README
1The RTP libraries
2---------------------
3
4 RTP Buffers
5 -----------
6 The real time protocol as described in RFC 3550 requires the use of special
7 packets containing an additional RTP header of at least 12 bytes. GStreamer
8 provides some helper functions for creating and parsing these RTP headers.
9 The result is a normal #GstBuffer with an additional RTP header.
10
11 RTP buffers are usually created with gst_rtp_buffer_new_allocate() or
12 gst_rtp_buffer_new_allocate_len(). These functions create buffers with a
13 preallocated space of memory. It will also ensure that enough memory
14 is allocated for the RTP header. The first function is used when the payload
15 size is known. gst_rtp_buffer_new_allocate_len() should be used when the size
16 of the whole RTP buffer (RTP header + payload) is known.
17
18 When receiving RTP buffers from a network, gst_rtp_buffer_new_take_data()
19 should be used when the user would like to parse that RTP packet. (TODO Ask
20 Wim what the real purpose of this function is as it seems to simply create a
21 duplicate GstBuffer with the same data as the previous one). The
22 function will create a new RTP buffer with the given data as the whole RTP
23 packet. Alternatively, gst_rtp_buffer_new_copy_data() can be used if the user
24 wishes to make a copy of the data before using it in the new RTP buffer.
25
26 It is now possible to use all the gst_rtp_buffer_get_*() or
27 gst_rtp_buffer_set_*() functions to read or write the different parts of the
28 RTP header such as the payload type, the sequence number or the RTP
29 timestamp. The use can also retrieve a pointer to the actual RTP payload data
30 using the gst_rtp_buffer_get_payload() function.
31
32 RTP Base Payloader Class (GstBaseRTPPayload)
33 --------------------------------------------
34
35 All RTP payloader elements (audio or video) should derive from this class.
36
37 RTP Base Audio Payloader Class (GstBaseRTPAudioPayload)
38 -------------------------------------------------------
39
40 This base class can be tested through it's children classes. Here is an
41 example using the iLBC payloader (frame based).
42
43 For 20ms mode :
44
45 GST_DEBUG="basertpaudiopayload:5" gst-launch-1.0 fakesrc sizetype=2
46 sizemax=114 datarate=1900 ! audio/x-iLBC, mode=20 ! rtpilbcpay
47 max-ptime="40000000" ! fakesink
48
49 For 30ms mode :
50
51 GST_DEBUG="basertpaudiopayload:5" gst-launch-1.0 fakesrc sizetype=2
52 sizemax=150 datarate=1662 ! audio/x-iLBC, mode=30 ! rtpilbcpay
53 max-ptime="60000000" ! fakesink
54
55 Here is an example using the uLaw payloader (sample based).
56
57 GST_DEBUG="basertpaudiopayload:5" gst-launch-1.0 fakesrc sizetype=2
58 sizemax=150 datarate=8000 ! audio/x-mulaw ! rtppcmupay max-ptime="6000000" !
59 fakesink
60
61 RTP Base Depayloader Class (GstBaseRTPDepayload)
62 ------------------------------------------------
63
64 All RTP depayloader elements (audio or video) should derive from this class.
65