RTSP introduction

Convention:

C denotes Client

S denotes Server

For the RTSP,RTP documents,you can get from the RFC website .

Real Time Streaming Protocol (RTSP)

In the modern Internet, applications are required to deliver value. One of the biggest conundrums in recent years has been   the battle to actually make the Internet a viable platform for making money. As we'll see throughout the course of this book,   one of the biggest drivers for delivering on the "Gold Rush" promise of Internet technologies is content. Making content attractive to end consumers to the point where they are willing to pay is a big challenge and one that has   been aided by the delivery of Application layer protocols such as RTSP, which enables the delivery of real-time video and audio in variable qualities. The other Application layer protocols we've   looked at so far in this chapter work in a request/response manner, whereby the client asks for some piece of content, the   content is delivered using TCP or UDP, and then the client application can display the content to the user. While these mechanisms are suitable for a large number   of applications in the Internet, there also exists a requirement to deliver content, be it images, audio, video, or a combination   of all three, in real time. Imagine if a user were to try to watch a full-screen video file of a one-hour movie using HTTP or FTP as the Application layer protocol. The movie file might be several hundred megabytes, if not several gigabytes, in size.   Even with modern broadband services deliverable to the home, this type of large file size does not fit well in the "download   then play" model we saw previously.

RTSP uses a combination of reliable transmission over TCP (used for control) and best-efforts delivery over UDP (used for content) to stream content to users. By this, we mean that the file delivery can start and the client-side application   can begin displaying the audio and video content before the complete file has arrived. In terms of our one-hour movie example,   this means that the client can request a movie file and watch a "live" feed similar to how one would watch a TV. Along with   this "on demand" type service, RTSP also enables the delivery of live broadcast content that would not be possible with traditional download and play type mechanisms.

The Components of RTSP Delivery

During our look at RTSP, we'll use the term to describe a number of protocols that work together in delivering content to the user.

RTSP

RTSP is the control protocol for the delivery of multimedia content across IP networks. It is based typically on TCP for reliable delivery and has a very similar operation and syntax to HTTP. RTSP is used by the client application to communicate to the server information such as the media file being requested, the type   of application the client is using, the mechanism of delivery of the file (unicast or multicast, UDP or TCP), and other important control information commands such as DESCRIBE, SETUP, and PLAY. The actual multimedia content is not   typically delivered over the RTSP connection(s), although it can be interleaved if required. RTSP is analogous to the remote control of the streaming protocols.

Real Time Transport Protocol (RTP)

RTP is the protocol used for the actual transport and delivery of the real-time audio and video data. As the delivery of the   actual data for audio and video is typically delay sensitive, the lighter weight UDP protocol is used as the Layer 4 delivery mechanism, although TCP might also be used in environments that suffer higher packet loss. The RTP flow when delivering the content is unidirectional from the server to the client. One interesting part of the RTP operation is that the source port used by the server when sending the UDP data is always even—although it is dynamically assigned. The destination port (i.e., the UDP port on which the client is listening) is chosen by the client and communicated over the RTSP control connection.

Real Time Control Protocol (RTCP)

RTCP is a complimentary protocol to RTP and is a bidirectional UDP-based mechanism to allow the client to communicate stream-quality information back to the object server. The RTCP UDP communication always uses the next UDP source port up from that used by the RTP stream, and consequently is always odd.Figure 1 shows how the three protocols work together.

Figure 1 The three main application protocols used in real-time streaming.

 

RTSP Operation

The RTSP protocol is very similar in structure and specifically syntax to HTTP. Both use the same URL structure to describe an object, with RTSP using the rtsp:// scheme rather than the http://. RTSP, however, introduces a number of additional headers (such as DESCRIBE, SETUP, and PLAY) and also allows data transport out-of-band   and over a different protocol, such as RTP described earlier. The best way to understand how the components described previously work together to deliver an audio/video   stream is to look at an example. The basic steps involved in the process are as follows:

1. The client establishes a TCP connection to the servers, typically on TCP port 554, the well-known port for RTSP.        

2. The client will then commence issuing a series of RTSP header commands that have a similar format to HTTP, each of which is acknowledged by the server. Within these RTSP commands, the client will describe to the server details of the session requirements, such as the version of RTSP it supports, the transport to be used for the data flow, and any associated UDP or TCP port information. This information is passed using the DESCRIBE and SETUP headers and is augmented on the server response            with a Session ID that the client, and any transitory proxy devices, can use to identify the stream in further exchanges.        

3. Once the negotiation of transport parameters has been completed, the client will issue a PLAY command to instruct the server            to commence delivery of the RTP data stream.        

4. Once the client decides to close the stream, a TEARDOWN command is issued along with the Session ID instructing the server            to cease the RTP delivery associated with that ID.

Example—RTSP with UDP-Based RTP Delivery

Let's consider an example interaction where the client and server will use a combination of TCP-based RTSP and UDP-based RTP and RTCP to deliver and view a video stream. In the first step, the client will establish a TCP connection to port 554 on the server and issue an OPTIONS command showing the protocol version used for the session. The   server acknowledges this with a 200 OK message, similar to HTTP.

C->S  OPTIONS rtsp://video.foocorp.com:554 RTSP/1.0
Cseq: 1

S->C  RTSP/1.0 200 OK
      Cseq: 1

Next, the client issues a DESCRIBE command that indicates to the server the URL of the media file being requested. The server responds with another 200 OK acknowledgment and includes a full media description   of the content, which is presented in either Session Description Protocol (SDP) or Multimedia and Hypermedia Experts Group (MHEG) format.

C->S  DESCRIBE rtsp://video.foocorp.com:554/streams/example.rm RTSP/1.0
      Cseq:2

S->C  RTSP/1.0 200 OK
      Cseq: 2
      Content-Type: application/sdp
Content-Length: 210
      <SDP Data...>

 In the third stage of the RTSP negotiation, the client issues a SETUP command that identifies to the server the transport mechanisms, in order of preference,   the client wants to use. We won't list all of the available transport options here (the RFC obviously contains an exhaustive   list), but we'll see the client request RTP over UDP on ports 5067 and 5068 for the data transport. The server responds with confirmation of the RTP over UDP transport mechanism and the client-side ports and includes the unique Session ID and server port information.

C->S  SETUP rtsp://video.foocorp.com:554/streams/example.rm RTSP/1.0
      Cseq: 3
      Transport: rtp/udp;unicast;client_port=5067-5068

S->C  RTSP/1.0 200 OK
      Cseq: 3
      Session: 12345678
      Transport: rtp/udp;client_port=5067-5068;server_port=6023-6024

Finally, the client is now ready to commence the receipt of the data stream and issues a PLAY command. This simply contains   the URL and Session ID previously provided by the server. The server acknowledges this PLAY command, and the RTP stream from the server to client will begin.

C->S  PLAY rtsp://video.foocorp.com:554/streams/example.rm RTSP/1.0
      Cseq: 4
      Session: 12345678

S->C  RTSP/1.0 200 OK
      Cseq: 4

Once the client decides that the stream can be stopped, a TEARDOWN command is issued over the RTSP connection referenced only by the Session ID. The server again acknowledges this and the RTP delivery will cease.

C->S  TEARDOWN rtsp://video.foocorp.com:554/streams/example.rm RTSP/1.0
      Cseq: 5
      Session: 12345678

S->C  RTSP/1.0 200 OK
      Cseq: 5

Figure 2 shows this example in a simplified graphic form