rtp.h
Go to the documentation of this file.
1
13#ifndef JANUS_RTP_H
14#define JANUS_RTP_H
15
16#include <arpa/inet.h>
17#if defined (__MACH__) || defined(__FreeBSD__)
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#include "plugins/plugin.h"
31
32#define RTP_HEADER_SIZE 12
33
35typedef struct rtp_header
36{
37#if __BYTE_ORDER == __BIG_ENDIAN
38 uint16_t version:2;
39 uint16_t padding:1;
40 uint16_t extension:1;
41 uint16_t csrccount:4;
42 uint16_t markerbit:1;
43 uint16_t type:7;
44#elif __BYTE_ORDER == __LITTLE_ENDIAN
45 uint16_t csrccount:4;
46 uint16_t extension:1;
47 uint16_t padding:1;
48 uint16_t version:2;
49 uint16_t type:7;
50 uint16_t markerbit:1;
51#endif
52 uint16_t seq_number;
53 uint32_t timestamp;
54 uint32_t ssrc;
55 uint32_t csrc[16];
58
60typedef struct janus_rtp_packet {
61 char *data;
62 gint length;
63 gint64 created;
67
70 uint16_t type;
71 uint16_t length;
73
75#define JANUS_RTP_EXTMAP_AUDIO_LEVEL "urn:ietf:params:rtp-hdrext:ssrc-audio-level"
77#define JANUS_RTP_EXTMAP_TOFFSET "urn:ietf:params:rtp-hdrext:toffset"
79#define JANUS_RTP_EXTMAP_ABS_SEND_TIME "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time"
81#define JANUS_RTP_EXTMAP_VIDEO_ORIENTATION "urn:3gpp:video-orientation"
83#define JANUS_RTP_EXTMAP_TRANSPORT_WIDE_CC "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01"
85#define JANUS_RTP_EXTMAP_PLAYOUT_DELAY "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay"
87#define JANUS_RTP_EXTMAP_MID "urn:ietf:params:rtp-hdrext:sdes:mid"
89#define JANUS_RTP_EXTMAP_RID "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id"
91#define JANUS_RTP_EXTMAP_REPAIRED_RID "urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id"
93#define JANUS_RTP_EXTMAP_DEPENDENCY_DESC "https://aomediacodec.github.io/av1-rtp-spec/#dependency-descriptor-rtp-header-extension"
95#define JANUS_RTP_EXTMAP_ENCRYPTED "urn:ietf:params:rtp-hdrext:encrypt"
96int janus_rtp_extension_id(const char *type);
97
98
99typedef enum janus_audiocodec {
110const char *janus_audiocodec_name(janus_audiocodec acodec);
113
114typedef enum janus_videocodec {
122const char *janus_videocodec_name(janus_videocodec vcodec);
125
126
130gboolean janus_is_rtp(char *buf, guint len);
131
137char *janus_rtp_payload(char *buf, int len, int *plen);
138
143int janus_rtp_header_extension_get_id(const char *sdp, const char *extension);
144
150const char *janus_rtp_header_extension_get_from_id(const char *sdp, int id);
151
161int janus_rtp_header_extension_parse_audio_level(char *buf, int len, int id, gboolean *vad, int *level);
162
172int janus_rtp_header_extension_parse_video_orientation(char *buf, int len, int id,
173 gboolean *c, gboolean *f, gboolean *r1, gboolean *r0);
174
182int janus_rtp_header_extension_parse_playout_delay(char *buf, int len, int id,
183 uint16_t *min_delay, uint16_t *max_delay);
184
192int janus_rtp_header_extension_parse_mid(char *buf, int len, int id,
193 char *sdes_item, int sdes_len);
194
202int janus_rtp_header_extension_parse_rid(char *buf, int len, int id,
203 char *sdes_item, int sdes_len);
204
212int janus_rtp_header_extension_parse_dependency_desc(char *buf, int len, int id,
213 uint8_t *dd_item, int *dd_len);
214
221int janus_rtp_header_extension_parse_abs_sent_time(char *buf, int len, int id, uint32_t *abs_ts);
222
229int janus_rtp_header_extension_set_abs_send_time(char *buf, int len, int id, uint32_t abs_ts);
230
237int janus_rtp_header_extension_parse_transport_wide_cc(char *buf, int len, int id, uint16_t *transSeqNum);
238
245int janus_rtp_header_extension_set_transport_wide_cc(char *buf, int len, int id, uint16_t transSeqNum);
246
254int janus_rtp_header_extension_replace_id(char *buf, int len, int id, int new_id);
255
265
269
275void janus_rtp_header_update(janus_rtp_header *header, janus_rtp_switching_context *context, gboolean video, int step);
276
277#define RTP_AUDIO_SKEW_TH_MS 120
278#define RTP_VIDEO_SKEW_TH_MS 120
279#define SKEW_DETECTION_WAIT_TIME_SECS 10
280
293
294
298
319 gboolean need_pli;
321
325
332void janus_rtp_simulcasting_prepare(json_t *simulcast, int *rid_ext_id, uint32_t *ssrcs, char **rids);
333
340void janus_rtp_simulcasting_cleanup(int *rid_ext_id, uint32_t *ssrcs, char **rids, janus_mutex *rid_mutex);
341
355 char *buf, int len, uint32_t *ssrcs, char **rids,
358
362
363typedef struct janus_av1_svc_context {
365 uint8_t tcnt;
367 uint8_t tioff;
369 GHashTable *templates;
373 gboolean updated;
375
381 uint8_t id;
387
391
403 uint8_t *dd, int dd_len, uint8_t *template_id);
405
406
407#endif
GMutex janus_mutex
Janus mutex implementation.
Definition: mutex.h:73
Plugin-Core communication (implementation)
struct json_t json_t
Definition: plugin.h:236
const char * janus_videocodec_name(janus_videocodec vcodec)
Definition: rtp.c:954
int janus_rtp_header_extension_set_abs_send_time(char *buf, int len, int id, uint32_t abs_ts)
Helper to set an abs-send-time RTP extension (http://www.webrtc.org/experiments/rtp-hdrext/abs-send-t...
Definition: rtp.c:353
struct janus_rtp_packet janus_rtp_packet
RTP packet.
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:498
struct janus_rtp_header_extension janus_rtp_header_extension
RTP extension.
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:728
void janus_av1_svc_context_reset(janus_av1_svc_context *context)
Set (or reset) the context fields to their default values.
Definition: rtp.c:1224
struct janus_rtp_switching_context janus_rtp_switching_context
RTP context, in order to make sure SSRC changes result in coherent seq/ts increases.
void janus_rtp_switching_context_reset(janus_rtp_switching_context *context)
Set (or reset) the context fields to their default values.
Definition: rtp.c:491
janus_audiocodec
Definition: rtp.h:99
@ JANUS_AUDIOCODEC_PCMU
Definition: rtp.h:104
@ JANUS_AUDIOCODEC_MULTIOPUS
Definition: rtp.h:102
@ JANUS_AUDIOCODEC_OPUSRED
Definition: rtp.h:103
@ JANUS_AUDIOCODEC_NONE
Definition: rtp.h:100
@ JANUS_AUDIOCODEC_ISAC_32K
Definition: rtp.h:107
@ JANUS_AUDIOCODEC_G722
Definition: rtp.h:106
@ JANUS_AUDIOCODEC_OPUS
Definition: rtp.h:101
@ JANUS_AUDIOCODEC_PCMA
Definition: rtp.h:105
@ JANUS_AUDIOCODEC_ISAC_16K
Definition: rtp.h:108
void janus_rtp_simulcasting_context_reset(janus_rtp_simulcasting_context *context)
Set (or reset) the context fields to their default values.
Definition: rtp.c:1009
janus_audiocodec janus_audiocodec_from_name(const char *name)
Definition: rtp.c:906
int janus_rtp_header_extension_parse_dependency_desc(char *buf, int len, int id, uint8_t *dd_item, int *dd_len)
Helper to parse a dependency descriptor RTP extension (https://aomediacodec.github....
Definition: rtp.c:313
void janus_rtp_simulcasting_cleanup(int *rid_ext_id, uint32_t *ssrcs, char **rids, janus_mutex *rid_mutex)
Helper method to cleanup some or all of the simulcasting info (rids and/or SSRCs) we may have prepare...
Definition: rtp.c:1050
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:401
int janus_rtp_header_extension_set_transport_wide_cc(char *buf, int len, int id, uint16_t transSeqNum)
Helper to set a transport wide sequence number (https://tools.ietf.org/html/draft-holmer-rmcat-transp...
Definition: rtp.c:387
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, janus_mutex *rid_mutex)
Process an RTP packet, and decide whether this should be relayed or not, updating the context accordi...
Definition: rtp.c:1070
janus_videocodec
Definition: rtp.h:114
@ JANUS_VIDEOCODEC_NONE
Definition: rtp.h:115
@ JANUS_VIDEOCODEC_H264
Definition: rtp.h:118
@ JANUS_VIDEOCODEC_AV1
Definition: rtp.h:119
@ JANUS_VIDEOCODEC_VP9
Definition: rtp.h:117
@ JANUS_VIDEOCODEC_VP8
Definition: rtp.h:116
@ JANUS_VIDEOCODEC_H265
Definition: rtp.h:120
int janus_rtp_header_extension_parse_abs_sent_time(char *buf, int len, int id, uint32_t *abs_ts)
Helper to parse an abs-send-time RTP extension (http://www.webrtc.org/experiments/rtp-hdrext/abs-send...
Definition: rtp.c:336
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:81
int janus_videocodec_pt(janus_videocodec vcodec)
Definition: rtp.c:989
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:251
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:367
gboolean janus_av1_svc_context_process_dd(janus_av1_svc_context *context, uint8_t *dd, int dd_len, uint8_t *template_id)
Process a Dependency Descriptor payload, updating the SVC context accordingly.
Definition: rtp.c:1233
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
rtp_header janus_rtp_header
Definition: rtp.h:57
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:270
struct janus_av1_svc_template janus_av1_svc_template
Helper struct to track SVC templates.
janus_videocodec janus_videocodec_from_name(const char *name)
Definition: rtp.c:973
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:291
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....
Definition: rtp.c:229
struct rtp_header rtp_header
RTP Header (http://tools.ietf.org/html/rfc3550#section-5.1)
void janus_rtp_simulcasting_prepare(json_t *simulcast, int *rid_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:1020
const char * janus_audiocodec_name(janus_audiocodec acodec)
Definition: rtp.c:881
struct janus_av1_svc_context janus_av1_svc_context
Helper struct for processing and tracking AV1-SVC streams.
int janus_audiocodec_pt(janus_audiocodec acodec)
Definition: rtp.c:928
gboolean janus_is_rtp(char *buf, guint len)
Helper method to demultiplex RTP from other protocols.
Definition: rtp.c:19
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
struct janus_rtp_simulcasting_context janus_rtp_simulcasting_context
Helper struct for processing and tracking simulcast streams.
int janus_rtp_header_extension_parse_audio_level(char *buf, int len, int id, gboolean *vad, int *level)
Helper to parse a ssrc-audio-level RTP extension (https://tools.ietf.org/html/rfc6464)
Definition: rtp.c:214
int janus_rtp_extension_id(const char *type)
Definition: rtp.c:466
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:614
Helper struct for processing and tracking AV1-SVC streams.
Definition: rtp.h:363
uint8_t tcnt
Number of templates advertised via Dependency Descriptor.
Definition: rtp.h:365
int spatial_layers
How many spatial and temporal layers are available.
Definition: rtp.h:371
GHashTable * templates
Map of templates advertised via Dependency Descriptor, indexed by ID.
Definition: rtp.h:369
gboolean updated
Whether this context changed since the last update.
Definition: rtp.h:373
uint8_t tioff
Template ID offset, as advertised via Dependency Descriptor.
Definition: rtp.h:367
int temporal_layers
Definition: rtp.h:371
Helper struct to track SVC templates.
Definition: rtp.h:379
uint8_t id
Template ID.
Definition: rtp.h:381
int spatial
Spatial layer associated to this template.
Definition: rtp.h:383
int temporal
Temporal layer associated to this template.
Definition: rtp.h:385
Janus plugin RTP extensions.
Definition: plugin.h:558
RTP extension.
Definition: rtp.h:69
uint16_t length
Definition: rtp.h:71
uint16_t type
Definition: rtp.h:70
RTP packet.
Definition: rtp.h:60
char * data
Definition: rtp.h:61
gint64 last_retransmit
Definition: rtp.h:64
gint length
Definition: rtp.h:62
gint64 created
Definition: rtp.h:63
janus_plugin_rtp_extensions extensions
Definition: rtp.h:65
Helper struct for processing and tracking simulcast streams.
Definition: rtp.h:299
gboolean changed_temporal
Whether the temporal layer has changed after processing a packet.
Definition: rtp.h:317
int templayer_target
As above, but to handle transitions (e.g., wait for keyframe)
Definition: rtp.h:309
int substream_target
As above, but to handle transitions (e.g., wait for keyframe, or get this if available)
Definition: rtp.h:305
gint rid_ext_id
RTP Stream extension ID, if any.
Definition: rtp.h:301
gboolean changed_substream
Whether the substream has changed after processing a packet.
Definition: rtp.h:315
guint32 drop_trigger
How much time (in us, default 250000) without receiving packets will make us drop to the substream be...
Definition: rtp.h:311
int substream_target_temp
Definition: rtp.h:305
int templayer
Which simulcast temporal layer we should forward back.
Definition: rtp.h:307
int substream
Which simulcast substream we should forward back.
Definition: rtp.h:303
gboolean need_pli
Whether we need to send the user a keyframe request (PLI)
Definition: rtp.h:319
gint64 last_relayed
When we relayed the last packet (used to detect when substreams become unavailable)
Definition: rtp.h:313
RTP context, in order to make sure SSRC changes result in coherent seq/ts increases.
Definition: rtp.h:257
uint32_t prev_ts
Definition: rtp.h:258
uint32_t last_ssrc
Definition: rtp.h:258
gint64 reference_time
Definition: rtp.h:263
uint32_t start_ts
Definition: rtp.h:258
uint16_t base_seq
Definition: rtp.h:259
uint32_t target_ts
Definition: rtp.h:258
gint64 evaluating_start_time
Definition: rtp.h:263
gboolean ts_reset
Definition: rtp.h:260
uint32_t base_ts
Definition: rtp.h:258
uint16_t prev_seq
Definition: rtp.h:259
uint16_t last_seq
Definition: rtp.h:259
gboolean new_ssrc
Definition: rtp.h:260
gint32 prev_delay
Definition: rtp.h:262
gint32 active_delay
Definition: rtp.h:262
uint32_t last_ts
Definition: rtp.h:258
gint16 seq_offset
Definition: rtp.h:261
gint64 start_time
Definition: rtp.h:263
gint64 last_time
Definition: rtp.h:263
uint32_t base_ts_prev
Definition: rtp.h:258
gboolean seq_reset
Definition: rtp.h:260
uint16_t base_seq_prev
Definition: rtp.h:259
gint32 ts_offset
Definition: rtp.h:262
RTP Header (http://tools.ietf.org/html/rfc3550#section-5.1)
Definition: rtp.h:36
uint16_t extension
Definition: rtp.h:40
uint32_t timestamp
Definition: rtp.h:53
uint16_t padding
Definition: rtp.h:39
uint32_t csrc[16]
Definition: rtp.h:55
uint16_t csrccount
Definition: rtp.h:41
uint16_t type
Definition: rtp.h:43
uint32_t ssrc
Definition: rtp.h:54
uint16_t seq_number
Definition: rtp.h:52
uint16_t version
Definition: rtp.h:38
uint16_t markerbit
Definition: rtp.h:42