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 =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= RTP/AVP -] media and payload from caps a=rtpmap: /[/] -> when >= 96 a=fmtp: =;... For above caps: m=audio 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: applied-rate: format: GST_FORMAT_TIME start: * GST_SECOND / stop: if != -1 - + start else -1 time: 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