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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.
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.
ssrc: (uint) [0 - MAXINT]
The ssrc value currently in use. (default = the SSRC of the first RTP
packet)
clock-base: (uint) [0 - MAXINT]
The RTP time representing time npt-start. (default = rtptime of first RTP
packet).
seqnum-base: (uint) [0 - MAXINT]
The RTP sequence number representing the first rtp packet. When this
parameter is given, all sequence numbers below this seqnum should be
ignored. (default = seqnum of first RTP packet).
npt-start: (uint64) [0 - MAXINT]
The Normal Play Time for clock-base. This is the position in the stream and
is between 0 and the duration of the stream. This value is expressed in
nanoseconds GstClockTime. (default = 0)
npt-stop: (uint64) [0 - MAXINT]
The last position in the stream. This value is expressed in nanoseconds
GstClockTime. (default = -1, stop unknown)
play-speed: (gdouble) [-MIN - MAX]
The intended playback speed of the stream. The client is delivered data at
the adjusted speed. The client should adjust its playback speed with this
value and thus corresponds to the GStreamer rate field in the NEWSEGMENT
event. (default = 1.0)
play-scale: (gdouble) [-MIN - MAX]
The rate already applied to the stream. The client is delivered a stream
that is scaled by this amount. This value is used to adjust position
reporting and corresponds to the GStream applied-rate field in the
NEWSEGMENT event. (default = 1.0)
Optional parameters as key/value pairs, media type specific. The value type
should be of type G_TYPE_STRING. The key is converted to lower-case. The
value is left in its original 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.
Mapping of caps to NEWSEGMENT:
rate: <play-speed>
applied-rate: <play-scale>
format: GST_FORMAT_TIME
start: <clock-base> * GST_SECOND / <clock-rate>
stop: if <ntp-stop> != -1
<npt-stop> - <npt-start> + start
else
-1
time: <npt-start>
Timestamping
------------
RTP in GStreamer uses a combination of the RTP timestamps and GStreamer buffer
timestamps to ensure proper synchronisation at the sender and the receiver end.
In RTP applications, the synchronisation is most complex at the receiver side.
At the sender side, the RTP timestamps are generated in the payloaders based on
GStreamer timestamps. At the receiver, GStreamer timestamps are reconstructed
from the RTP timestamps and the GStreamer timestamps in the jitterbuffer. This
process is explained in more detail below.
= synchronisation at the sender
Individual streams at the sender are synchronised using GStreamer timestamps.
The payloader at the sender will convert the GStreamer timestamp into an RTP
timestamp using the following formula:
RTP = ((RT - RT-base) * clock-rate / GST_SECOND) + RTP-offset
RTP: the RTP timestamp for the stream. This value is truncated to
32 bits.
RT: the GStreamer running time corresponding to the timestamp of the
packet to payload
RT-base: the GStreamer running time of the first packet encoded
clock-rate: the clock-rate of the stream
RTP-offset: a random RTP offset
The RTP timestamp corresponding to RT-base is the clock-base (see caps above).
In addition to setting an RTP timestamp in the RTP packet, the payloader is also
responsible for putting the GStreamer timestamp on the resulting output buffer.
This timestamp is used for further synchronisation at the sender pipeline, such
as for sending out the packet on the network.
Notice that the absolute timing information is lost; if the sender is sending
multiple streams, the RTP timestamps in the packets do not contain enough
information to synchronize them in the receiver. The receiver can however use
the RTP timestamps to reconstruct the timing of the stream as it was created by
the sender according to the sender's clock.
Because the payloaded packet contains both an RTP timestamp and a GStreamer
timestamp, it is possible for an RTP session manager to derive the relation
between the RTP and GST timestamps. This information is used by a session
manager to create SR reports. The NTP time in the report will contain the
running time converted to NTP time and the corresponding RTP timestamp.
Note that at the sender side, the RTP and GStreamer timestamp both increment at
the same rate, the sender rate. This rate depends on the global pipeline clock
of the sender.
Some pipelines to illustrate the process:
gst-launch v4l2src ! ffenc_h263p ! rtph263ppay ! udpsink
v4l2src puts a GStreamer timestamp on the video frames base on the current
running_time. The encoder encodes and passed the timestamp on. The payloader
generates an RTP timestamp using the above formula and puts it in the RTP
packet. It also copies the incomming GStreamer timestamp on the output RTP
packet. udpsink synchronizes on the gstreamer timestamp before pushing out the
packet.
= synchronisation at the receiver
The receiver is responsible for timestamping the received RTP packet with the
running_time of the clock at the time the packet was received. This GStreamer
timestamp reflects the receiver rate and depends on the global pipeline clock of
the receiver. The gstreamer timestamp of the received RTP packet contains a
certain amount of jitter introduced by the network.
The most simple option for the receiver is to depayload the RTP packet and play
it back as soon as possible, this is with the timestamp when it was received
from the network. For the above sender pipeline this would be done with the
following pipeline:
gst-launch udpsrc caps="application/x-rtp, media=(string)video,
clock-rate=(int)90000, encoding-name=(string)H263-1998" ! rtph263pdepay !
ffdec_h263 ! xvimagesink
It is important that the depayloader copies the incomming GStreamer timestamp
directly to the depayloaded output buffer. It should never attempt to perform
any logic with the RTP timestamp, this task is for the jitterbuffer as we will
see next.
The above pipeline does not attempt to deal with reordered packets or network
jitter, which could result in jerky playback in the case of high jitter or
corrupted video in the case of packet loss or reordering. This functionality is
performed by the gstrtpjitterbuffer in GStreamer.
The task of the gstrtpjitterbuffer element is to:
- deal with reordered packets based on the seqnum
- calculate the drift between the sender and receiver clocks using the
GStreamer timestamps (receiver clock rate) and RTP timestamps (sender clock
rate).
To deal with reordered packet, the jitterbuffer holds on to the received RTP
packets in a queue for a configurable amount of time, called the latency.
The jitterbuffer also eliminates network jitter and then tracks the drift
between the local clock (as expressed in the GStreamer timestamps) and the
remote clock (as expressed in the RTP timestamps). It will remove the jitter
and will apply the drift correction to the GStreamer timestamp before pushing
the buffer downstream. The result is that the depayloader receives a smoothed
GStreamer timestamp on the RTP packet, which is copied to the depayloaded data.
The following pipeline illustrates a receiver with a jitterbuffer.
gst-launch udpsrc caps="application/x-rtp, media=(string)video,
clock-rate=(int)90000, encoding-name=(string)H263-1998" !
gstrtpjitterbuffer latency=100 ! rtph263pdepay ! ffdec_h263 ! xvimagesink
The latency property on the jitterbuffer controls the amount of delay (in
milliseconds) to apply to the outgoing packets. A higher latency will produce
smoother playback in networks with high jitter but cause a higher latency.
Choosing a good value for the latency is a tradeoff between the quality and
latency. The better the network, the lower the latency can be set.
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
The receiver now displays an h263 image. Since there is no jitterbuffer in the
pipeline, frames will be displayed at the time when they are received. This can
result in jerky playback in the case of high network jitter or currupted video
when packets are dropped or reordered.
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 above pipeline sets sync=false on the audio and video sink which means that
no synchronisation will be performed in the sinks, they play the data when it
arrives. If you want to enable synchronisation in the sinks it is highly
recommended to use a gstrtpjitterbuffer after the udpsrc elements.
Even when sync is enabled, the two different streams will not play synchronised
against eachother because the receiver does not have enough information to
perform this task. For this you need to add the gstrtpbin element in both the
sender and receiver pipeline and use additional sources and sinks to transmit
RTCP packets used for inter-stream synchronisation.
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
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