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. (default = the SSRC of the 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) clock-base: (uint) [0 - MAXINT] The RTP time representing time npt-start. (default = rtptime of first RTP packet). 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) 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). 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 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: 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