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Original commit message from CVS: 2006-02-23 Philippe Kalaf <philippe.kalaf at collabora.co.uk> * rtp/gst/gstrtppcmadepay.c: * rtp/gst/gstrtppcmadepay.h: * rtp/gst/gstgstrtppcmapay.c: * rtp/gst/gstgstrtppcmapay.h: * rtp/gst/gstrtppcmudepay.c: * rtp/gst/gstrtppcmudepay.h: * rtp/gst/gstrtppcmupay.c: * rtp/gst/gstrtppcmupay.h: * rtp/gst/Makefile.am: * rtp/gst/gstrtp.c: * rtp/gst/README: Separated the G711 payloaders/depayloaders into separate elements for mulaw/alaw. Also removed the old g711 payloaders/depayloaders.
137 lines
4.8 KiB
Text
137 lines
4.8 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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* 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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ssrc: (uint) [0 - MAXINT]
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The ssrc value currently in use.
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clock-base: (uint) [0 - MAXINT]
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The RTP time representing time 0
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seqnum-base:
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The RTP sequence number representing the first rtp packet
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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
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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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Optional parameters as key/value pairs, media type specific. The value type
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should be of type G_TYPE_STRING.
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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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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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TODO
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----
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- implement packing up to the MTU.
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- discont events in the case of packet loss
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- figure out the clocking.
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- implement various RFCs dealing with different payload types.
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(as modules?)
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- Throw-out the the caps-nego & other session control things to the
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Application Developer( App ), by turning rtcp work into, signals
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in gstrtpsend & props/args in gstrtprecv.
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The App would then be free to use any sort of session control
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protocal like RTSP.( done )
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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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