mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-09-17 01:20:19 +00:00
22eb34e2fe
Original commit message from CVS: * gst/rtp/README: * gst/rtp/gstrtpilbcdepay.c: * gst/rtp/gstrtpilbcpay.c: * gst/rtp/gstrtpmp4gdepay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpspeexdepay.c: * gst/rtp/gstrtpspeexpay.c: * gst/rtp/gstrtpsv3vdepay.c: * gst/rtp/gstrtptheoradepay.c: * gst/rtp/gstrtptheorapay.c: * gst/rtp/gstrtpvorbisdepay.c: * gst/rtp/gstrtpvorbispay.c: Fix case of encoding-name and key/value pairs to match the document. This is to make interoperation with SDP case-insensitive as required by the relevant RFCs.
217 lines
9 KiB
Text
217 lines
9 KiB
Text
This directory contains some RTP payloaders/depayloaders for different payload
|
|
types. Use one payloader/depayloder pair per payload. If several payloads can be
|
|
payloaded/depayloaded by the same element, make different copies of it, one for
|
|
each payload.
|
|
|
|
The application/x-rtp mime type
|
|
-------------------------------
|
|
|
|
For valid RTP packets encapsulated in GstBuffers, we use the caps with
|
|
mime type application/x-rtp.
|
|
|
|
The following fields can or must (*) be specified in the structure:
|
|
|
|
* media: (String) [ "audio", "video", "application", "data", "control" ]
|
|
Defined in RFC 2327 in the SDP media announcement field.
|
|
Converted to lower case.
|
|
|
|
* payload: (int) [0, 127]
|
|
For audio and video, these will normally be a media payload type as
|
|
defined in the RTP Audio/Video Profile. For dynamicaly allocated
|
|
payload types, this value will be >= 96 and the encoding-name must be
|
|
set.
|
|
|
|
* clock-rate: (int) [0 - MAXINT]
|
|
the RTP clock rate
|
|
|
|
ssrc: (uint) [0 - MAXINT]
|
|
The ssrc value currently in use.
|
|
|
|
clock-base: (uint) [0 - MAXINT]
|
|
The RTP time representing time 0
|
|
|
|
seqnum-base: (uint) [0 - MAXINT]
|
|
The RTP sequence number representing the first rtp packet
|
|
|
|
encoding-name: (String) ANY
|
|
typically second part of the mime type. ex. MP4V-ES. only required if
|
|
payload type >= 96. Converted to upper case.
|
|
|
|
encoding-params: (String) ANY
|
|
extra encoding parameters (as in the SDP a=rtpmap: field). only required
|
|
if different from the default of the encoding-name.
|
|
Converted to lower-case.
|
|
|
|
Optional parameters as key/value pairs, media type specific. The value type
|
|
should be of type G_TYPE_STRING. The key and value are converted to lower-case.
|
|
A parameter with no value is converted to <param>=1.
|
|
|
|
Example:
|
|
|
|
"application/x-rtp",
|
|
"media", G_TYPE_STRING, "audio", -]
|
|
"payload", G_TYPE_INT, 96, ] - required
|
|
"clock-rate", G_TYPE_INT, 8000, -]
|
|
"encoding-name", G_TYPE_STRING, "AMR", -] - required since payload >= 96
|
|
"encoding-params", G_TYPE_STRING, "1", -] - optional param for AMR
|
|
"octet-align", G_TYPE_STRING, "1", -]
|
|
"crc", G_TYPE_STRING, "0", ]
|
|
"robust-sorting", G_TYPE_STRING, "0", ] AMR specific params.
|
|
"interleaving", G_TYPE_STRING, "0", -]
|
|
|
|
Mapping of caps to and from SDP fields:
|
|
|
|
m=<media> <udp port> RTP/AVP <payload> -] media and payload from caps
|
|
a=rtpmap:<payload> <encoding-name>/<clock-rate>[/<encoding-params>]
|
|
-> when <payload> >= 96
|
|
a=fmtp:<payload> <param>=<value>;...
|
|
|
|
For above caps:
|
|
|
|
m=audio <udp port> RTP/AVP 96
|
|
a=rtpmap:96 AMR/8000/1
|
|
a=fmtp:96 octet-align=1;crc=0;robust-sorting=0;interleaving=0
|
|
|
|
in RTSP, the SSRC is also sent.
|
|
|
|
The optional parameters in the SDP fields are case insensitive. In the caps we
|
|
always use the lowercase names so that the SDP -> caps mapping remains
|
|
possible.
|
|
|
|
|
|
usage with UDP
|
|
--------------
|
|
|
|
To correctly and completely use the RTP payloaders on the sender and the
|
|
receiver you need to write an application. It is not possible to write a full
|
|
blown RTP server with a single gst-launch line.
|
|
|
|
That said, it is possible to do something functional with a few gst-launch
|
|
lines. The biggest problem when constructing a correct gst-launch line lies on
|
|
the receiver end.
|
|
|
|
The receiver needs to know about the type of the RTP data along with a set of
|
|
RTP configuration parameters. This information is usually transmitted to the
|
|
client using some sort of session description language (SDP) over some reliable
|
|
channel (HTTP/RTSP/...).
|
|
|
|
All of the required parameters to connect and use the RTP session on the
|
|
server can be found in the caps on the server end. The client receives this
|
|
information in some way (caps are converted to and from SDP, as explained above,
|
|
for example).
|
|
|
|
Some gst-launch lines:
|
|
|
|
gst-launch-0.10 -v videotestsrc ! ffenc_h263p ! rtph263ppay ! udpsink
|
|
|
|
Setting pipeline to PAUSED ...
|
|
/pipeline0/videotestsrc0.src: caps = video/x-raw-yuv, format=(fourcc)I420,
|
|
width=(int)320, height=(int)240, framerate=(fraction)30/1
|
|
Pipeline is PREROLLING ...
|
|
....
|
|
/pipeline0/udpsink0.sink: caps = application/x-rtp, media=(string)video,
|
|
payload=(int)96, clock-rate=(int)90000, encoding-name=(string)H263-1998,
|
|
ssrc=(guint)527842345, clock-base=(guint)1150776941, seqnum-base=(guint)30982
|
|
....
|
|
Pipeline is PREROLLED ...
|
|
Setting pipeline to PLAYING ...
|
|
New clock: GstSystemClock
|
|
|
|
Write down the caps on the udpsink and set them as the caps of the UDP
|
|
receiver:
|
|
|
|
gst-launch-0.10 -v udpsrc caps="application/x-rtp, media=(string)video,
|
|
payload=(int)96, clock-rate=(int)90000, encoding-name=(string)H263-1998,
|
|
ssrc=(guint)527842345, clock-base=(guint)1150776941, seqnum-base=(guint)30982"
|
|
! rtph263pdepay ! ffdec_h263 ! xvimagesink sync=false
|
|
|
|
The receiver now displays an h263 image. Note that the sync parameter on
|
|
xvimagesink needs to be FALSE because we do not have an RTP session manager
|
|
that controls the synchronisation in this pipeline.
|
|
|
|
Stream a quicktime file with mpeg4 video and AAC audio on port 5000 and port
|
|
5002.
|
|
|
|
gst-launch-0.10 -v filesrc location=~/data/sincity.mp4 ! qtdemux name=d ! queue ! rtpmp4vpay ! udpsink port=5000
|
|
d. ! queue ! rtpmp4gpay ! udpsink port=5002
|
|
....
|
|
/pipeline0/udpsink0.sink: caps = application/x-rtp, media=(string)video,
|
|
payload=(int)96, clock-rate=(int)90000, encoding-name=(string)MP4V-ES,
|
|
ssrc=(guint)1162703703, clock-base=(guint)816135835, seqnum-base=(guint)9294,
|
|
profile-level-id=(string)3, config=(string)000001b003000001b50900000100000001200086c5d4c307d314043c1463000001b25876694430303334
|
|
/pipeline0/udpsink1.sink: caps = application/x-rtp, media=(string)audio,
|
|
payload=(int)96, clock-rate=(int)44100, encoding-name=(string)MPEG4-GENERIC,
|
|
ssrc=(guint)3246149898, clock-base=(guint)4134514058, seqnum-base=(guint)57633,
|
|
encoding-params=(string)2, streamtype=(string)5, profile-level-id=(string)1,
|
|
mode=(string)aac-hbr, config=(string)1210, sizelength=(string)13,
|
|
indexlength=(string)3, indexdeltalength=(string)3
|
|
....
|
|
|
|
Again copy the caps on both sinks to the receiver launch line
|
|
|
|
gst-launch
|
|
udpsrc port=5000 caps="application/x-rtp, media=(string)video, payload=(int)96,
|
|
clock-rate=(int)90000, encoding-name=(string)MP4V-ES, ssrc=(guint)1162703703,
|
|
clock-base=(guint)816135835, seqnum-base=(guint)9294, profile-level-id=(string)3,
|
|
config=(string)000001b003000001b50900000100000001200086c5d4c307d314043c1463000001b25876694430303334"
|
|
! rtpmp4vdepay ! ffdec_mpeg4 ! xvimagesink sync=false
|
|
udpsrc port=5002 caps="application/x-rtp, media=(string)audio, payload=(int)96,
|
|
clock-rate=(int)44100, encoding-name=(string)MPEG4-GENERIC, ssrc=(guint)3246149898,
|
|
clock-base=(guint)4134514058, seqnum-base=(guint)57633, encoding-params=(string)2,
|
|
streamtype=(string)5, profile-level-id=(string)1, mode=(string)aac-hbr,
|
|
config=(string)1210, sizelength=(string)13, indexlength=(string)3,
|
|
indexdeltalength=(string)3"
|
|
! rtpmp4gdepay ! faad ! alsasink sync=false
|
|
|
|
The caps on the udpsinks can be retrieved when the server pipeline prerolled to
|
|
PAUSED.
|
|
|
|
The caps on the receiver side can be set on the UDP source elements when the
|
|
pipeline went to PAUSED. In that state no data is received from the UDP sources
|
|
as they are live sources and only produce data in PLAYING.
|
|
|
|
|
|
Relevant RFCs
|
|
-------------
|
|
|
|
3550 RTP: A Transport Protocol for Real-Time Applications. ( 1889 Obsolete )
|
|
|
|
2198 RTP Payload for Redundant Audio Data.
|
|
3119 A More Loss-Tolerant RTP Payload Format for MP3 Audio.
|
|
|
|
2793 RTP Payload for Text Conversation.
|
|
|
|
2032 RTP Payload Format for H.261 Video Streams.
|
|
2190 RTP Payload Format for H.263 Video Streams.
|
|
2250 RTP Payload Format for MPEG1/MPEG2 Video.
|
|
2343 RTP Payload Format for Bundled MPEG.
|
|
2429 RTP Payload Format for the 1998 Version of ITU-T Rec. H.263 Video
|
|
2431 RTP Payload Format for BT.656 Video Encoding.
|
|
2435 RTP Payload Format for JPEG-compressed Video.
|
|
3016 RTP Payload Format for MPEG-4 Audio/Visual Streams.
|
|
3047 RTP Payload Format for ITU-T Recommendation G.722.1.
|
|
3189 RTP Payload Format for DV (IEC 61834) Video.
|
|
3190 RTP Payload Format for 12-bit DAT Audio and 20- and 24-bit Linear Sampled Audio.
|
|
3389 Real-time Transport Protocol (RTP) Payload for Comfort Noise (CN)
|
|
2733 An RTP Payload Format for Generic Forward Error Correction.
|
|
2833 RTP Payload for DTMF Digits, Telephony Tones and Telephony
|
|
Signals.
|
|
2862 RTP Payload Format for Real-Time Pointers.
|
|
3351 RTP Profile for Audio and Video Conferences with Minimal Control. ( 1890 Obsolete )
|
|
3555 MIME Type Registration of RTP Payload Formats.
|
|
|
|
2508 Compressing IP/UDP/RTP Headers for Low-Speed Serial Links.
|
|
1305 Network Time Protocol (Version 3) Specification, Implementation and Analysis.
|
|
3339 Date and Time on the Internet: Timestamps.
|
|
2246 The TLS Protocol Version 1.0
|
|
3546 Transport Layer Security (TLS) Extensions. ( Updates 2246 )
|
|
|
|
do we care?
|
|
-----------
|
|
|
|
2029 RTP Payload Format of Sun's CellB Video Encoding.
|
|
|
|
usefull
|
|
-------
|
|
|
|
http://www.iana.org/assignments/rtp-parameters
|