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398 lines
17 KiB
Text
398 lines
17 KiB
Text
This directory contains some RTP payloaders/depayloaders for different payload
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types. Use one payloader/depayloder pair per payload. If several payloads can be
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payloaded/depayloaded by the same element, make different copies of it, one for
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each payload.
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The application/x-rtp mime type
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-------------------------------
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For valid RTP packets encapsulated in GstBuffers, we use the caps with
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mime type application/x-rtp.
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The following fields can or must (*) be specified in the structure:
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* media: (String) [ "audio", "video", "application", "data", "control" ]
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Defined in RFC 2327 in the SDP media announcement field.
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Converted to lower case.
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* payload: (int) [0, 127]
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For audio and video, these will normally be a media payload type as
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defined in the RTP Audio/Video Profile. For dynamicaly allocated
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payload types, this value will be >= 96 and the encoding-name must be
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set.
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* clock-rate: (int) [0 - MAXINT]
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The RTP clock rate.
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encoding-name: (String) ANY
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typically second part of the mime type. ex. MP4V-ES. only required if
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payload type >= 96. Converted to upper case.
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encoding-params: (String) ANY
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extra encoding parameters (as in the SDP a=rtpmap: field). only required
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if different from the default of the encoding-name.
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Converted to lower-case.
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ssrc: (uint) [0 - MAXINT]
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The ssrc value currently in use. (default = the SSRC of the first RTP
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packet)
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timestamp-offset: (uint) [0 - MAXINT]
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The RTP time representing time npt-start. (default = rtptime of first RTP
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packet).
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seqnum-offset: (uint) [0 - MAXINT]
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The RTP sequence number representing the first rtp packet. When this
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parameter is given, all sequence numbers below this seqnum should be
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ignored. (default = seqnum of first RTP packet).
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npt-start: (uint64) [0 - MAXINT]
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The Normal Play Time for clock-base. This is the position in the stream and
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is between 0 and the duration of the stream. This value is expressed in
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nanoseconds GstClockTime. (default = 0)
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npt-stop: (uint64) [0 - MAXINT]
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The last position in the stream. This value is expressed in nanoseconds
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GstClockTime. (default = -1, stop unknown)
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play-speed: (gdouble) [-MIN - MAX]
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The intended playback speed of the stream. The client is delivered data at
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the adjusted speed. The client should adjust its playback speed with this
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value and thus corresponds to the GStreamer rate field in the NEWSEGMENT
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event. (default = 1.0)
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play-scale: (gdouble) [-MIN - MAX]
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The rate already applied to the stream. The client is delivered a stream
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that is scaled by this amount. This value is used to adjust position
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reporting and corresponds to the GStream applied-rate field in the
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NEWSEGMENT event. (default = 1.0)
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maxptime: (uint) [0, MAX]
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The maxptime as defined in RFC 4566, this defines the maximum size of a
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packet. It overrides the max-ptime property of payloaders.
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Optional parameters as key/value pairs, media type specific. The value type
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should be of type G_TYPE_STRING. The key is converted to lower-case. The
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value is left in its original case.
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A parameter with no value is converted to <param>=1.
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Example:
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"application/x-rtp",
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"media", G_TYPE_STRING, "audio", -.
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"payload", G_TYPE_INT, 96, | - required
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"clock-rate", G_TYPE_INT, 8000, -'
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"encoding-name", G_TYPE_STRING, "AMR", -. - required since payload >= 96
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"encoding-params", G_TYPE_STRING, "1", -' - optional param for AMR
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"octet-align", G_TYPE_STRING, "1", -.
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"crc", G_TYPE_STRING, "0", |
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"robust-sorting", G_TYPE_STRING, "0", | AMR specific params.
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"interleaving", G_TYPE_STRING, "0", -'
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Mapping of caps to and from SDP fields:
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m=<media> <udp port> RTP/AVP <payload> -] media and payload from caps
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a=rtpmap:<payload> <encoding-name>/<clock-rate>[/<encoding-params>]
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-> when <payload> >= 96
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a=fmtp:<payload> <param>=<value>;...
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For above caps:
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m=audio <udp port> RTP/AVP 96
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a=rtpmap:96 AMR/8000/1
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a=fmtp:96 octet-align=1;crc=0;robust-sorting=0;interleaving=0
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Attributes are converted as follows:
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IANA registered attribute names are prepended with 'a-' before putting them in
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the caps. Unregistered keys (starting with 'x-') are copied directly into the
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caps.
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in RTSP, the SSRC is also sent.
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The optional parameters in the SDP fields are case insensitive. In the caps we
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always use the lowercase names so that the SDP -> caps mapping remains
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possible.
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Mapping of caps to NEWSEGMENT:
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rate: <play-speed>
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applied-rate: <play-scale>
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format: GST_FORMAT_TIME
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start: <clock-base> * GST_SECOND / <clock-rate>
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stop: if <ntp-stop> != -1
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<npt-stop> - <npt-start> + start
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else
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-1
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time: <npt-start>
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Timestamping
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------------
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RTP in GStreamer uses a combination of the RTP timestamps and GStreamer buffer
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timestamps to ensure proper synchronisation at the sender and the receiver end.
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In RTP applications, the synchronisation is most complex at the receiver side.
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At the sender side, the RTP timestamps are generated in the payloaders based on
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GStreamer timestamps. At the receiver, GStreamer timestamps are reconstructed
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from the RTP timestamps and the GStreamer timestamps in the jitterbuffer. This
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process is explained in more detail below.
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= synchronisation at the sender
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Individual streams at the sender are synchronised using GStreamer timestamps.
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The payloader at the sender will convert the GStreamer timestamp into an RTP
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timestamp using the following formula:
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RTP = ((RT - RT-base) * clock-rate / GST_SECOND) + RTP-offset
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RTP: the RTP timestamp for the stream. This value is truncated to
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32 bits.
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RT: the GStreamer running time corresponding to the timestamp of the
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packet to payload
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RT-base: the GStreamer running time of the first packet encoded
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clock-rate: the clock-rate of the stream
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RTP-offset: a random RTP offset
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The RTP timestamp corresponding to RT-base is the clock-base (see caps above).
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In addition to setting an RTP timestamp in the RTP packet, the payloader is also
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responsible for putting the GStreamer timestamp on the resulting output buffer.
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This timestamp is used for further synchronisation at the sender pipeline, such
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as for sending out the packet on the network.
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Notice that the absolute timing information is lost; if the sender is sending
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multiple streams, the RTP timestamps in the packets do not contain enough
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information to synchronize them in the receiver. The receiver can however use
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the RTP timestamps to reconstruct the timing of the stream as it was created by
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the sender according to the sender's clock.
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Because the payloaded packet contains both an RTP timestamp and a GStreamer
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timestamp, it is possible for an RTP session manager to derive the relation
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between the RTP and GST timestamps. This information is used by a session
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manager to create SR reports. The NTP time in the report will contain the
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running time converted to NTP time and the corresponding RTP timestamp.
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Note that at the sender side, the RTP and GStreamer timestamp both increment at
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the same rate, the sender rate. This rate depends on the global pipeline clock
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of the sender.
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Some pipelines to illustrate the process:
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gst-launch-1.0 v4l2src ! ffenc_h263p ! rtph263ppay ! udpsink
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v4l2src puts a GStreamer timestamp on the video frames base on the current
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running_time. The encoder encodes and passed the timestamp on. The payloader
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generates an RTP timestamp using the above formula and puts it in the RTP
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packet. It also copies the incomming GStreamer timestamp on the output RTP
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packet. udpsink synchronizes on the gstreamer timestamp before pushing out the
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packet.
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= synchronisation at the receiver
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The receiver is responsible for timestamping the received RTP packet with the
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running_time of the clock at the time the packet was received. This GStreamer
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timestamp reflects the receiver rate and depends on the global pipeline clock of
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the receiver. The gstreamer timestamp of the received RTP packet contains a
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certain amount of jitter introduced by the network.
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The most simple option for the receiver is to depayload the RTP packet and play
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it back as soon as possible, this is with the timestamp when it was received
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from the network. For the above sender pipeline this would be done with the
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following pipeline:
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gst-launch-1.0 udpsrc caps="application/x-rtp, media=(string)video,
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clock-rate=(int)90000, encoding-name=(string)H263-1998" ! rtph263pdepay !
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ffdec_h263 ! xvimagesink
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It is important that the depayloader copies the incomming GStreamer timestamp
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directly to the depayloaded output buffer. It should never attempt to perform
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any logic with the RTP timestamp, this task is for the jitterbuffer as we will
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see next.
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The above pipeline does not attempt to deal with reordered packets or network
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jitter, which could result in jerky playback in the case of high jitter or
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corrupted video in the case of packet loss or reordering. This functionality is
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performed by the gstrtpjitterbuffer in GStreamer.
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The task of the gstrtpjitterbuffer element is to:
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- deal with reordered packets based on the seqnum
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- calculate the drift between the sender and receiver clocks using the
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GStreamer timestamps (receiver clock rate) and RTP timestamps (sender clock
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rate).
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To deal with reordered packet, the jitterbuffer holds on to the received RTP
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packets in a queue for a configurable amount of time, called the latency.
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The jitterbuffer also eliminates network jitter and then tracks the drift
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between the local clock (as expressed in the GStreamer timestamps) and the
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remote clock (as expressed in the RTP timestamps). It will remove the jitter
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and will apply the drift correction to the GStreamer timestamp before pushing
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the buffer downstream. The result is that the depayloader receives a smoothed
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GStreamer timestamp on the RTP packet, which is copied to the depayloaded data.
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The following pipeline illustrates a receiver with a jitterbuffer.
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gst-launch-1.0 udpsrc caps="application/x-rtp, media=(string)video,
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clock-rate=(int)90000, encoding-name=(string)H263-1998" !
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gstrtpjitterbuffer latency=100 ! rtph263pdepay ! ffdec_h263 ! xvimagesink
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The latency property on the jitterbuffer controls the amount of delay (in
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milliseconds) to apply to the outgoing packets. A higher latency will produce
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smoother playback in networks with high jitter but cause a higher latency.
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Choosing a good value for the latency is a tradeoff between the quality and
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latency. The better the network, the lower the latency can be set.
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usage with UDP
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--------------
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To correctly and completely use the RTP payloaders on the sender and the
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receiver you need to write an application. It is not possible to write a full
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blown RTP server with a single gst-launch-1.0 line.
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That said, it is possible to do something functional with a few gst-launch
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lines. The biggest problem when constructing a correct gst-launch-1.0 line lies on
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the receiver end.
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The receiver needs to know about the type of the RTP data along with a set of
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RTP configuration parameters. This information is usually transmitted to the
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client using some sort of session description language (SDP) over some reliable
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channel (HTTP/RTSP/...).
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All of the required parameters to connect and use the RTP session on the
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server can be found in the caps on the server end. The client receives this
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information in some way (caps are converted to and from SDP, as explained above,
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for example).
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Some gst-launch-1.0 lines:
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gst-launch-0.10 -v videotestsrc ! ffenc_h263p ! rtph263ppay ! udpsink
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Setting pipeline to PAUSED ...
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/pipeline0/videotestsrc0.src: caps = video/x-raw, format=(string)I420,
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width=(int)320, height=(int)240, framerate=(fraction)30/1
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Pipeline is PREROLLING ...
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....
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/pipeline0/udpsink0.sink: caps = application/x-rtp, media=(string)video,
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payload=(int)96, clock-rate=(int)90000, encoding-name=(string)H263-1998,
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ssrc=(guint)527842345, clock-base=(guint)1150776941, seqnum-base=(guint)30982
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....
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Pipeline is PREROLLED ...
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Setting pipeline to PLAYING ...
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New clock: GstSystemClock
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Write down the caps on the udpsink and set them as the caps of the UDP
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receiver:
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gst-launch-0.10 -v udpsrc caps="application/x-rtp, media=(string)video,
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payload=(int)96, clock-rate=(int)90000, encoding-name=(string)H263-1998,
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ssrc=(guint)527842345, clock-base=(guint)1150776941, seqnum-base=(guint)30982"
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! rtph263pdepay ! ffdec_h263 ! xvimagesink
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The receiver now displays an h263 image. Since there is no jitterbuffer in the
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pipeline, frames will be displayed at the time when they are received. This can
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result in jerky playback in the case of high network jitter or currupted video
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when packets are dropped or reordered.
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Stream a quicktime file with mpeg4 video and AAC audio on port 5000 and port
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5002.
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gst-launch-0.10 -v filesrc location=~/data/sincity.mp4 ! qtdemux name=d ! queue ! rtpmp4vpay ! udpsink port=5000
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d. ! queue ! rtpmp4gpay ! udpsink port=5002
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....
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/pipeline0/udpsink0.sink: caps = application/x-rtp, media=(string)video,
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payload=(int)96, clock-rate=(int)90000, encoding-name=(string)MP4V-ES,
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ssrc=(guint)1162703703, clock-base=(guint)816135835, seqnum-base=(guint)9294,
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profile-level-id=(string)3, config=(string)000001b003000001b50900000100000001200086c5d4c307d314043c1463000001b25876694430303334
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/pipeline0/udpsink1.sink: caps = application/x-rtp, media=(string)audio,
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payload=(int)96, clock-rate=(int)44100, encoding-name=(string)MPEG4-GENERIC,
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ssrc=(guint)3246149898, clock-base=(guint)4134514058, seqnum-base=(guint)57633,
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encoding-params=(string)2, streamtype=(string)5, profile-level-id=(string)1,
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mode=(string)aac-hbr, config=(string)1210, sizelength=(string)13,
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indexlength=(string)3, indexdeltalength=(string)3
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....
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Again copy the caps on both sinks to the receiver launch line
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gst-launch-1.0
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udpsrc port=5000 caps="application/x-rtp, media=(string)video, payload=(int)96,
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clock-rate=(int)90000, encoding-name=(string)MP4V-ES, ssrc=(guint)1162703703,
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clock-base=(guint)816135835, seqnum-base=(guint)9294, profile-level-id=(string)3,
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config=(string)000001b003000001b50900000100000001200086c5d4c307d314043c1463000001b25876694430303334"
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! rtpmp4vdepay ! ffdec_mpeg4 ! xvimagesink sync=false
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udpsrc port=5002 caps="application/x-rtp, media=(string)audio, payload=(int)96,
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clock-rate=(int)44100, encoding-name=(string)MPEG4-GENERIC, ssrc=(guint)3246149898,
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clock-base=(guint)4134514058, seqnum-base=(guint)57633, encoding-params=(string)2,
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streamtype=(string)5, profile-level-id=(string)1, mode=(string)aac-hbr,
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config=(string)1210, sizelength=(string)13, indexlength=(string)3,
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indexdeltalength=(string)3"
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! rtpmp4gdepay ! faad ! alsasink sync=false
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The caps on the udpsinks can be retrieved when the server pipeline prerolled to
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PAUSED.
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The above pipeline sets sync=false on the audio and video sink which means that
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no synchronisation will be performed in the sinks, they play the data when it
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arrives. If you want to enable synchronisation in the sinks it is highly
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recommended to use a gstrtpjitterbuffer after the udpsrc elements.
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Even when sync is enabled, the two different streams will not play synchronised
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against eachother because the receiver does not have enough information to
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perform this task. For this you need to add the gstrtpbin element in both the
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sender and receiver pipeline and use additional sources and sinks to transmit
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RTCP packets used for inter-stream synchronisation.
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The caps on the receiver side can be set on the UDP source elements when the
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pipeline went to PAUSED. In that state no data is received from the UDP sources
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as they are live sources and only produce data in PLAYING.
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Relevant RFCs
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-------------
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3550 RTP: A Transport Protocol for Real-Time Applications. ( 1889 Obsolete )
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2198 RTP Payload for Redundant Audio Data.
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3119 A More Loss-Tolerant RTP Payload Format for MP3 Audio.
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2793 RTP Payload for Text Conversation.
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2032 RTP Payload Format for H.261 Video Streams.
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2190 RTP Payload Format for H.263 Video Streams.
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2250 RTP Payload Format for MPEG1/MPEG2 Video.
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2343 RTP Payload Format for Bundled MPEG.
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2429 RTP Payload Format for the 1998 Version of ITU-T Rec. H.263 Video
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2431 RTP Payload Format for BT.656 Video Encoding.
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2435 RTP Payload Format for JPEG-compressed Video.
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3016 RTP Payload Format for MPEG-4 Audio/Visual Streams.
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3047 RTP Payload Format for ITU-T Recommendation G.722.1.
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3189 RTP Payload Format for DV (IEC 61834) Video.
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3190 RTP Payload Format for 12-bit DAT Audio and 20- and 24-bit Linear Sampled Audio.
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3389 Real-time Transport Protocol (RTP) Payload for Comfort Noise (CN)
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2733 An RTP Payload Format for Generic Forward Error Correction.
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2833 RTP Payload for DTMF Digits, Telephony Tones and Telephony
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Signals.
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2862 RTP Payload Format for Real-Time Pointers.
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3351 RTP Profile for Audio and Video Conferences with Minimal Control. ( 1890 Obsolete )
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3555 MIME Type Registration of RTP Payload Formats.
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2508 Compressing IP/UDP/RTP Headers for Low-Speed Serial Links.
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1305 Network Time Protocol (Version 3) Specification, Implementation and Analysis.
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3339 Date and Time on the Internet: Timestamps.
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2246 The TLS Protocol Version 1.0
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3546 Transport Layer Security (TLS) Extensions. ( Updates 2246 )
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do we care?
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-----------
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2029 RTP Payload Format of Sun's CellB Video Encoding.
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usefull
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-------
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http://www.iana.org/assignments/rtp-parameters
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