You are getting something wrong... RTSP is a realtime streaming protocol. Meaning, you can stream whatever you want in real time. So you can use it to stream LIVE content (no matter what it is, video, audio, text, presentation...). RTP is a transport protocol which is used to transport media data which is negotiated over RTSP.

You use RTSP to control media transmission over RTP. You use it to setup, play, pause, teardown the stream...

So, if you want your server to just start streaming when the URL is requested, you can implement some sort of RTP-only server. But if you want more control and if you are streaming live video, you must use RTSP, because it transmits SDP and other important decoding data.

Read the documents I linked here, they are a good starting point.

AFAIK, RTSP does not transmit streams at all, it is just an out-of-band control protocol with functions like PLAY and STOP.

Raw UDP or RTP over UDP are transmission protocols for streams just like raw TCP or HTTP over TCP.

To be able to stream a certain program over the given transmission protocol, an encapsulation method has to be defined for your container format. For example TS container can be transmitted over UDP but Matroska can not.

Pretty much everything can be transported through TCP though.

(The fact that which codec do you use also matters indirectly as it restricts the container formats you can use.)

Some basics:

RTSP server can be used for dead source as well as for live source. RTSP protocols provides you commands (Like your VCR Remote), and functionality depends upon your implementation.

RTP is real time protocol used for transporting audio and video in real time. Transport used can be unicast, multicast or broadcast, depending upon transport address and port. Besides transporting RTP does lots of things for you like packetization, reordering, jitter control, QoS, support for Lip sync.....

In your case if you want broadcasting streaming server then you need both RTSP (for control) as well as RTP (broadcasting audio and video)

To start with you can go through sample code provided by live555

I hear your pain. I m going through this right now (years later). From what I ve learned, you can think of RTSP as a "VCR controller", the protocol allows you to specify which streams (presentations) you want to play, it will then send you a description of the media, and then you can use RTSP to play, stop, pause, and record the remote stream. The media itself goes over RTP. RTSP is normally implemented over a different socket or communication layer. Although it is simply a protocol, most often it s implemented by a server over a socket. For live streams, the RTSP stream you request is simply a name of a stream. It doesn t need to refer to a file on the server, the server s RTSP implementation can parse that stream, put together a live graph, and then provide the SDP (description) for that stream name. But, this is of course specific to the way the RTSP server has been implemented. For "live" streams, it s probably simpler to just use RTP, but you ll need a way to transfer the SDP from the RTP server to the client that wants to play that stream.

I think thats correct. RTSP may use RTP internally.

RTSP is widely used in IP camera, running as RTSP server in camera, so that user could play(pull) the RTSP stream from camera. It is a low cost solution, because we don t need a central media server (think about thousands of camera streams). The arch is bellow:

IP Camera      ----RTSP(pull)--->    Player
(RTSP server)                      (User Agent)

The RTSP protocol actually contains:

• A signaling over TCP, at port 554, used to exchange the SDP (also used in WebRTC), about media capabilities.
• UDP/TCP streams over serval ports, generally two ports, one for RTCP and one for RTP (also used in WebRTC).

Comparing to WebRTC, which is now available in H5:

• A signaling over HTTP/WebSocket or exchange by any other protocols, used to exchange the SDP.
• UDP streams(RTP/RTCP) over one or many ports, generally bind to one port, to make cloud services load balancer happy.

In protocol view, RTSP and WebRTC are similar, but the use scenario is very different, because it s off the topic, let s grossly simplified, WebRTC is design for web conference, while RTSP is used for IP camera systems.

So it s clear both RTSP and WebRTC are solution and protocol, used in different scenario. While RTP is transport protocol, also it can be used in live streaming by WebRTC.

Note: RTSP is not available for H5 or internet live streaming, but we could covert it by FFmpeg and a gateway server, please see here.

RTP is the transport protocol for real-time data. It provides timestamp, sequence number, and other means to handle the timing issues in real-time data transport.

RTSP is a control protocol that initiates and directs delivery of streaming multimedia data from media servers. It is the "Internet VCR remote control protocol." Its role is to provide the remote control; how­ever, the actual data delivery is done separately, most likely by RTP.

also, RTCP is the control part of RTP that helps with quality of service and membership management.

These three related protocols are used for real-time multimedia data over the Internet. Read the excellent full documentation at this link: RTP, RTCP & RTSP

RTSP (actually RTP) can be used for streaming video, but also many other types of media including live presentations. Rtsp is just the protocol used to setup the RTP session.

For all the details you can check out my open source RTSP Server implementation on the following address: https://net7mma.codeplex.com/

Or my article @ http://www.codeproject.com/Articles/507218/Managed-Media-Aggregation-using-Rtsp-and-Rtp

It supports re-sourcing streams as well as the dynamic creation of streams, various RFC s are implemented and the library achieves better performance and less memory then FFMPEG and just about any other solutions in the transport layer and thus makes it a good candidate to use as a centralized point of access for most scenarios.