rtp.h
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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 #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];
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"
74 #define JANUS_RTP_EXTMAP_TOFFSET "urn:ietf:params:rtp-hdrext:toffset"
76 #define JANUS_RTP_EXTMAP_ABS_SEND_TIME "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time"
78 #define JANUS_RTP_EXTMAP_VIDEO_ORIENTATION "urn:3gpp:video-orientation"
80 #define JANUS_RTP_EXTMAP_TRANSPORT_WIDE_CC "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01"
82 #define JANUS_RTP_EXTMAP_PLAYOUT_DELAY "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay"
84 #define JANUS_RTP_EXTMAP_MID "urn:ietf:params:rtp-hdrext:sdes:mid"
86 #define JANUS_RTP_EXTMAP_RID "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id"
88 #define JANUS_RTP_EXTMAP_REPAIRED_RID "urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id"
90 #define JANUS_RTP_EXTMAP_ENCRYPTED "urn:ietf:params:rtp-hdrext:encrypt"
91 
92 
93 typedef enum janus_audiocodec {
103 const char *janus_audiocodec_name(janus_audiocodec acodec);
106 
107 typedef enum janus_videocodec {
115 const char *janus_videocodec_name(janus_videocodec vcodec);
118 
119 
123 gboolean janus_is_rtp(char *buf, guint len);
124 
130 char *janus_rtp_payload(char *buf, int len, int *plen);
131 
136 int janus_rtp_header_extension_get_id(const char *sdp, const char *extension);
137 
143 const char *janus_rtp_header_extension_get_from_id(const char *sdp, int id);
144 
154 int janus_rtp_header_extension_parse_audio_level(char *buf, int len, int id, gboolean *vad, int *level);
155 
165 int janus_rtp_header_extension_parse_video_orientation(char *buf, int len, int id,
166  gboolean *c, gboolean *f, gboolean *r1, gboolean *r0);
167 
175 int janus_rtp_header_extension_parse_playout_delay(char *buf, int len, int id,
176  uint16_t *min_delay, uint16_t *max_delay);
177 
185 int janus_rtp_header_extension_parse_mid(char *buf, int len, int id,
186  char *sdes_item, int sdes_len);
187 
195 int janus_rtp_header_extension_parse_rid(char *buf, int len, int id,
196  char *sdes_item, int sdes_len);
197 
204 int janus_rtp_header_extension_parse_transport_wide_cc(char *buf, int len, int id, uint16_t *transSeqNum);
205 
212 int janus_rtp_header_extension_set_transport_wide_cc(char *buf, int len, int id, uint16_t transSeqNum);
213 
221 int janus_rtp_header_extension_replace_id(char *buf, int len, int id, int new_id);
222 
231  gint16 a_seq_offset,
238 
242 
248 void janus_rtp_header_update(janus_rtp_header *header, janus_rtp_switching_context *context, gboolean video, int step);
249 
250 #define RTP_AUDIO_SKEW_TH_MS 120
251 #define RTP_VIDEO_SKEW_TH_MS 120
252 #define SKEW_DETECTION_WAIT_TIME_SECS 10
253 
266 
267 
281  guint32 drop_trigger;
283  gint64 last_relayed;
289  gboolean need_pli;
291 
295 
302 void janus_rtp_simulcasting_prepare(json_t *simulcast, int *rid_ext_id, uint32_t *ssrcs, char **rids);
303 
316  char *buf, int len, uint32_t *ssrcs, char **rids,
318 
319 #endif
struct json_t json_t
Definition: plugin.h:236
struct janus_rtp_packet janus_rtp_packet
RTP packet.
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
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:358
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:588
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
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:351
janus_audiocodec
Definition: rtp.h:93
@ JANUS_AUDIOCODEC_PCMU
Definition: rtp.h:97
@ JANUS_AUDIOCODEC_MULTIOPUS
Definition: rtp.h:96
@ JANUS_AUDIOCODEC_NONE
Definition: rtp.h:94
@ JANUS_AUDIOCODEC_ISAC_32K
Definition: rtp.h:100
@ JANUS_AUDIOCODEC_G722
Definition: rtp.h:99
@ JANUS_AUDIOCODEC_OPUS
Definition: rtp.h:95
@ JANUS_AUDIOCODEC_PCMA
Definition: rtp.h:98
@ JANUS_AUDIOCODEC_ISAC_16K
Definition: rtp.h:101
void janus_rtp_simulcasting_context_reset(janus_rtp_simulcasting_context *context)
Set (or reset) the context fields to their default values.
Definition: rtp.c:897
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:938
const char * janus_audiocodec_name(janus_audiocodec acodec)
Definition: rtp.c:775
janus_audiocodec janus_audiocodec_from_name(const char *name)
Definition: rtp.c:798
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:307
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:293
janus_videocodec
Definition: rtp.h:107
@ JANUS_VIDEOCODEC_NONE
Definition: rtp.h:108
@ JANUS_VIDEOCODEC_H264
Definition: rtp.h:111
@ JANUS_VIDEOCODEC_AV1
Definition: rtp.h:112
@ JANUS_VIDEOCODEC_VP9
Definition: rtp.h:110
@ JANUS_VIDEOCODEC_VP8
Definition: rtp.h:109
@ JANUS_VIDEOCODEC_H265
Definition: rtp.h:113
int janus_videocodec_pt(janus_videocodec vcodec)
Definition: rtp.c:877
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:214
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:273
rtp_header janus_rtp_header
Definition: rtp.h:55
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:230
janus_videocodec janus_videocodec_from_name(const char *name)
Definition: rtp.c:861
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:251
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:192
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:908
int janus_audiocodec_pt(janus_audiocodec acodec)
Definition: rtp.c:818
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:177
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:474
const char * janus_videocodec_name(janus_videocodec vcodec)
Definition: rtp.c:842
RTP extension.
Definition: rtp.h:66
uint16_t length
Definition: rtp.h:68
uint16_t type
Definition: rtp.h:67
RTP packet.
Definition: rtp.h:58
char * data
Definition: rtp.h:59
gint64 last_retransmit
Definition: rtp.h:62
gint length
Definition: rtp.h:60
gint64 created
Definition: rtp.h:61
Helper struct for processing and tracking simulcast streams.
Definition: rtp.h:269
gboolean changed_temporal
Whether the temporal layer has changed after processing a packet.
Definition: rtp.h:287
int templayer_target
As above, but to handle transitions (e.g., wait for keyframe)
Definition: rtp.h:279
int substream_target
As above, but to handle transitions (e.g., wait for keyframe, or get this if available)
Definition: rtp.h:275
gint rid_ext_id
RTP Stream extension ID, if any.
Definition: rtp.h:271
gboolean changed_substream
Whether the substream has changed after processing a packet.
Definition: rtp.h:285
guint32 drop_trigger
How much time (in us, default 250000) without receiving packets will make us drop to the substream be...
Definition: rtp.h:281
int substream_target_temp
Definition: rtp.h:275
int templayer
Which simulcast temporal layer we should forward back.
Definition: rtp.h:277
int substream
Which simulcast substream we should forward back.
Definition: rtp.h:273
gboolean need_pli
Whether we need to send the user a keyframe request (PLI)
Definition: rtp.h:289
gint64 last_relayed
When we relayed the last packet (used to detect when substreams become unavailable)
Definition: rtp.h:283
RTP context, in order to make sure SSRC changes result in coherent seq/ts increases.
Definition: rtp.h:224
uint32_t v_start_ts
Definition: rtp.h:226
uint32_t a_prev_ts
Definition: rtp.h:225
gboolean a_seq_reset
Definition: rtp.h:229
gboolean v_seq_reset
Definition: rtp.h:230
uint32_t v_base_ts
Definition: rtp.h:226
gint64 a_reference_time
Definition: rtp.h:235
gint32 v_prev_delay
Definition: rtp.h:234
gint32 v_ts_offset
Definition: rtp.h:234
uint16_t v_base_seq_prev
Definition: rtp.h:228
gint32 v_active_delay
Definition: rtp.h:234
uint32_t a_last_ssrc
Definition: rtp.h:225
uint16_t a_base_seq
Definition: rtp.h:227
uint16_t v_base_seq
Definition: rtp.h:228
uint32_t a_target_ts
Definition: rtp.h:225
gint64 a_evaluating_start_time
Definition: rtp.h:235
uint32_t a_start_ts
Definition: rtp.h:225
gboolean a_new_ssrc
Definition: rtp.h:229
uint32_t a_base_ts
Definition: rtp.h:225
uint32_t a_base_ts_prev
Definition: rtp.h:225
gboolean v_new_ssrc
Definition: rtp.h:230
gint64 v_last_time
Definition: rtp.h:236
gint16 v_seq_offset
Definition: rtp.h:232
uint16_t a_base_seq_prev
Definition: rtp.h:227
uint16_t v_last_seq
Definition: rtp.h:228
gint64 v_start_time
Definition: rtp.h:236
uint16_t v_prev_seq
Definition: rtp.h:228
gint64 a_start_time
Definition: rtp.h:235
gint32 a_prev_delay
Definition: rtp.h:233
uint32_t v_last_ts
Definition: rtp.h:226
gint64 a_last_time
Definition: rtp.h:235
gint16 a_seq_offset
Definition: rtp.h:231
uint32_t a_last_ts
Definition: rtp.h:225
gint32 a_active_delay
Definition: rtp.h:233
uint32_t v_target_ts
Definition: rtp.h:226
uint16_t a_prev_seq
Definition: rtp.h:227
gint64 v_evaluating_start_time
Definition: rtp.h:236
uint16_t a_last_seq
Definition: rtp.h:227
uint32_t v_prev_ts
Definition: rtp.h:226
gint64 v_reference_time
Definition: rtp.h:236
uint32_t v_base_ts_prev
Definition: rtp.h:226
gint32 a_ts_offset
Definition: rtp.h:233
uint32_t v_last_ssrc
Definition: rtp.h:226
RTP Header (http://tools.ietf.org/html/rfc3550#section-5.1)
Definition: rtp.h:34
uint16_t extension
Definition: rtp.h:38
uint32_t timestamp
Definition: rtp.h:51
uint16_t padding
Definition: rtp.h:37
uint32_t csrc[16]
Definition: rtp.h:53
uint16_t csrccount
Definition: rtp.h:39
uint16_t type
Definition: rtp.h:41
uint32_t ssrc
Definition: rtp.h:52
uint16_t seq_number
Definition: rtp.h:50
uint16_t version
Definition: rtp.h:36
uint16_t markerbit
Definition: rtp.h:40