1 This directory contains some RTP payloaders/depayloaders for different payload
2 types. Use one payloader/depayloder pair per payload. If several payloads can be
3 payloaded/depayloaded by the same element, make different copies of it, one for
6 The application/x-rtp mime type
7 -------------------------------
9 For valid RTP packets encapsulated in GstBuffers, we use the caps with
10 mime type application/x-rtp.
12 The following fields can or must (*) be specified in the structure:
14 * media: (String) [ "audio", "video", "application", "data", "control" ]
15 Defined in RFC 2327 in the SDP media announcement field.
16 Converted to lower case.
18 * payload: (int) [0, 127]
19 For audio and video, these will normally be a media payload type as
20 defined in the RTP Audio/Video Profile. For dynamically allocated
21 payload types, this value will be >= 96 and the encoding-name must be
24 * clock-rate: (int) [0 - MAXINT]
27 encoding-name: (String) ANY
28 typically second part of the mime type. ex. MP4V-ES. only required if
29 payload type >= 96. Converted to upper case.
31 encoding-params: (String) ANY
32 extra encoding parameters (as in the SDP a=rtpmap: field). only required
33 if different from the default of the encoding-name.
34 Converted to lower-case.
36 ssrc: (uint) [0 - MAXINT]
37 The ssrc value currently in use. (default = the SSRC of the first RTP
40 timestamp-offset: (uint) [0 - MAXINT]
41 The RTP time representing time npt-start. (default = rtptime of first RTP
44 seqnum-offset: (uint) [0 - MAXINT]
45 The RTP sequence number representing the first rtp packet. When this
46 parameter is given, all sequence numbers below this seqnum should be
47 ignored. (default = seqnum of first RTP packet).
49 npt-start: (uint64) [0 - MAXINT]
50 The Normal Play Time for clock-base. This is the position in the stream and
51 is between 0 and the duration of the stream. This value is expressed in
52 nanoseconds GstClockTime. (default = 0)
54 npt-stop: (uint64) [0 - MAXINT]
55 The last position in the stream. This value is expressed in nanoseconds
56 GstClockTime. (default = -1, stop unknown)
58 play-speed: (gdouble) [-MIN - MAX]
59 The intended playback speed of the stream. The client is delivered data at
60 the adjusted speed. The client should adjust its playback speed with this
61 value and thus corresponds to the GStreamer rate field in the NEWSEGMENT
62 event. (default = 1.0)
64 play-scale: (gdouble) [-MIN - MAX]
65 The rate already applied to the stream. The client is delivered a stream
66 that is scaled by this amount. This value is used to adjust position
67 reporting and corresponds to the GStream applied-rate field in the
68 NEWSEGMENT event. (default = 1.0)
70 maxptime: (uint) [0, MAX]
71 The maxptime as defined in RFC 4566, this defines the maximum size of a
72 packet. It overrides the max-ptime property of payloaders.
74 Optional parameters as key/value pairs, media type specific. The value type
75 should be of type G_TYPE_STRING. The key is converted to lower-case. The
76 value is left in its original case.
77 A parameter with no value is converted to <param>=1.
82 "media", G_TYPE_STRING, "audio", -.
83 "payload", G_TYPE_INT, 96, | - required
84 "clock-rate", G_TYPE_INT, 8000, -'
85 "encoding-name", G_TYPE_STRING, "AMR", -. - required since payload >= 96
86 "encoding-params", G_TYPE_STRING, "1", -' - optional param for AMR
87 "octet-align", G_TYPE_STRING, "1", -.
88 "crc", G_TYPE_STRING, "0", |
89 "robust-sorting", G_TYPE_STRING, "0", | AMR specific params.
90 "interleaving", G_TYPE_STRING, "0", -'
92 Mapping of caps to and from SDP fields:
94 m=<media> <udp port> RTP/AVP <payload> -] media and payload from caps
95 a=rtpmap:<payload> <encoding-name>/<clock-rate>[/<encoding-params>]
96 -> when <payload> >= 96
97 a=fmtp:<payload> <param>=<value>;...
101 m=audio <udp port> RTP/AVP 96
102 a=rtpmap:96 AMR/8000/1
103 a=fmtp:96 octet-align=1;crc=0;robust-sorting=0;interleaving=0
105 Attributes are converted as follows:
107 IANA registered attribute names are prepended with 'a-' before putting them in
108 the caps. Unregistered keys (starting with 'x-') are copied directly into the
111 in RTSP, the SSRC is also sent.
113 The optional parameters in the SDP fields are case insensitive. In the caps we
114 always use the lowercase names so that the SDP -> caps mapping remains
117 Mapping of caps to NEWSEGMENT:
120 applied-rate: <play-scale>
121 format: GST_FORMAT_TIME
122 start: <clock-base> * GST_SECOND / <clock-rate>
123 stop: if <ntp-stop> != -1
124 <npt-stop> - <npt-start> + start
133 RTP in GStreamer uses a combination of the RTP timestamps and GStreamer buffer
134 timestamps to ensure proper synchronisation at the sender and the receiver end.
136 In RTP applications, the synchronisation is most complex at the receiver side.
138 At the sender side, the RTP timestamps are generated in the payloaders based on
139 GStreamer timestamps. At the receiver, GStreamer timestamps are reconstructed
140 from the RTP timestamps and the GStreamer timestamps in the jitterbuffer. This
141 process is explained in more detail below.
143 = synchronisation at the sender
145 Individual streams at the sender are synchronised using GStreamer timestamps.
146 The payloader at the sender will convert the GStreamer timestamp into an RTP
147 timestamp using the following formula:
149 RTP = ((RT - RT-base) * clock-rate / GST_SECOND) + RTP-offset
151 RTP: the RTP timestamp for the stream. This value is truncated to
153 RT: the GStreamer running time corresponding to the timestamp of the
155 RT-base: the GStreamer running time of the first packet encoded
156 clock-rate: the clock-rate of the stream
157 RTP-offset: a random RTP offset
159 The RTP timestamp corresponding to RT-base is the clock-base (see caps above).
161 In addition to setting an RTP timestamp in the RTP packet, the payloader is also
162 responsible for putting the GStreamer timestamp on the resulting output buffer.
163 This timestamp is used for further synchronisation at the sender pipeline, such
164 as for sending out the packet on the network.
166 Notice that the absolute timing information is lost; if the sender is sending
167 multiple streams, the RTP timestamps in the packets do not contain enough
168 information to synchronize them in the receiver. The receiver can however use
169 the RTP timestamps to reconstruct the timing of the stream as it was created by
170 the sender according to the sender's clock.
172 Because the payloaded packet contains both an RTP timestamp and a GStreamer
173 timestamp, it is possible for an RTP session manager to derive the relation
174 between the RTP and GST timestamps. This information is used by a session
175 manager to create SR reports. The NTP time in the report will contain the
176 running time converted to NTP time and the corresponding RTP timestamp.
178 Note that at the sender side, the RTP and GStreamer timestamp both increment at
179 the same rate, the sender rate. This rate depends on the global pipeline clock
182 Some pipelines to illustrate the process:
184 gst-launch-1.0 v4l2src ! videoconvert ! avenc_h263p ! rtph263ppay ! udpsink
186 v4l2src puts a GStreamer timestamp on the video frames base on the current
187 running_time. The encoder encodes and passed the timestamp on. The payloader
188 generates an RTP timestamp using the above formula and puts it in the RTP
189 packet. It also copies the incoming GStreamer timestamp on the output RTP
190 packet. udpsink synchronizes on the gstreamer timestamp before pushing out the
194 = synchronisation at the receiver
196 The receiver is responsible for timestamping the received RTP packet with the
197 running_time of the clock at the time the packet was received. This GStreamer
198 timestamp reflects the receiver rate and depends on the global pipeline clock of
199 the receiver. The gstreamer timestamp of the received RTP packet contains a
200 certain amount of jitter introduced by the network.
202 The most simple option for the receiver is to depayload the RTP packet and play
203 it back as soon as possible, this is with the timestamp when it was received
204 from the network. For the above sender pipeline this would be done with the
207 gst-launch-1.0 udpsrc caps="application/x-rtp, media=(string)video,
208 clock-rate=(int)90000, encoding-name=(string)H263-1998" ! rtph263pdepay !
209 avdec_h263 ! autovideosink
211 It is important that the depayloader copies the incoming GStreamer timestamp
212 directly to the depayloaded output buffer. It should never attempt to perform
213 any logic with the RTP timestamp, this task is for the jitterbuffer as we will
216 The above pipeline does not attempt to deal with reordered packets or network
217 jitter, which could result in jerky playback in the case of high jitter or
218 corrupted video in the case of packet loss or reordering. This functionality is
219 performed by the gstrtpjitterbuffer in GStreamer.
221 The task of the gstrtpjitterbuffer element is to:
223 - deal with reordered packets based on the seqnum
224 - calculate the drift between the sender and receiver clocks using the
225 GStreamer timestamps (receiver clock rate) and RTP timestamps (sender clock
228 To deal with reordered packet, the jitterbuffer holds on to the received RTP
229 packets in a queue for a configurable amount of time, called the latency.
231 The jitterbuffer also eliminates network jitter and then tracks the drift
232 between the local clock (as expressed in the GStreamer timestamps) and the
233 remote clock (as expressed in the RTP timestamps). It will remove the jitter
234 and will apply the drift correction to the GStreamer timestamp before pushing
235 the buffer downstream. The result is that the depayloader receives a smoothed
236 GStreamer timestamp on the RTP packet, which is copied to the depayloaded data.
238 The following pipeline illustrates a receiver with a jitterbuffer.
240 gst-launch-1.0 udpsrc caps="application/x-rtp, media=(string)video,
241 clock-rate=(int)90000, encoding-name=(string)H263-1998" !
242 rtpjitterbuffer latency=100 ! rtph263pdepay ! avdec_h263 ! autovideosink
244 The latency property on the jitterbuffer controls the amount of delay (in
245 milliseconds) to apply to the outgoing packets. A higher latency will produce
246 smoother playback in networks with high jitter but cause a higher latency.
247 Choosing a good value for the latency is a tradeoff between the quality and
248 latency. The better the network, the lower the latency can be set.
254 To correctly and completely use the RTP payloaders on the sender and the
255 receiver you need to write an application. It is not possible to write a full
256 blown RTP server with a single gst-launch-1.0 line.
258 That said, it is possible to do something functional with a few gst-launch
259 lines. The biggest problem when constructing a correct gst-launch-1.0 line lies on
262 The receiver needs to know about the type of the RTP data along with a set of
263 RTP configuration parameters. This information is usually transmitted to the
264 client using some sort of session description language (SDP) over some reliable
265 channel (HTTP/RTSP/...).
267 All of the required parameters to connect and use the RTP session on the
268 server can be found in the caps on the server end. The client receives this
269 information in some way (caps are converted to and from SDP, as explained above,
272 Some gst-launch-1.0 lines:
274 gst-launch-1.0 -v videotestsrc ! videoconvert ! avenc_h263p ! rtph263ppay ! udpsink
276 Setting pipeline to PAUSED ...
277 /pipeline0/videotestsrc0.src: caps = video/x-raw, format=(string)I420,
278 width=(int)320, height=(int)240, framerate=(fraction)30/1
279 Pipeline is PREROLLING ...
281 /pipeline0/udpsink0.sink: caps = application/x-rtp, media=(string)video,
282 payload=(int)96, clock-rate=(int)90000, encoding-name=(string)H263-1998,
283 ssrc=(guint)527842345, clock-base=(guint)1150776941, seqnum-base=(guint)30982
285 Pipeline is PREROLLED ...
286 Setting pipeline to PLAYING ...
287 New clock: GstSystemClock
289 Write down the caps on the udpsink and set them as the caps of the UDP
292 gst-launch-1.0 -v udpsrc caps="application/x-rtp, media=(string)video,
293 payload=(int)96, clock-rate=(int)90000, encoding-name=(string)H263-1998,
294 ssrc=(guint)527842345, clock-base=(guint)1150776941, seqnum-base=(guint)30982"
295 ! rtph263pdepay ! avdec_h263 ! autovideosink
297 The receiver now displays an h263 image. Since there is no jitterbuffer in the
298 pipeline, frames will be displayed at the time when they are received. This can
299 result in jerky playback in the case of high network jitter or corrupted video
300 when packets are dropped or reordered.
302 Stream a quicktime file with mpeg4 video and AAC audio on port 5000 and port
305 gst-launch-1.0 -v filesrc location=~/data/sincity.mp4 ! qtdemux name=d ! queue ! rtpmp4vpay ! udpsink port=5000
306 d. ! queue ! rtpmp4gpay ! udpsink port=5002
308 /pipeline0/udpsink0.sink: caps = application/x-rtp, media=(string)video,
309 payload=(int)96, clock-rate=(int)90000, encoding-name=(string)MP4V-ES,
310 ssrc=(guint)1162703703, clock-base=(guint)816135835, seqnum-base=(guint)9294,
311 profile-level-id=(string)3, config=(string)000001b003000001b50900000100000001200086c5d4c307d314043c1463000001b25876694430303334
312 /pipeline0/udpsink1.sink: caps = application/x-rtp, media=(string)audio,
313 payload=(int)96, clock-rate=(int)44100, encoding-name=(string)MPEG4-GENERIC,
314 ssrc=(guint)3246149898, clock-base=(guint)4134514058, seqnum-base=(guint)57633,
315 encoding-params=(string)2, streamtype=(string)5, profile-level-id=(string)1,
316 mode=(string)aac-hbr, config=(string)1210, sizelength=(string)13,
317 indexlength=(string)3, indexdeltalength=(string)3
320 Again copy the caps on both sinks to the receiver launch line
323 udpsrc port=5000 caps="application/x-rtp, media=(string)video, payload=(int)96,
324 clock-rate=(int)90000, encoding-name=(string)MP4V-ES, ssrc=(guint)1162703703,
325 clock-base=(guint)816135835, seqnum-base=(guint)9294, profile-level-id=(string)3,
326 config=(string)000001b003000001b50900000100000001200086c5d4c307d314043c1463000001b25876694430303334"
327 ! rtpmp4vdepay ! ffdec_mpeg4 ! autovideosink sync=false
328 udpsrc port=5002 caps="application/x-rtp, media=(string)audio, payload=(int)96,
329 clock-rate=(int)44100, encoding-name=(string)MPEG4-GENERIC, ssrc=(guint)3246149898,
330 clock-base=(guint)4134514058, seqnum-base=(guint)57633, encoding-params=(string)2,
331 streamtype=(string)5, profile-level-id=(string)1, mode=(string)aac-hbr,
332 config=(string)1210, sizelength=(string)13, indexlength=(string)3,
333 indexdeltalength=(string)3"
334 ! rtpmp4gdepay ! faad ! alsasink sync=false
336 The caps on the udpsinks can be retrieved when the server pipeline prerolled to
339 The above pipeline sets sync=false on the audio and video sink which means that
340 no synchronisation will be performed in the sinks, they play the data when it
341 arrives. If you want to enable synchronisation in the sinks it is highly
342 recommended to use a gstrtpjitterbuffer after the udpsrc elements.
344 Even when sync is enabled, the two different streams will not play synchronised
345 against each other because the receiver does not have enough information to
346 perform this task. For this you need to add the rtpbin element in both the
347 sender and receiver pipeline and use additional sources and sinks to transmit
348 RTCP packets used for inter-stream synchronisation.
350 The caps on the receiver side can be set on the UDP source elements when the
351 pipeline went to PAUSED. In that state no data is received from the UDP sources
352 as they are live sources and only produce data in PLAYING.
358 3550 RTP: A Transport Protocol for Real-Time Applications. ( 1889 Obsolete )
360 2198 RTP Payload for Redundant Audio Data.
361 3119 A More Loss-Tolerant RTP Payload Format for MP3 Audio.
363 2793 RTP Payload for Text Conversation.
365 2032 RTP Payload Format for H.261 Video Streams.
366 2190 RTP Payload Format for H.263 Video Streams.
367 2250 RTP Payload Format for MPEG1/MPEG2 Video.
368 2343 RTP Payload Format for Bundled MPEG.
369 2429 RTP Payload Format for the 1998 Version of ITU-T Rec. H.263 Video
370 2431 RTP Payload Format for BT.656 Video Encoding.
371 2435 RTP Payload Format for JPEG-compressed Video.
372 3016 RTP Payload Format for MPEG-4 Audio/Visual Streams.
373 3047 RTP Payload Format for ITU-T Recommendation G.722.1.
374 3189 RTP Payload Format for DV (IEC 61834) Video.
375 3190 RTP Payload Format for 12-bit DAT Audio and 20- and 24-bit Linear Sampled Audio.
376 3389 Real-time Transport Protocol (RTP) Payload for Comfort Noise (CN)
377 2733 An RTP Payload Format for Generic Forward Error Correction.
378 2833 RTP Payload for DTMF Digits, Telephony Tones and Telephony
380 2862 RTP Payload Format for Real-Time Pointers.
381 3351 RTP Profile for Audio and Video Conferences with Minimal Control. ( 1890 Obsolete )
382 3555 MIME Type Registration of RTP Payload Formats.
384 2508 Compressing IP/UDP/RTP Headers for Low-Speed Serial Links.
385 1305 Network Time Protocol (Version 3) Specification, Implementation and Analysis.
386 3339 Date and Time on the Internet: Timestamps.
387 2246 The TLS Protocol Version 1.0
388 3546 Transport Layer Security (TLS) Extensions. ( Updates 2246 )
393 2029 RTP Payload Format of Sun's CellB Video Encoding.
398 http://www.iana.org/assignments/rtp-parameters