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rtp.h
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1 
13 #ifndef _JANUS_RTP_H
14 #define _JANUS_RTP_H
15 
16 #include <arpa/inet.h>
17 #ifdef __MACH__
18 #include <machine/endian.h>
19 #define __BYTE_ORDER BYTE_ORDER
20 #define __BIG_ENDIAN BIG_ENDIAN
21 #define __LITTLE_ENDIAN LITTLE_ENDIAN
22 #else
23 #include <endian.h>
24 #endif
25 #include <inttypes.h>
26 #include <string.h>
27 #include <glib.h>
28 #include <jansson.h>
29 
30 #define RTP_HEADER_SIZE 12
31 
33 typedef struct rtp_header
34 {
35 #if __BYTE_ORDER == __BIG_ENDIAN
36  uint16_t version:2;
37  uint16_t padding:1;
38  uint16_t extension:1;
39  uint16_t csrccount:4;
40  uint16_t markerbit:1;
41  uint16_t type:7;
42 #elif __BYTE_ORDER == __LITTLE_ENDIAN
43  uint16_t csrccount:4;
44  uint16_t extension:1;
45  uint16_t padding:1;
46  uint16_t version:2;
47  uint16_t type:7;
48  uint16_t markerbit:1;
49 #endif
50  uint16_t seq_number;
51  uint32_t timestamp;
52  uint32_t ssrc;
53  uint32_t csrc[16];
54 } rtp_header;
56 
58 typedef struct janus_rtp_packet {
59  char *data;
60  gint length;
61  gint64 created;
64 
67  uint16_t type;
68  uint16_t length;
70 
72 #define JANUS_RTP_EXTMAP_AUDIO_LEVEL "urn:ietf:params:rtp-hdrext:ssrc-audio-level"
73 
74 #define JANUS_RTP_EXTMAP_TOFFSET "urn:ietf:params:rtp-hdrext:toffset"
75 
76 #define JANUS_RTP_EXTMAP_ABS_SEND_TIME "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time"
77 
78 #define JANUS_RTP_EXTMAP_VIDEO_ORIENTATION "urn:3gpp:video-orientation"
79 
80 #define JANUS_RTP_EXTMAP_TRANSPORT_WIDE_CC "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01"
81 
82 #define JANUS_RTP_EXTMAP_PLAYOUT_DELAY "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay"
83 
84 #define JANUS_RTP_EXTMAP_MID "urn:ietf:params:rtp-hdrext:sdes:mid"
85 
86 #define JANUS_RTP_EXTMAP_RID "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id"
87 
88 #define JANUS_RTP_EXTMAP_REPAIRED_RID "urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id"
89 
90 #define JANUS_RTP_EXTMAP_FRAME_MARKING "http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07"
91 
92 #define JANUS_RTP_EXTMAP_ENCRYPTED "urn:ietf:params:rtp-hdrext:encrypt"
93 
94 
95 typedef enum janus_audiocodec {
104 const char *janus_audiocodec_name(janus_audiocodec acodec);
107 
108 typedef enum janus_videocodec {
114 const char *janus_videocodec_name(janus_videocodec vcodec);
117 
118 
122 gboolean janus_is_rtp(char *buf, guint len);
123 
129 char *janus_rtp_payload(char *buf, int len, int *plen);
130 
135 int janus_rtp_header_extension_get_id(const char *sdp, const char *extension);
136 
142 const char *janus_rtp_header_extension_get_from_id(const char *sdp, int id);
143 
150 int janus_rtp_header_extension_parse_audio_level(char *buf, int len, int id, int *level);
151 
161 int janus_rtp_header_extension_parse_video_orientation(char *buf, int len, int id,
162  gboolean *c, gboolean *f, gboolean *r1, gboolean *r0);
163 
171 int janus_rtp_header_extension_parse_playout_delay(char *buf, int len, int id,
172  uint16_t *min_delay, uint16_t *max_delay);
173 
181 int janus_rtp_header_extension_parse_mid(char *buf, int len, int id,
182  char *sdes_item, int sdes_len);
183 
191 int janus_rtp_header_extension_parse_rid(char *buf, int len, int id,
192  char *sdes_item, int sdes_len);
193 
202 int janus_rtp_header_extension_parse_framemarking(char *buf, int len, int id, janus_videocodec codec, uint8_t *tid);
203 
210 int janus_rtp_header_extension_parse_transport_wide_cc(char *buf, int len, int id,
211  uint16_t *transSeqNum);
212 
220 int janus_rtp_header_extension_replace_id(char *buf, int len, int id, int new_id);
221 
224  uint32_t a_last_ssrc, a_last_ts, a_base_ts, a_base_ts_prev, a_prev_ts, a_target_ts, a_start_ts,
225  v_last_ssrc, v_last_ts, v_base_ts, v_base_ts_prev, v_prev_ts, v_target_ts, v_start_ts;
226  uint16_t a_last_seq, a_prev_seq, a_base_seq, a_base_seq_prev,
227  v_last_seq, v_prev_seq, v_base_seq, v_base_seq_prev;
228  gboolean a_seq_reset, a_new_ssrc,
229  v_seq_reset, v_new_ssrc;
230  gint16 a_seq_offset,
231  v_seq_offset;
232  gint32 a_prev_delay, a_active_delay, a_ts_offset,
233  v_prev_delay, v_active_delay, v_ts_offset;
234  gint64 a_last_time, a_reference_time, a_start_time, a_evaluating_start_time,
235  v_last_time, v_reference_time, v_start_time, v_evaluating_start_time;
237 
241 
247 void janus_rtp_header_update(janus_rtp_header *header, janus_rtp_switching_context *context, gboolean video, int step);
248 
249 #define RTP_AUDIO_SKEW_TH_MS 120
250 #define RTP_VIDEO_SKEW_TH_MS 120
251 #define SKEW_DETECTION_WAIT_TIME_SECS 10
252 
265 
266 
282  gint64 last_relayed;
288  gboolean need_pli;
290 
294 
302 void janus_rtp_simulcasting_prepare(json_t *simulcast, int *rid_ext_id, int *framemarking_ext_id, uint32_t *ssrcs, char **rids);
303 
317  char *buf, int len, uint32_t *ssrcs, char **rids,
319 
320 #endif
gboolean changed_temporal
Whether the temporal layer has changed after processing a packet.
Definition: rtp.h:286
Definition: rtp.h:109
int templayer_target
As above, but to handle transitions (e.g., wait for keyframe)
Definition: rtp.h:280
gint64 v_start_time
Definition: rtp.h:234
uint32_t timestamp
Definition: rtp.h:51
struct json_t json_t
Definition: plugin.h:225
void janus_rtp_simulcasting_prepare(json_t *simulcast, int *rid_ext_id, int *framemarking_ext_id, uint32_t *ssrcs, char **rids)
Helper method to prepare the simulcasting info (rids and/or SSRCs) from the simulcast object the core...
Definition: rtp.c:891
gint64 created
Definition: rtp.h:61
int janus_rtp_header_extension_parse_mid(char *buf, int len, int id, char *sdes_item, int sdes_len)
Helper to parse a sdes-mid RTP extension (https://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-ne...
Definition: rtp.c:228
Definition: rtp.h:100
gint64 last_relayed
When we relayed the last packet (used to detect when substreams become unavailable) ...
Definition: rtp.h:282
gint rid_ext_id
RTP Stream extension ID, if any.
Definition: rtp.h:270
Definition: rtp.h:102
int janus_rtp_header_extension_replace_id(char *buf, int len, int id, int new_id)
Helper to replace the ID of an RTP extension with a different one (e.g., to turn a repaired-rtp-strea...
Definition: rtp.c:312
int janus_videocodec_pt(janus_videocodec vcodec)
Definition: rtp.c:865
uint16_t markerbit
Definition: rtp.h:40
int templayer
Which simulcast temporal layer we should forward back.
Definition: rtp.h:278
Definition: rtp.h:101
int janus_rtp_skew_compensate_video(janus_rtp_header *header, janus_rtp_switching_context *context, gint64 now)
Use the context info to compensate for video source skew, if needed.
Definition: rtp.c:479
gboolean changed_substream
Whether the substream has changed after processing a packet.
Definition: rtp.h:284
Definition: rtp.h:110
int janus_rtp_header_extension_parse_video_orientation(char *buf, int len, int id, gboolean *c, gboolean *f, gboolean *r1, gboolean *r0)
Helper to parse a video-orientation RTP extension (http://www.3gpp.org/ftp/Specs/html-info/26114.htm)
Definition: rtp.c:190
janus_videocodec
Definition: rtp.h:108
struct janus_rtp_header_extension janus_rtp_header_extension
RTP extension.
int janus_rtp_header_extension_parse_rid(char *buf, int len, int id, char *sdes_item, int sdes_len)
Helper to parse a rtp-stream-id RTP extension (https://tools.ietf.org/html/draft-ietf-avtext-rid-09) ...
Definition: rtp.c:249
char * janus_rtp_payload(char *buf, int len, int *plen)
Helper to quickly access the RTP payload, skipping header and extensions.
Definition: rtp.c:26
uint32_t ssrc
Definition: rtp.h:52
uint16_t extension
Definition: rtp.h:38
uint16_t seq_number
Definition: rtp.h:50
const char * janus_videocodec_name(janus_videocodec vcodec)
Definition: rtp.c:838
Definition: rtp.h:96
gboolean janus_is_rtp(char *buf, guint len)
Helper method to demultiplex RTP from other protocols.
Definition: rtp.c:19
struct rtp_header rtp_header
RTP Header (http://tools.ietf.org/html/rfc3550#section-5.1)
struct janus_rtp_simulcasting_context janus_rtp_simulcasting_context
Helper struct for processing and tracking simulcast streams.
int janus_rtp_header_extension_parse_transport_wide_cc(char *buf, int len, int id, uint16_t *transSeqNum)
Helper to parse a transport wide sequence number (https://tools.ietf.org/html/draft-holmer-rmcat-tran...
Definition: rtp.c:291
uint16_t csrccount
Definition: rtp.h:39
janus_audiocodec
Definition: rtp.h:95
RTP packet.
Definition: rtp.h:58
void janus_rtp_header_update(janus_rtp_header *header, janus_rtp_switching_context *context, gboolean video, int step)
Use the context info to update the RTP header of a packet, if needed.
Definition: rtp.c:593
int janus_rtp_skew_compensate_audio(janus_rtp_header *header, janus_rtp_switching_context *context, gint64 now)
Use the context info to compensate for audio source skew, if needed.
Definition: rtp.c:363
janus_videocodec janus_videocodec_from_name(const char *name)
Definition: rtp.c:853
uint32_t csrc[16]
Definition: rtp.h:53
RTP context, in order to make sure SSRC changes result in coherent seq/ts increases.
Definition: rtp.h:223
Definition: rtp.h:98
gboolean need_pli
Whether we need to send the user a keyframe request (PLI)
Definition: rtp.h:288
int substream
Which simulcast substream we should forward back.
Definition: rtp.h:274
uint16_t padding
Definition: rtp.h:37
gint32 v_ts_offset
Definition: rtp.h:232
gint64 last_retransmit
Definition: rtp.h:62
gint framemarking_ext_id
Frame marking extension ID, if any.
Definition: rtp.h:272
RTP extension.
Definition: rtp.h:66
Definition: rtp.h:97
Definition: rtp.h:111
struct janus_rtp_packet janus_rtp_packet
RTP packet.
janus_audiocodec janus_audiocodec_from_name(const char *name)
Definition: rtp.c:798
int substream_target
As above, but to handle transitions (e.g., wait for keyframe, or get this if available) ...
Definition: rtp.h:276
struct janus_rtp_switching_context janus_rtp_switching_context
RTP context, in order to make sure SSRC changes result in coherent seq/ts increases.
uint16_t version
Definition: rtp.h:36
RTP Header (http://tools.ietf.org/html/rfc3550#section-5.1)
Definition: rtp.h:33
gboolean v_seq_reset
Definition: rtp.h:228
uint32_t v_target_ts
Definition: rtp.h:224
uint16_t length
Definition: rtp.h:68
int janus_audiocodec_pt(janus_audiocodec acodec)
Definition: rtp.c:816
gint16 v_seq_offset
Definition: rtp.h:230
rtp_header janus_rtp_header
Definition: rtp.h:55
Definition: rtp.h:112
char * data
Definition: rtp.h:59
const char * janus_rtp_header_extension_get_from_id(const char *sdp, int id)
Ugly and dirty helper to quickly get the RTP extension namespace associated with an id (extmap) in an...
Definition: rtp.c:80
uint16_t v_prev_seq
Definition: rtp.h:226
int janus_rtp_header_extension_get_id(const char *sdp, const char *extension)
Ugly and dirty helper to quickly get the id associated with an RTP extension (extmap) in an SDP...
Definition: rtp.c:52
Helper struct for processing and tracking simulcast streams.
Definition: rtp.h:268
void janus_rtp_switching_context_reset(janus_rtp_switching_context *context)
Set (or reset) the context fields to their default values.
Definition: rtp.c:356
uint16_t type
Definition: rtp.h:67
int janus_rtp_header_extension_parse_audio_level(char *buf, int len, int id, int *level)
Helper to parse a ssrc-audio-level RTP extension (https://tools.ietf.org/html/rfc6464) ...
Definition: rtp.c:177
gint length
Definition: rtp.h:60
gboolean janus_rtp_simulcasting_context_process_rtp(janus_rtp_simulcasting_context *context, char *buf, int len, uint32_t *ssrcs, char **rids, janus_videocodec vcodec, janus_rtp_switching_context *sc)
Process an RTP packet, and decide whether this should be relayed or not, updating the context accordi...
Definition: rtp.c:925
Definition: rtp.h:99
const char * janus_audiocodec_name(janus_audiocodec acodec)
Definition: rtp.c:777
int janus_rtp_header_extension_parse_playout_delay(char *buf, int len, int id, uint16_t *min_delay, uint16_t *max_delay)
Helper to parse a playout-delay RTP extension (https://webrtc.org/experiments/rtp-hdrext/playout-dela...
Definition: rtp.c:212
uint16_t type
Definition: rtp.h:41
void janus_rtp_simulcasting_context_reset(janus_rtp_simulcasting_context *context)
Set (or reset) the context fields to their default values.
Definition: rtp.c:881
int janus_rtp_header_extension_parse_framemarking(char *buf, int len, int id, janus_videocodec codec, uint8_t *tid)
Helper to parse a frame-marking RTP extension (http://tools.ietf.org/html/draft-ietf-avtext-framemark...
Definition: rtp.c:271