport parameters, which a server MAY allow. If it does not allow this, it MUST respond with error "455 Method Not Valid in This State". For the benefit of any intervening firewalls, a client must indicate the transport parameters even if it has no influence over these parameters, for example, where the server advertises a fixed multicast address.
Since SETUP includes all transport initialization information, firewalls and other intermediate network devices (which need this information) are spared the more arduous task of parsing the DESCRIBE response, which has been reserved for media initialization.
The Transport header specifies the transport parameters acceptable to the client for data transmission; the response will contain the transport parameters selected by the server.
The server generates session identifiers in response to SETUP requests. If a SETUP request to a server includes a session identifier, the server MUST bundle this setup request into the existing session or return error "459 Aggregate Operation Not Allowed”.
PLAY
The PLAY method tells the server to start sending data via the mechanism specified in SETUP. A client MUST NOT issue a PLAY request until any outstanding SETUP requests have been acknowledged as successful.
The PLAY request positions the normal play time to the beginning of the range specified and delivers stream data until the end of the range is reached. PLAY requests may be pipelined (queued); a server MUST queue PLAY requests to be executed in order. That is, a PLAY request arriving while a previous PLAY request is still active is delayed until the first has been completed.
This allows precise editing.
For example, regardless of how closely spaced the two PLAY requests in the example below arrive, the server will first play seconds 10 through 15, then, immediately following, seconds 20 to 25, and finally seconds 30 through the end.
C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0
CSeq: 835
Session: 12345678
Range: npt=10-15
C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0
CSeq: 836
Session: 12345678
Range: npt=20-25
C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0
CSeq: 837
Session: 12345678
Range: npt=30-
A PLAY request without a Range header is legal. It starts playing a stream from the beginning unless the stream has been paused. If a stream has been paused via PAUSE, stream delivery resumes at the pause point. If a stream is playing, such a PLAY request causes no further action and can be used by the client to test server liveness.
The Range header may also contain a time parameter. This parameter specifies a time in UTC at which the playback should start. If the message is received after the specified time, playback is started immediately. The time parameter may be used to aid in synchronization of streams obtained from different sources.
For an on-demand stream, the server replies with the actual range that will be played back. This may differ from the requested range if alignment of the requested range to valid frame boundaries is required for the media source. If no range is specified in the request, the current position is returned in the reply. The unit of the range in the reply is the same as that in the request.
After playing the desired range, the presentation is automatically paused, as if a PAUSE request had been issued. A media server only supporting playback MUST support the npt format and MAY support the clock and smpte formats.
PAUSE
The PAUSE request causes the stream delivery to be interrupted (halted) temporarily. If the request URL names a stream, only playback and recording of that stream is halted. For example, for audio, this is equivalent to muting. If the request URL names a presentation or group of streams, delivery of all currently active streams within the presentation or group is halted. After resuming playback or recording, synchronization of the tracks MUST be maintained. Any server resources are kept, though servers MAY close the session and free resources after being paused for the duration specified with the timeout parameter of the Session header in the SETUP message.
The PAUSE request may contain a Range header specifying when the stream or presentation is to be halted. We refer to this point as the "pause point". The header must contain exactly one value rather than a time range. The normal play time for the stream is set to the pause point. The pause request becomes effective the first time the server is encountering the time point specified in any of the currently pending PLAY requests. If the Range header specifies a time outside any currently pending PLAY requests, the error "457 Invalid Range" is returned. If a media unit (such as an audio or video frame) starts presentation at exactly the pause point, it is not played or recorded. If the Range header is missing, stream delivery is interrupted immediately on receipt of the message and the pause point is set to the current normal play time.
A PAUSE request discards all queued PLAY requests. However, the pause point in the media stream MUST be maintained. A subsequent PLAY request without Range header resumes from the pause point.
As another example, if a server has received requests to play ranges 10 to 15 and then 13 to 20 (that is, overlapping ranges), the PAUSE request for NPT=14 would take effect while the server plays the first range, with the second PLAY request effectively being ignored, assuming the PAUSE request arrives before the server has started playing the second, overlapping range. Regardless of when the PAUSE request arrives, it sets the NPT to 14.
If the server has already sent data beyond the time specified in the Range header, a PLAY would still resume at that point in time, as it is assumed that the client has discarded data after that point. This ensures continuous pause/play cycling without gaps.
TEARDOWN
The TEARDOWN request stops the stream delivery for the given URI, freeing the resources associated with it. If the URI is the presentation URI for this presentation, any RTSP session identifier associated with the session is no longer valid. Unless all transport parameters are defined by the session description, a SETUP request has to be issued before the session can be played again.
GET_PARAMETER
The GET_PARAMETER request retrieves the value of a parameter of a presentation or stream specified in the URI. The content of the reply and response is left to the implementation. GET_PARAMETER with no entity body may be used to test client or server liveness ("ping").
The "text/parameters" section is only an example type for parameter. This method is intentionally loosely defined with the intention that the reply content and response content will be defined after further experimentation.
SET_PARAMETER
This method requests to set the value of a parameter for a presentation or stream specified by the URI.
A request SHOULD only contain a single parameter to allow the client to determine why a particular request failed. If the request contains several parameters, the server MUST only act on the request if all of the parameters can be set successfully. A server MUST allow a parameter to be set repeatedly to the same value, but it MAY disallow changing parameter values.

Restricting setting transport parameters to SETUP is for the benefit of firewalls. The parameters are split in a fine-grained fashion so that there can be more meaningful error indications. However, it may make sense to allow the setting of several parameters if an atomic setting is desirable. Imagine device control where the client does not want the camera to pan unless it can also tilt to the right angle at the same time.
The "text/parameters" section is only an example type for parameter. This method is intentionally loosely defined with the intention that the reply content and response content will be defined after further experimentation.

REDIRECT
A redirect request informs the client that it must connect to another server location. It contains the mandatory header Location, which indicates that the client should issue requests for that URL. It may contain the parameter Range, which indicates when the redirection takes effect. If the client wants to continue to send or receive media for this URI, the client MUST issue a TEARDOWN request for the current session and a SETUP for the new session at the designated host.
RECORD
This method initiates recording a range of media data according to the presentation description. The timestamp reflects start and end time (UTC). If no time range is given, use the start or end time provided in the presentation description. If the session has already started, commence recording immediately.
The server decides whether to store the recorded data under the request-URI or another URI. If the server does not use the request- URI, the response SHOULD be 201 (Created) and contain an entity which describes the status of the request and refers to the new resource, and a Location header.
A media server supporting recording of live presentations MUST support the clock range format; the smpte format does not make sense.
Embedded (Interleaved) Binary Data
Certain firewall designs and other circumstances may force a server to interleave RTSP methods and stream data. This interleaving should generally be avoided unless necessary since it complicates client and server operation and imposes additional overhead. Interleaved binary data SHOULD only be used if RTSP is carried over TCP.
Stream data such as RTP packets is encapsulated by an ASCII dollar sign (24 hexadecimal), followed by a one-byte channel identifier, followed by the length of the encapsulated binary data as a binary, two-byte integer in network byte order. The stream data follows immediately afterwards, without a CRLF, but including the upper-layer protocol headers. Each $ block contains exactly one upper-layer protocol data unit, e.g., one RTP packet.
The channel identifier is defined in the Transport header with the interleaved parameter.
When the transport choice is RTP, RTCP messages are also interleaved by the server over the TCP connection. As a default, RTCP packets are sent on the first available channel higher than the RTP channel. The client MAY explicitly request RTCP packets on another channel. This is done by specifying two channels in the interleaved parameter of the Transport header.
RTCP is needed for synchronization when two or more streams are interleaved in such a fashion. Also, this provides a convenient way to tunnel RTP/RTCP packets through the TCP control connection when required by the network configuration and transfer them onto UDP when possible.

RTSP requests can be transmitted in several different ways:
* Persistent transport connections used for several
request-response transactions;
* One connection per request/response transaction;
* Connectionless mode.
The type of transport connection is defined by the RTSP URI. For the scheme "rtsp", a persistent connection is assumed, while the scheme "rtspu" calls for RTSP requests to be sent without setting up a connection.
Unlike HTTP, RTSP allows the media server to send requests to the media client. However, this is only supported for persistent connections, as the media server otherwise has no reliable way of reaching the client. Also, this is the only way that requests from media server to client are likely to traverse firewalls.
Pipelining
A client that supports persistent connections or connectionless mode MAY "pipeline" its requests (i.e., send multiple requests without waiting for each response). A server MUST send its responses to those requests in the same order that the requests were received.
Reliability and Acknowledgements
Requests are acknowledged by the receiver unless they are sent to a multicast group. If there is no acknowledgement, the sender may resend the same message after a timeout of one round-trip time (RTT). The round-trip time is estimated as in TCP, with an initial round-trip value of 500 ms. an implementation MAY cache the last RTT measurement as the initial value for future connections.
If a reliable transport protocol is used to carry RTSP, requests MUST NOT be retransmitted; the RTSP application MUST instead rely on the underlying transport to provide reliability.
If both the underlying reliable transport such as TCP and the RTSP application retransmit requests, it is possible that each packet loss results in two retransmissions. The receiver cannot typically take advantage of the application-layer retransmission since the transport stack will not deliver the application-layer
retransmission before the first attempt has reached the receiver. If the packet loss is caused by congestion, multiple retransmissions at different layers will exacerbate the congestion.
If RTSP is used over a small-RTT LAN, standard procedures for optimizing initial TCP round trip estimates, such as those used in T/TCP, can be beneficial.
The Timestamp header is used to avoid the retransmission ambiguity problem. Each request carries a sequence number in the CSeq header, which is incremented by one for each distinct request transmitted. If a request is repeated because of lack of acknowledgement, the request MUST carry the original sequence number (i.e., the sequence number is not incremented).
Systems implementing RTSP MUST support carrying RTSP over TCP and MAY support UDP. The default port for the RTSP server is 554 for both UDP and TCP.
A number of RTSP packets destined for the same control end point may be packed into a single lower-layer PDU or encapsulated into a TCP stream. RTSP data MAY be interleaved with RTP and RTCP packets. Unlike HTTP, an RTSP message MUST contain a Content-Length header whenever that message contains a payload. Otherwise, an RTSP packet is terminated with an empty line immediately following the last message header.

RTSP defines additional status codes and does not define some HTTP codes. After receiving and interpreting a request message, the recipient responds with an RTSP response message.
Response = Status-Line
*( general-header
| response-header
| entity-header )
CRLF
[ message-body ]
Status-Line
The first line of a Response message is the Status-Line, consisting of the protocol version followed by a numeric status code, and the textual phrase associated with the status code, with each element separated by SP characters. No CR or LF is allowed except in the final CRLF sequence.
Status-Line = RTSP-Version SP Status-Code SP Reason-Phrase CRLF

Status Code and Reason Phrase:
The Status-Code element is a 3-digit integer result code of the attempt to understand and satisfy the request.The Reason-Phrase is intended to give a short textual description of the Status-Code. The Status-Code is intended for use by automata and the Reason-Phrase is intended for the human user. The client is not required to examine or display the Reason- Phrase.
The first digit of the Status-Code defines the class of response. The last two digits do not have any categorization role. There are 5 values for the first digit:
* 1xx: Informational - Request received, continuing process
* 2xx: Success - The action was successfully received, understood,
and accepted
* 3xx: Redirection - Further action must be taken in order to
complete the request
* 4xx: Client Error - The request contains bad syntax or cannot be
fulfilled
* 5xx: Server Error - The server failed to fulfill an apparently
valid request
The individual values of the numeric status codes defined for RTSP/1.0, and an example set of corresponding Reason-Phrase's, are presented below. The reason phrases listed here are only recommended
- they may be replaced by local equivalents without affecting the
protocol. RTSP adopts most HTTP/1.1 status codes and
adds RTSP-specific status codes starting at x50 to avoid conflicts
with newly defined HTTP status codes.
RTSP status codes are extensible. RTSP applications are not required to understand the meaning of all registered status codes, though such understanding is obviously desirable. However, applications MUST understand the class of any status code, as indicated by the first digit, and treat any unrecognized response as being equivalent to the x00 status code of that class, with the exception that an unrecognized response MUST NOT be cached. For example, if an unrecognized status code of 431 is received by the client, it can safely assume that there was something wrong with its request and treat the response as if it had received a 400 status code. In such cases, user agents SHOULD present to the user the entity returned with the response.
Response Header Fields
The response-header fields allow the request recipient to pass additional information about the response which cannot be placed in the Status-Line. These header fields give information about the server and about further access to the resource identified by the Request-URI.
Response-header = Location
| Proxy-Authenticate
| Public
| Retry-After
| Server
| Vary
| WWW-Authenticate
Response-header field names can be extended reliably only in combination with a change in the protocol version. However, new or experimental header fields MAY be given the semantics of response- header fields if all parties in the communication recognize them to be response-header fields. Unrecognized header fields are treated as entity-header fields.

general-header = Cache-Control
| Connection
| Date
| Via
9.Request:
A request message from a client to a server or vice versa includes, within the first line of that message, the method to be applied to the resource, the identifier of the resource, and the protocol version in use.
Request = Request-Line
*( general-header
| request-header
| entity-header )
CRLF
[ message-body ]

RTSP requests always contain the absolute URL (that is, including the scheme, host and port) rather than just the absolute path.
HTTP/1.1 requires servers to understand the absolute URL, but clients are supposed to use the Host request header. This is purely needed for backward-compatibility with HTTP/1.0 servers, a consideration that does not apply to RTSP.
The asterisk "*" in the Request-URI means that the request does not apply to a particular resource, but to the server itself, and is only allowed when the method used does not necessarily apply to a resource.

RTSP is a text-based protocol and uses the ISO 10646 character set in UTF-8 encoding. Lines are terminated by CRLF, but receivers should be prepared to also interpret CR and LF by themselves as line terminators. Text-based protocols make it easier to add optional parameters in a self-describing manner. Since the number of parameters and the frequency of commands are low, processing efficiency is not a concern. Text-based protocols, if done carefully, also allow easy implementation of research prototypes in scripting languages such as Tcl, Visual Basic and Perl.
The 10646 character set avoids tricky character set switching, but is invisible to the application as long as US-ASCII is being used. This is also the encoding used for RTCP. ISO 8859-1 translates directly into Unicode with a high-order octet of zero.RTSP messages can be carried over any lower-layer transport protocol that is 8-bit clean.
Requests contain methods, the object the method is operating upon and parameters to further describe the method. Methods are idempotent, unless otherwise noted. Methods are also designed to require little or no state maintenance at the media server.
Message Length
When a message body is included with a message, the length of that body is determined by one of the following (in order of precedence):
1. Any response message which MUST NOT include a message body
(such as the 1xx, 204, and 304 responses) is always terminated
by the first empty line after the header fields, regardless of
the entity-header fields present in the message.

2 If a Content-Length header field is present, its value in bytes
represents the length of the message-body.
If this header field is not present, a value of zero is assumed.

3 By the server closing the connection. (Closing the connection
cannot be used to indicate the end of a request body, since
that would leave no possibility for the server to send back a
response.)
Given the moderate length of presentation descriptions returned, the server should always be able to determine its length, even if it is generated dynamically, making the chunked transfer encoding unnecessary. Even though Content-Length must be present if there is any entity body, the rules ensure reasonable behavior even if the length is not given explicitly.

RTSP Version
RTSP URL
The "rtsp" and "rtspu" schemes are used to refer to network resources via the RTSP protocol. This section defines the scheme-specific syntax and semantics for RTSP URLs.
rtsp_URL = (“rtsp:" | "rtspu :”)
"//" host [“:" port] [abs_path]
host = A legal Internet host domain name of IP address
(in dotted decimal form), as defined by
of RFC 1123 \cite{rfc1123}> port = *DIGIT .
The scheme rtsp requires that commands are issued via a reliable protocol (within the Internet, TCP), while the scheme rtspu identifies an unreliable protocol (within the Internet, UDP).
If the port is empty or not given, port 554 is assumed. The semantics are that the identified resource can be controlled by RTSP at the server listening for TCP (scheme "rtsp") connections or UDP (scheme "rtspu") packets on that port of host, and the Request-URI for the resource is rtsp_URL.
A presentation or a stream is identified by a textual media identifier, using the character set and escape conventions of URLs. URLs may refer to a stream or an aggregate of streams, i.e., a presentation.
For example, the RTSP URL:
rtsp://media.example.com:554/twister/audiotrack
identifies the audio stream within the presentation "twister", which can be controlled via RTSP requests issued over a TCP connection to port 554 of host media.example.com.
Also, the RTSP URL:
rtsp://media.example.com:554/twister
identifies the presentation "twister", which may be composed of audio and video streams.
This does not imply a standard way to reference streams in URLs. The presentation description defines the hierarchical relationships in the presentation and the URLs for the individual streams. A presentation description may name a stream "a.mov" and the whole presentation "b.mov".

The path components of the RTSP URL are opaque to the client and do not imply any particular file system structure for the server.
This decoupling also allows presentation descriptions to be used with non-RTSP media control protocols simply by replacing the scheme in the URL.
Conference Identifiers

Conference identifiers are opaque to RTSP and are encoded using standard URL encoding methods (i.e., LWS is escaped with %). They can contain any octet value. The conference identifier MUST be globally unique

Conference identifiers are used to allow RTSP sessions to obtain parameters from multimedia conferences the media server is participating in. These conferences are created by protocols outside the scope of this specification, e.g., H.323 or SIP. Instead of the RTSP client explicitly providing transport information, for example, it asks the media server to use the values in the conference description instead.

Session Identifiers
Session identifiers are opaque strings of arbitrary length. Linear white space must be URL-escaped. A session identifier MUST be chosen randomly and MUST be at least eight octets long to make guessing it more difficult.
SMPTE Relative Timestamps
A SMPTE relative timestamp expresses time relative to the start of the clip. Relative timestamps are expressed as SMPTE time codes for frame-level access accuracy. The time code has the format hours:minutes:seconds:frames.subframes, with the origin at the start of the clip. The default smpte format is "SMPTE 30 drop" format, with frame rate is 29.97 frames per second. Other SMPTE codes MAY be supported through the use of alternative use of "smpte time". For the "frames" field in the time value can assume the values 0 through 29. The difference between 30 and 29.97 frames per second is handled by dropping the first two frame indices (values 00 and 01) of every minute, except every tenth minute. If the frame value is zero, it may be omitted. Sub frames are measured in one-hundredth of a frame.
Examples:
smpte=10:12:33:20-


Normal Play Time:
Normal play time (NPT) indicates the stream absolute position relative to the beginning of the presentation. The timestamp consists of a decimal fraction. The part left of the decimal may be expressed in either seconds or hours, minutes, and seconds. The part right of the decimal point measures fractions of a second.
The beginning of a presentation corresponds to 0.0 seconds. Negative values are not defined. The special constant now is defined as the current instant of a live event. It may be used only for live events.
It is often digitally displayed on a VCR. NPT advances normally when in normal play mode (scale = 1), advances at a faster rate when in fast scan forward (high positive scale ratio), decrements when in scan reverse (high negative scale ratio) and is fixed in pause mode. NPT is (logically) equivalent to SMPTE time codes."
Absolute Time
Absolute time is expressed as ISO 8601 timestamps, using UTC (GMT). Fractions of a second may be indicated.
Example for November 8, 1996 at 14h37 and 20 and a quarter seconds UTC: 19961108T143720.25Z
Option Tags
Option tags are unique identifiers used to designate new options in RTSP. These tags are used in require and Proxy- Require header fields.
Syntax: option-tag = 1*xchar
The creator of a new RTSP option should either prefix the option with a reverse domain name (e.g., "com.foo.mynewfeature" is an apt name for a feature whose inventor can be reached at "foo.com"), or register the new option with the Internet Assigned Numbers Authority (IANA).

RTSP has the following properties:
Extendable:
New methods and parameters can be easily added to RTSP.
Easy to parse:
RTSP can be parsed by standard HTTP or MIME parsers.
Secure:
RTSP re-uses web security mechanisms. All HTTP authentication mechanisms such as basic (RFC 2068) and digest authentication (RFC 2069) are directly applicable. One may also reuse transport or network layer security mechanisms.
Transport-independent:
RTSP may use either an unreliable datagram protocol (UDP), a reliable datagram protocol or a reliable stream protocol such as TCP as it implements application-level reliability.
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.
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").
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 or H.323 may be used to invite a server to a conference.
Suitable for professional applications:
RTSP supports frame-level accuracy through SMPTE time stamps to allow remote digital editing.
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.
Proxy and firewall friendly:
The protocol should be readily handled by both application and transport-layer (SOCKS [14]) firewalls. A firewall may need to understand the SETUP method to open a "hole" for the UDP media stream.
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 [15, 16]) 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.
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.
Transport negotiation:
The client can negotiate the transport method prior to actually needing to process a continuous media stream.
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.
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.
Extending RTSP
RTSP can be extended in three ways, listed here in order of the magnitude of changes supported:
* 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 required field.
* 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 OPTIONS method to inquire about
methods supported by the server. The server SHOULD list the
methods it supports using the Public response header.
* A new version of the protocol can be defined, allowing almost all
aspects (except the position of the protocol version number) to
change.

RTCP is a part of RTP and helps with lip synchronization and QOS management, among others.
RTSP is a control protocol that initiating and directing delivery of streaming multimedia from media servers, the "Internet VCR remote control protocol". RTSP does not deliver data (though the RTSP connection may be used to tunnel RTP traffic for ease of use with firewalls and other network devices). RTP and RTSP will likely be used together in many systems, but either protocol can be used without the other. The RTSP draft contains a section on the use of RTP with RTSP.
RTP vs. RTSP
It’s important to distinguish between RTP and Real-Time Streaming protocol (RTSP), another transfer protocol. RTSP is used when viewers communicate with a unicast server. RTSP allows two-way communication; that is, viewers can communicate with the streaming server and do things like rewind the movie, go to a chapter, and so on. By contrast, RTP is a one-way protocol used to send live or stored streams from the server to the client.
Perceived benefits
RTSP provides the following specific benefits to Internet content providers and users:
· Provides for on-demand access of multimedia items such as stored real-time audio/video files, live real-time feeds, or stored non-real-time items with bidirectionality enabling full stream control
· Allows interoperability between client-server multimedia products from multiple vendors. Can be implemented across multiple operating system platforms (Macintosh, Windows 95/NT, Windows 3.1 for clients; Macintosh, Windows NT, UNIX variants on servers).
· Provides intellectual property rights protection
· Designed for scalability
· Addresses key concerns of Internet Content Providers and Users: quality of service, efficiency of delivery, rights management, measurement and provides a design aimed at high reliability over current net infrastructure including: UDP, TCP/IP, RTP, RTCP, SCP and IP Multicast
· low overhead on data delivery
· full exploitation of emerging technologies and protocols
· security

whether this happened to you? You try to start your computer. Everything seems fine until it gets to the Windows logo screen and then restarts again. If you let it go it will possibly go on forever.
Two main causes for this problem. Not saying that these are the only answers for this problem, however these are the most common for me that i found.
1.Faulty random access memory (ram)
Having faulty memory can cause a number of symptoms including a continual restart of Windows. I have tested this theory in a number of computers, including Acer and Dell computers. The computers were working fine until I installed the faulty memory. As soon as the bad memory is removed and the working memory is replaced the computer is fine again. Faulty memory can also cause your computer to show a complete black screen or even the blue screen of death. Also see

Computer has a black screen and will not Boot or
Startup

If you have two sticks of memory in your computer, remove one and then start your computer. If this doesn't work then swap them over and try the other one. If the computer still restarts after loading the Windows Logo screen, then try the next option
2.The hard drive is dead or the boot sector is destroyed
The computer will not boot or start if there is no hard drive to go to. I repaired a computer with exactly this problem. It loaded the windows logo screen and then kept restarting. I first thought it was a memory problem, however it turned out to be the hard drive. I removed the hard drive and placed it in an external case, and still could not access it. I fitted a new hard drive to the computer and put a Windows Xp disc in the cd drive. And away it went, formatting and installing Windows XP. The odd thing is that you always presume that the computer will come up with an error message saying that there is no disk present. But this doesn't always happen.

So next time your computer loads the Windows Logo screen and then restarts be sure to check for these two common problems before investigating the problem further.

In case of Vista, the default workgroup is MSHOME,and in case XP WORKGROUP. So make sure both system’s are on the same workgroup before attempting to share files between XP and Vista.

First thing to do:
In XP, go to Control Panel > System. Then on the “Computer Name” tab, click the change button, and edit the workgroup to be the same as Vista (MSHOME).


On the Vista machine:
1.First go to the Network and Sharing Centre and make sure you have file sharing turned on. Having your network set to Private also helps.



1.afterthat go to the folder you want to share and Right-click on the file to bring a menu up > Choose Properties > Then Sharing. Click on Advanced Sharing. More details on how to share a file.
2.Now click the Share this folder check box, and then click Permissions.
3.Let the Permission window should open.
4.After now click on add to add a user. Then type in guest in the textbox that says Enter the object names to select.
5.Click Check Names which should then find Vista’s guest account and prefix your PC’s name in this format: YourComputerName\guest.
6.Click Ok to go back to the Permissions window.
7.You should now have the Guest account under group or users names.
8.Highlight the guest account and give it the relevant permissions. (Full control, Change or Read)
9.Click Ok and your done!

This is how simple it is to sharing file worked between Windows XP computers , and now you can have it enabled for Vista too!

Here are some networking commands that i find out. This will help freshers to quickly understand networking basics .

1. IPCONFIG
2. PING
3. NSLOOKUP
4. TRACERT
5. ARP
6. NETSTAT
7. NBTSTAT
8. ROUTE
9. WHOIS database
11. TELNET
10. Installing IPv6

1.IPCONFIG
The IPCONFIG command is used to display internet configuration of the system.
In IPCONFIG the default is to display the IP address, subnet mask and default gateway for each
adapter bound to TCP/IP. It helps in detecting bad IP addresses, incorrect subnet masks,
and improper broadcast addresses.
EXAMPLE:
c:>ipconfig
Windows IP Configuration
Ethernet adapter Local Area Connection:
Connection-specific DNS Suffix . :
IP Address. . . . . . . . . . . . : 192.168.6.124
Subnet Mask . . . . . . . . . . . : 255.255.255.0
Default Gateway . . . . . . . . . : 192.168.6.254
Two common commands:
1.To display full configuration information
c:>ipconfig /all
Windows IP Configuration
Host Name . . . . . . . . . . . . : LABVIEW
Primary Dns Suffix . . . . . . . :
Node Type . . . . . . . . . . . . : Unknown
IP Routing Enabled. . . . . . . . : No
WINS Proxy Enabled. . . . . . . . : No

Ethernet Adapter Local Area Connection :
Connection-specific DNS Suffix . :
Description . . . . . . . . . . . : Broadcom NetXtreme Gigabit Etherne
Physical Address. . . . . . . . . : 00-16-17-20-1D-9F
Dhcp Enabled. . . . . . . . . . . : No
IP Address. . . . . . . . . . . . : 192.168.6.124
Subnet Mask . . . . . . . . . . . : 255.255.255.0
Default Gateway . . . . . . . . . : 192.168.6.254
DNS Servers . . . . . . . . . . . : 192.168.1.250


2.To release and renew (get) an address from the DHCP server
c:>ipconfig /release

c:>ipconfig /renew

2.PING
Ping command is used to indicates whether a remote host can be reached.
It also displays information about packet loss and packet delivery time.
If the host is reachable then it send replay message Replay from that specific IP address otherwise Destination host unreachable.
EXAMPLE:
C:>ping 192.1689.1.250
Reply from 192.168.1.250: bytes=32 time<1ms TTL=63 Reply from 192.168.1.250: bytes=32 time<1ms TTL=63 Reply from 192.168.1.250: bytes=32 time<1ms TTL=63 Reply from 192.168.1.250: bytes=32 time<1ms TTL=63 Ping statistics for 192.168.1.250: Packets: Sent = 4, Received = 4, Lost = 0 (0% loss), Approximate round trip times in milli-seconds: Minimum = 0ms, Maximum = 0ms, Average = 0ms
Normal Use: send four packets
Usage: ping target_host
EXAMPLE:
c:>ping 192.168.1.250

Send ping packets count times
Usage: ping -n count target_host
Example:
c:>ping -n 192.168.1.250

Send ping packets with a specific buffer size (byte)
Usage: ping -l size target_host
Example:
C:> ping -l 100 192.168.5.254

3.TRACERT
TRACERT command is used to print information about each routing hop that
packets take going from your system to a remote system.
EXAMPLE:
c:>tracert
Usage: tracert [-d] [-h maximum_hops] [-j host-list] [-w timeout] target_name
Options:
-d Do not resolve addresses to hostnames.
-h maximum_hops Maximum number of hops to search for target.
-j host-list Loose source route along host-list.
-w timeout Wait timeout milliseconds for each reply.

Real Time Streaming Protocol (RTSP)
Abstract:
The traffic in the internet is becoming more and more involved with multimedia applications. To cope with such bandwith-demanding data, specific protocols have been created. The most difficult area is realtime (or realtimish) streams that enable us to make such things as videophonecalls and remote groupworks in much more efficient way.
The Real Time Streaming Protocol, or RTSP, is an application-level client –server 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 provide a means for choosing delivery mechanisms based upon RTP.
The protocol defines the connection between a streaming media client and a streaming media server, and also provides a standard way for clients and servers from multiple vendors to stream multimedia content.
It takes advantage of Internet and Intranet infrastructure such as IP Multicast, RTCP and RTP, providing a means for choosing delivery mechanisms