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Makefile.amH A D19-Apr-2019556 1612

Makefile.inH A D03-May-202235.1 KiB1,002905

READMEH A D19-Apr-201914.1 KiB378293

gstrtpdec.cH A D19-Apr-201925.1 KiB899603

gstrtpdec.hH A D19-Apr-20193.5 KiB8932

gstrtsp.cH A D19-Apr-20192.8 KiB7525

gstrtsp.hH A D19-Apr-20192.2 KiB545

gstrtspext.cH A D19-Apr-20197.5 KiB269175

gstrtspext.hH A D19-Apr-20193.8 KiB8430

gstrtspsrc.cH A D27-Aug-2019285.4 KiB9,6337,076

gstrtspsrc.hH A D19-Apr-201910 KiB336197

meson.buildH A D19-Apr-2019433 1715

README

1RTSP source
2-----------
3
4The RTSP source establishes a connection to an RTSP server and sets up
5the UDP sources and RTP session handlers.
6
7An RTSP session is created as follows:
8
9- Parse RTSP URL:
10
11   ex:
12     rtsp://thread:5454/south-rtsp.mp3
13
14- Open a connection to the server with the url. All further conversation with
15  the server should be done with this connection. Each request/reply has
16  a CSeq number added to the header.
17
18- Send a DESCRIBE request for the url. We currently support a response in
19  SDP.
20
21  ex:
22
23    >> DESCRIBE rtsp://thread:5454/south-rtsp.mp3 RTSP/1.0
24    >> Accept: application/sdp
25    >> CSeq: 0
26    >>
27    << RTSP/1.0 200 OK
28    << Content-Length: 84
29    << Content-Type: application/sdp
30    << CSeq: 0
31    << Date: Wed May 11 13:09:37 2005 GMT
32    <<
33    << v=0
34    << o=- 0 0 IN IP4 192.168.1.1
35    << s=No Title
36    << m=audio 0 RTP/AVP 14
37    << a=control:streamid=0
38
39- Parse the SDP message, for each stream (m=) we create an GstRTPStream. We need
40  to allocate two local UDP ports for receiving the RTP and RTCP data because we
41  need to send the port numbers to the server in the next request.
42
43  In RTSPSrc we first create an element that can handle the udp://0.0.0.0:0 uri. This
44  will create an udp source element with a random port number. We get the port
45  number by getting the "port" property of the element after setting the element to
46  PAUSED. This element is used for the RTP packets and has to be an even number. If
47  the random port number is not an even number we retry to allocate a new udp source.
48
49  We then create another UDP source element with the next (uneven) port number to
50  receive the RTCP packet on. After this step we have two udp ports we can use to
51  accept RTP packets.
52
53    +-----------------+
54    | +------------+  |
55    | | udpsrc0    |  |
56    | |  port=5000 |  |
57    | +------------+  |
58    | +------------+  |
59    | | udpsrc1    |  |
60    | |  port=5001 |  |
61    | +------------+  |
62    +-----------------+
63
64- Send a SETUP message to the server with the RTP ports. We get the setup URI from
65  the a= attribute in the SDP message. This can be an absolute URL or a relative
66  url.
67
68  ex:
69
70    >> SETUP rtsp://thread:5454/south-rtsp.mp3/streamid=0 RTSP/1.0
71    >> CSeq: 1
72    >> Transport: RTP/AVP/UDP;unicast;client_port=5000-5001,RTP/AVP/UDP;multicast,RTP/AVP/TCP
73    >>
74    << RTSP/1.0 200 OK
75    << Transport: RTP/AVP/UDP;unicast;client_port=5000-5001;server_port=6000-6001
76    << CSeq: 1
77    << Date: Wed May 11 13:21:43 2005 GMT
78    << Session: 5d5cb94413288ccd
79    <<
80
81  The client needs to send the local ports to the server along with the supported
82  transport types. The server selects the final transport which it returns in the
83  Transport header field. The server also includes its ports where RTP and RTCP
84  messages can be sent to.
85
86  In the above example UDP was choosen as a transport. At this point the RTSPSrc element
87  will furter configure its elements to process this stream.
88
89  The RTSPSrc will create and connect an RTP session manager element and will
90  connect it to the src pads of the udp element. The data pad from the RTP session
91  manager is ghostpadded to RTPSrc.
92  The RTCP pad of the rtpdec is routed to a new udpsink that sends data to the RTCP
93  port of the server as returned in the Transport: header field.
94
95    +---------------------------------------------+
96    | +------------+                              |
97    | | udpsrc0    |   +--------+                 |
98    | |  port=5000 ----- rtpdec --------------------
99    | +------------+   |        |                 |
100    | +------------+   |        |  +------------+ |
101    | | udpsrc1    ----- RTCP   ---- udpsink    | |
102    | |  port=5001 |   +--------+  |  port=6001 | |
103    | +------------+               +------------+ |
104    +---------------------------------------------+
105
106  The output type of rtpdec is configured as the media type specified in the SDP
107  message.
108
109- All the elements are set to PAUSED/PLAYING and the PLAY RTSP message is
110  sent.
111
112    >> PLAY rtsp://thread:5454/south-rtsp.mp3 RTSP/1.0
113    >> CSeq: 2
114    >> Session: 5d5cb94413288ccd
115    >>
116    << RTSP/1.0 200 OK
117    << CSeq: 2
118    << Date: Wed May 11 13:21:43 2005 GMT
119    << Session: 5d5cb94413288ccd
120    <<
121
122- The udp source elements receive data from that point and the RTP/RTCP messages
123  are processed by the elements.
124
125- In the case of interleaved mode, the SETUP method yields:
126
127    >> SETUP rtsp://thread:5454/south-rtsp.mp3/streamid=0 RTSP/1.0
128    >> CSeq: 1
129    >> Transport: RTP/AVP/UDP;unicast;client_port=5000-5001,RTP/AVP/UDP;multicast,RTP/AVP/TCP
130    >>
131    << RTSP/1.0 200 OK
132    << Transport: RTP/AVP/TCP;interleaved=0-1
133    << CSeq: 1
134    << Date: Wed May 11 13:21:43 2005 GMT
135    << Session: 5d5cb94413288ccd
136    <<
137
138  This means that RTP/RTCP messages will be sent on channel 0/1 respectively and that
139  the data will be received on the same connection as the RTSP connection.
140
141  At this point, we remove the UDP source elements as we don't need them anymore. We
142  set up the rtpsess session manager element though as follows:
143
144    +---------------------------------------------+
145    | +------------+                              |
146    | | _loop()    |   +--------+                 |
147    | |            ----- rtpses --------------------
148    | |            |   |        |                 |
149    | |            |   |        |  +------------+ |
150    | |            ----- RTCP   ---- udpsink    | |
151    | |            |   +--------+  |  port=6001 | |
152    | +------------+               +------------+ |
153    +---------------------------------------------+
154
155  We start an interal task to start reading from the RTSP connection waiting
156  for data. The received data is then pushed to the rtpdec element.
157
158  When reading from the RTSP connection we receive data packets in the
159  following layout (see also RFC2326)
160
161    $<1 byte channel><2 bytes length, big endian><length bytes of data>
162
163
164RTSP server
165-----------
166
167An RTSP server listen on a port (default 554) for client connections. The client
168typically keeps this channel open during the RTSP session to instruct the server
169to pause/play/stop the stream.
170
171The server exposes a stream consisting of one or more media streams using an
172URL. The media streams are typically audio and video.
173
174  ex:
175     rtsp://thread:5454/alien-rtsp.mpeg
176
177     exposes an audio/video stream. The video is mpeg packetized in RTP and
178     the audio is mp3 in RTP.
179
180The streaming server typically uses a different channel to send the media
181data to clients, typically using RTP over UDP. It is also possible to stream
182the data to the client using the initial RTSP TCP session (the interleaved
183mode). This last mode is useful when the client is behind a firewall but
184does not take advantage of the RTP/UDP features.
185
186In both cases, media data is send to the clients in an unmultiplexed format
187packetized as RTP packets.
188
189The streaming server has to negotiate a connection protocol for each of the
190media streams with the client.
191
192Minimal server requirements:
193
194- The server should copy the CSeq header field in a client request to the
195  response so that the client can match the response to the request.
196
197- The server should keep a session for each client after the client issued
198  a SETUP command. The client should use the same session id for all future
199  request to this server.
200
201- the server must support an OPTIONS request send over the RTSP connection.
202
203    >> OPTIONS * RTSP/1.0
204    >> CSeq: 1
205    >>
206    << RTSP/1.0 200 OK
207    << CSeq: 1
208    << Date: Wed May 11 13:21:43 2005 GMT
209    << Session: 5d5cb94413288ccd
210    << Public: DESCRIBE, SETUP, TEARDOWN, PLAY
211    <<
212
213   The OPTIONS request should list all supported methods on the server.
214
215 - The server should support the DESCRIBE method.
216
217    >> DESCRIBE rtsp://thread:5454/south-rtsp.mp3 RTSP/1.0
218    >> Accept: application/sdp
219    >> CSeq: 2
220    >>
221    << RTSP/1.0 200 OK
222    << Content-Length: 84
223    << Content-Type: application/sdp
224    << CSeq: 2
225    << Date: Wed May 11 13:09:37 2005 GMT
226    <<
227    << v=0
228    << o=- 0 0 IN IP4 192.168.1.1
229    << s=No Title
230    << m=audio 0 RTP/AVP 14
231    << a=control:streamid=0
232
233    The client issues a DESCRIBE command for a specific URL that corresponds
234    to an available stream. The client will also send an Accept header to
235    list its supported formats.
236
237    The server answers this request with a reply in one of the client supported
238    formats (application/sdp is the most common). The server typically sends a
239    fixed reply to all clients for each configured stream.
240
241 - The server must support the SETUP command to configure the media streams
242   that were listed in the DESCRIBE command.
243
244    >> SETUP rtsp://thread:5454/south-rtsp.mp3/streamid=0 RTSP/1.0
245    >> CSeq: 3
246    >> Transport: RTP/AVP/UDP;unicast;client_port=5000-5001,RTP/AVP/UDP;multicast,RTP/AVP/TCP
247    >>
248    << RTSP/1.0 200 OK
249    << Transport: RTP/AVP/UDP;unicast;client_port=5000-5001;server_port=6000-6001
250    << CSeq: 3
251    << Date: Wed May 11 13:21:43 2005 GMT
252    << Session: 5d5cb94413288ccd
253
254    The client will send a SETUP command for each of the streams listed in the
255    DESCRIBE reply. For sdp will use a URL as listed in the a=control: property.
256
257    The client will list the supported transports in the Transport: header field.
258    Each transport is separated with a comma (,) and listed in order of preference.
259    The server has to select the first supported transport.
260
261    In the above example 3 transport are listed:
262
263       RTP/AVP/UDP;unicast;client_port=5000-5001
264
265         The client will accept RTP over UDP on the port pair 5000-5001. Port
266	 5000 will accept the RTP packets, 5001 the RTSP packets send by the
267	 server.
268
269       RTP/AVP/UDP;multicast
270
271         The client can join a multicast group for the specific media stream.
272	 The port numbers of the multicast group it will connect to have to
273	 be specified by the server in the reply.
274
275       RTP/AVP/TCP
276
277         the client can accept RTP packets interleaved on the RTSP connection.
278
279    The server selects a supported transport an allocates an RTP port pair to
280    receive RTP and RTSP data from the client. This last step is optional when
281    the server does not accept RTP data.
282
283    The server should allocate a session for the client and should send the
284    sessionId to the client. The client should use this session id for all
285    future requests.
286
287    The server may refuse a client that does not use the same transport method
288    for all media streams.
289
290    The server stores all client port pairs in the server client session along
291    with the transport method.
292
293    ex:
294
295      For an on-demand stream the server could construct a (minimal) RTP GStreamer
296      pipeline for the client as follows (for an mp3 stream):
297
298      +---------+    +-----------+    +------------+    +-------------+
299      | filesrc |    | rtpmp3enc |    | rtpsession |    | udpsink     |
300      |         |    |           |    |            |    |   host=XXX  |
301      |         |    |           |    |            |    |   port=5000 |
302      |        src--sink        src--rtpsink   rtpsrc--sink           |
303      +---------+    +-----------+    |            |    +-------------+
304                                      |            |    +-------------+
305                                      |            |    | udpsink     |
306                                      |            |    |   host=XXX  |
307                                      |            |    |   port=5001 |
308                                      |       rtspsrc--sink           |
309                                      +------------+    +-------------+
310
311      The server would set the above pipeline to PAUSE to make sure no data
312      is sent to the client yet.
313
314      optionally udpsrc elements can be configured to receive client RTP and
315      RTSP messages.
316
317    ex:
318
319      For a live stream the server could construct a (minimal) RTP GStreamer
320      pipeline for the clients as follows (for an mp3 stream):
321
322      +---------+    +--------+    +-----------+    +------------+    +--------------+
323      | source  |    | mp3enc |    | rtpmp3enc |    | rtpsession |    | multiudpsink |
324      |         |    |        |    |           |    |            |    |   host=...   |
325      |         |    |        |    |           |    |            |    |   port=...   |
326      |        src--sink     src--sink        src--rtpsink   rtpsrc--sink            |
327      +---------+    +--------+    +-----------+    |            |    +--------------+
328                                                    |            |    +--------------+
329                                                    |            |    | multiudpsink |
330                                                    |            |    |   host=...   |
331                                                    |            |    |   port=...   |
332                                                    |       rtspsrc--sink            |
333                                                    +------------+    +--------------+
334
335      Media data is streamed to clients by adding the client host and port numbers
336      to the multiudpsinks.
337
338      optionally udpsrc elements can be configured to receive client RTP and
339      RTSP messages.
340
341 - The server must support the PLAY command to start playback of the configured
342   media streams.
343
344    >> PLAY rtsp://thread:5454/south-rtsp.mp3 RTSP/1.0
345    >> CSeq: 2
346    >> Session: 5d5cb94413288ccd
347    >>
348    << RTSP/1.0 200 OK
349    << CSeq: 2
350    << Date: Wed May 11 13:21:43 2005 GMT
351    << Session: 5d5cb94413288ccd
352    <<
353
354    Using the Session: header field, the server finds the pipeline of the session
355    to PLAY and sets the pipeline to PLAYING at which point the client receives
356    the media stream data.
357
358    In case of a live stream, the server adds the port numbers to a multiudpsink
359    element.
360
361 - The server must support the TEARDOWN command to stop playback and free the
362   session of a client.
363
364    >> TEARDOWN rtsp://thread:5454/south-rtsp.mp3 RTSP/1.0
365    >> CSeq: 4
366    >> Session: 5d5cb94413288ccd
367    >>
368    << RTSP/1.0 200 OK
369    << CSeq: 4
370    << Date: Wed May 11 13:21:43 2005 GMT
371    <<
372
373    The server destroys the client pipeline in case of an on-demand stream or
374    removes the client ports from the multiudpsinks. This effectively stops
375    streaming to the client.
376
377
378