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Data Structures | Macros | Typedefs | Enumerations | Functions
rtp.h File Reference

RTP processing (headers) More...

#include <arpa/inet.h>
#include <endian.h>
#include <inttypes.h>
#include <string.h>
#include <glib.h>
#include <jansson.h>
#include "plugins/plugin.h"
Include dependency graph for rtp.h:
This graph shows which files directly or indirectly include this file:

Go to the source code of this file.

Data Structures

struct  rtp_header
 RTP Header (http://tools.ietf.org/html/rfc3550#section-5.1) More...
 
struct  janus_rtp_packet
 RTP packet. More...
 
struct  janus_rtp_header_extension
 RTP extension. More...
 
struct  janus_rtp_rfc2833_payload
 RTP RFC2833 payload. More...
 
struct  janus_rtp_switching_context
 RTP context, in order to make sure SSRC changes result in coherent seq/ts increases. More...
 
struct  janus_rtp_simulcasting_context
 Helper struct for processing and tracking simulcast streams. More...
 
struct  janus_av1_svc_context
 Helper struct for processing and tracking AV1-SVC streams. More...
 
struct  janus_av1_svc_template
 Helper struct to track SVC templates. More...
 

Macros

#define RTP_HEADER_SIZE   12
 
#define JANUS_RTP_EXTMAP_AUDIO_LEVEL   "urn:ietf:params:rtp-hdrext:ssrc-audio-level"
 a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
 
#define JANUS_RTP_EXTMAP_TOFFSET   "urn:ietf:params:rtp-hdrext:toffset"
 a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
 
#define JANUS_RTP_EXTMAP_ABS_SEND_TIME   "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time"
 a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
 
#define JANUS_RTP_EXTMAP_VIDEO_ORIENTATION   "urn:3gpp:video-orientation"
 a=extmap:4 urn:3gpp:video-orientation
 
#define JANUS_RTP_EXTMAP_TRANSPORT_WIDE_CC   "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01"
 a=extmap:5 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
 
#define JANUS_RTP_EXTMAP_PLAYOUT_DELAY   "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay"
 a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
 
#define JANUS_RTP_EXTMAP_MID   "urn:ietf:params:rtp-hdrext:sdes:mid"
 a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid
 
#define JANUS_RTP_EXTMAP_RID   "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id"
 a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
 
#define JANUS_RTP_EXTMAP_REPAIRED_RID   "urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id"
 a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
 
#define JANUS_RTP_EXTMAP_DEPENDENCY_DESC   "https://aomediacodec.github.io/av1-rtp-spec/#dependency-descriptor-rtp-header-extension"
 a=extmap:10 https://aomediacodec.github.io/av1-rtp-spec/#dependency-descriptor-rtp-header-extension
 
#define JANUS_RTP_EXTMAP_ENCRYPTED   "urn:ietf:params:rtp-hdrext:encrypt"
 
#define RTP_AUDIO_SKEW_TH_MS   120
 
#define RTP_VIDEO_SKEW_TH_MS   120
 
#define SKEW_DETECTION_WAIT_TIME_SECS   10
 

Typedefs

typedef struct rtp_header rtp_header
 RTP Header (http://tools.ietf.org/html/rfc3550#section-5.1)
 
typedef rtp_header janus_rtp_header
 
typedef struct janus_rtp_packet janus_rtp_packet
 RTP packet.
 
typedef struct janus_rtp_header_extension janus_rtp_header_extension
 RTP extension.
 
typedef struct janus_rtp_rfc2833_payload janus_rtp_rfc2833_payload
 RTP RFC2833 payload.
 
typedef enum janus_audiocodec janus_audiocodec
 
typedef enum janus_videocodec janus_videocodec
 
typedef struct janus_rtp_switching_context janus_rtp_switching_context
 RTP context, in order to make sure SSRC changes result in coherent seq/ts increases.
 

Enumerations

enum  janus_audiocodec {
  JANUS_AUDIOCODEC_NONE , JANUS_AUDIOCODEC_OPUS , JANUS_AUDIOCODEC_MULTIOPUS , JANUS_AUDIOCODEC_OPUSRED ,
  JANUS_AUDIOCODEC_PCMU , JANUS_AUDIOCODEC_PCMA , JANUS_AUDIOCODEC_G722 , JANUS_AUDIOCODEC_ISAC_32K ,
  JANUS_AUDIOCODEC_ISAC_16K
}
 
enum  janus_videocodec {
  JANUS_VIDEOCODEC_NONE , JANUS_VIDEOCODEC_VP8 , JANUS_VIDEOCODEC_VP9 , JANUS_VIDEOCODEC_H264 ,
  JANUS_VIDEOCODEC_AV1 , JANUS_VIDEOCODEC_H265
}
 

Functions

const char * janus_audiocodec_name (janus_audiocodec acodec)
 
janus_audiocodec janus_audiocodec_from_name (const char *name)
 
int janus_audiocodec_pt (janus_audiocodec acodec)
 
const char * janus_videocodec_name (janus_videocodec vcodec)
 
janus_videocodec janus_videocodec_from_name (const char *name)
 
int janus_videocodec_pt (janus_videocodec vcodec)
 
gboolean janus_is_rtp (char *buf, guint len)
 Helper method to demultiplex RTP from other protocols.
 
char * janus_rtp_payload (char *buf, int len, int *plen)
 Helper to quickly access the RTP payload, skipping header and extensions.
 
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.
 
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 SDP.
 
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)
 
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)
 
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-delay)
 
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-negotiation-54)
 
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)
 
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.io/av1-rtp-spec/#dependency-descriptor-rtp-header-extension)
 
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-time)
 
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-time)
 
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-transport-wide-cc-extensions-01)
 
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-transport-wide-cc-extensions-01)
 
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-stream-id into a rtp-stream-id after a successful rtx)
 
void janus_rtp_switching_context_reset (janus_rtp_switching_context *context)
 Set (or reset) the context fields to their default values.
 
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.
 
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.
 
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.
 

Janus simulcast processing methods

typedef struct janus_rtp_simulcasting_context janus_rtp_simulcasting_context
 Helper struct for processing and tracking simulcast streams.
 
void janus_rtp_simulcasting_context_reset (janus_rtp_simulcasting_context *context)
 Set (or reset) the context fields to their default values.
 
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 passes to plugins for new PeerConnections.
 
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 accordingly.
 

Janus AV1-SVC processing methods

typedef struct janus_av1_svc_context janus_av1_svc_context
 Helper struct for processing and tracking AV1-SVC streams.
 
typedef struct janus_av1_svc_template janus_av1_svc_template
 Helper struct to track SVC templates.
 
void janus_av1_svc_context_reset (janus_av1_svc_context *context)
 Set (or reset) the context fields to their default values.
 
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.
 

Detailed Description

RTP processing (headers)

Author
Lorenzo Miniero loren.nosp@m.zo@m.nosp@m.eetec.nosp@m.ho.c.nosp@m.om

Implementation of the RTP header. Since the server does not much more than relaying frames around, the only thing we're interested in is the RTP header and how to get its payload, and parsing extensions.

Protocols

Macro Definition Documentation

◆ JANUS_RTP_EXTMAP_ABS_SEND_TIME

#define JANUS_RTP_EXTMAP_ABS_SEND_TIME   "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time"

◆ JANUS_RTP_EXTMAP_AUDIO_LEVEL

#define JANUS_RTP_EXTMAP_AUDIO_LEVEL   "urn:ietf:params:rtp-hdrext:ssrc-audio-level"

a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level

◆ JANUS_RTP_EXTMAP_DEPENDENCY_DESC

#define JANUS_RTP_EXTMAP_DEPENDENCY_DESC   "https://aomediacodec.github.io/av1-rtp-spec/#dependency-descriptor-rtp-header-extension"

◆ JANUS_RTP_EXTMAP_ENCRYPTED

#define JANUS_RTP_EXTMAP_ENCRYPTED   "urn:ietf:params:rtp-hdrext:encrypt"
Note
Note: We don't support encrypted extensions yet

◆ JANUS_RTP_EXTMAP_MID

#define JANUS_RTP_EXTMAP_MID   "urn:ietf:params:rtp-hdrext:sdes:mid"

a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid

◆ JANUS_RTP_EXTMAP_PLAYOUT_DELAY

#define JANUS_RTP_EXTMAP_PLAYOUT_DELAY   "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay"

◆ JANUS_RTP_EXTMAP_REPAIRED_RID

#define JANUS_RTP_EXTMAP_REPAIRED_RID   "urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id"

a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id

◆ JANUS_RTP_EXTMAP_RID

#define JANUS_RTP_EXTMAP_RID   "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id"

a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id

◆ JANUS_RTP_EXTMAP_TOFFSET

#define JANUS_RTP_EXTMAP_TOFFSET   "urn:ietf:params:rtp-hdrext:toffset"

a=extmap:2 urn:ietf:params:rtp-hdrext:toffset

◆ JANUS_RTP_EXTMAP_TRANSPORT_WIDE_CC

#define JANUS_RTP_EXTMAP_TRANSPORT_WIDE_CC   "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01"

◆ JANUS_RTP_EXTMAP_VIDEO_ORIENTATION

#define JANUS_RTP_EXTMAP_VIDEO_ORIENTATION   "urn:3gpp:video-orientation"

a=extmap:4 urn:3gpp:video-orientation

◆ RTP_AUDIO_SKEW_TH_MS

#define RTP_AUDIO_SKEW_TH_MS   120

◆ RTP_HEADER_SIZE

#define RTP_HEADER_SIZE   12

◆ RTP_VIDEO_SKEW_TH_MS

#define RTP_VIDEO_SKEW_TH_MS   120

◆ SKEW_DETECTION_WAIT_TIME_SECS

#define SKEW_DETECTION_WAIT_TIME_SECS   10

Typedef Documentation

◆ janus_audiocodec

◆ janus_av1_svc_context

typedef struct janus_av1_svc_context janus_av1_svc_context

Helper struct for processing and tracking AV1-SVC streams.

◆ janus_av1_svc_template

typedef struct janus_av1_svc_template janus_av1_svc_template

Helper struct to track SVC templates.

Note
This is very incomplete, since we only track the spatial and temporal layer associated with a specific template ID for now

◆ janus_rtp_header

◆ janus_rtp_header_extension

typedef struct janus_rtp_header_extension janus_rtp_header_extension

RTP extension.

◆ janus_rtp_packet

typedef struct janus_rtp_packet janus_rtp_packet

RTP packet.

◆ janus_rtp_rfc2833_payload

typedef struct janus_rtp_rfc2833_payload janus_rtp_rfc2833_payload

RTP RFC2833 payload.

◆ janus_rtp_simulcasting_context

typedef struct janus_rtp_simulcasting_context janus_rtp_simulcasting_context

Helper struct for processing and tracking simulcast streams.

◆ janus_rtp_switching_context

typedef struct janus_rtp_switching_context janus_rtp_switching_context

RTP context, in order to make sure SSRC changes result in coherent seq/ts increases.

◆ janus_videocodec

◆ rtp_header

typedef struct rtp_header rtp_header

Enumeration Type Documentation

◆ janus_audiocodec

Enumerator
JANUS_AUDIOCODEC_NONE 
JANUS_AUDIOCODEC_OPUS 
JANUS_AUDIOCODEC_MULTIOPUS 
JANUS_AUDIOCODEC_OPUSRED 
JANUS_AUDIOCODEC_PCMU 
JANUS_AUDIOCODEC_PCMA 
JANUS_AUDIOCODEC_G722 
JANUS_AUDIOCODEC_ISAC_32K 
JANUS_AUDIOCODEC_ISAC_16K 

◆ janus_videocodec

Enumerator
JANUS_VIDEOCODEC_NONE 
JANUS_VIDEOCODEC_VP8 
JANUS_VIDEOCODEC_VP9 
JANUS_VIDEOCODEC_H264 
JANUS_VIDEOCODEC_AV1 
JANUS_VIDEOCODEC_H265 

Function Documentation

◆ janus_audiocodec_from_name()

janus_audiocodec janus_audiocodec_from_name ( const char * name)

◆ janus_audiocodec_name()

const char * janus_audiocodec_name ( janus_audiocodec acodec)

◆ janus_audiocodec_pt()

int janus_audiocodec_pt ( janus_audiocodec acodec)

◆ janus_av1_svc_context_process_dd()

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.

Note
At the moment, this code is quite naive, as it mostly looks at the target spatial/temporal layers, and the one written in the Dependency Descriptor data. In the future, this should become more sophisticated, and use additional information like dependency chains and stuff like that
Parameters
[in]contextThe av1svc context to use
[in]ddPointer to the Dependency Descriptor data
[in]dd_lenThe length of the Dependendy Descriptor data
[out]template_idPointer to the ID of the template referenced in this packet
Returns
TRUE if the packet is valid, FALSE if it should be dropped instead

◆ janus_av1_svc_context_reset()

void janus_av1_svc_context_reset ( janus_av1_svc_context * context)

Set (or reset) the context fields to their default values.

Parameters
[in]contextThe context to (re)set

◆ janus_is_rtp()

gboolean janus_is_rtp ( char * buf,
guint len )

Helper method to demultiplex RTP from other protocols.

Parameters
[in]bufBuffer to inspect
[in]lenLength of the buffer to inspect

◆ janus_rtp_header_extension_get_from_id()

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 SDP.

Note
This only looks for the extensions we know about, those defined in rtp.h
Parameters
sdpThe SDP to parse
idThe extension id to look for
Returns
The extension namespace, if found, NULL otherwise

◆ janus_rtp_header_extension_get_id()

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.

Parameters
sdpThe SDP to parse
extensionThe extension namespace to look for
Returns
The extension id, if found, -1 otherwise

◆ janus_rtp_header_extension_parse_abs_sent_time()

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-time)

Parameters
[in]bufThe packet data
[in]lenThe packet data length in bytes
[in]idThe extension ID to look for
[out]abs_tsVariable where the parsed abs-send-time value will be stored
Returns
0 if found, -1 otherwise

◆ janus_rtp_header_extension_parse_audio_level()

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)

Note
Browsers apparently always set the VAD to 1, so it's unreliable and should be ignored: only use this method if you're interested in the audio-level value itself.
Parameters
[in]bufThe packet data
[in]lenThe packet data length in bytes
[in]idThe extension ID to look for
[out]vadWhether the encoder thinks there's voice activity
[out]levelThe level value in dBov (0=max, 127=min)
Returns
0 if found, -1 otherwise

◆ janus_rtp_header_extension_parse_dependency_desc()

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.io/av1-rtp-spec/#dependency-descriptor-rtp-header-extension)

Parameters
[in]bufThe packet data
[in]lenThe packet data length in bytes
[in]idThe extension ID to look for
[out]dd_itemBuffer where the dependency descriptor will be written
[out]dd_lenSize of the input/output buffer, will be updated with the size of the data
Returns
0 if found, -1 otherwise

◆ janus_rtp_header_extension_parse_mid()

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-negotiation-54)

Parameters
[in]bufThe packet data
[in]lenThe packet data length in bytes
[in]idThe extension ID to look for
[out]sdes_itemBuffer where the RTP stream ID will be written
[in]sdes_lenSize of the input/output buffer
Returns
0 if found, -1 otherwise

◆ janus_rtp_header_extension_parse_playout_delay()

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-delay)

Parameters
[in]bufThe packet data
[in]lenThe packet data length in bytes
[in]idThe extension ID to look for
[out]min_delayThe minimum delay value
[out]max_delayThe maximum delay value
Returns
0 if found, -1 otherwise

◆ janus_rtp_header_extension_parse_rid()

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)

Parameters
[in]bufThe packet data
[in]lenThe packet data length in bytes
[in]idThe extension ID to look for
[out]sdes_itemBuffer where the RTP stream ID will be written
[in]sdes_lenSize of the input/output buffer
Returns
0 if found, -1 otherwise

◆ janus_rtp_header_extension_parse_transport_wide_cc()

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-transport-wide-cc-extensions-01)

Parameters
[in]bufThe packet data
[in]lenThe packet data length in bytes
[in]idThe extension ID to look for
[out]transSeqNumVariable to read the transport wide sequence number in
Returns
0 if found, -1 otherwise

◆ janus_rtp_header_extension_parse_video_orientation()

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)

Parameters
[in]bufThe packet data
[in]lenThe packet data length in bytes
[in]idThe extension ID to look for
[out]cThe value of the Camera (C) bit
[out]fThe value of the Flip (F) bit
[out]r1The value of the first Rotation (R1) bit
[out]r0The value of the second Rotation (R0) bit
Returns
0 if found, -1 otherwise

◆ janus_rtp_header_extension_replace_id()

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-stream-id into a rtp-stream-id after a successful rtx)

Parameters
[in]bufThe packet data
[in]lenThe packet data length in bytes
[in]idThe extension ID to look for and replace
[in]new_idThe new value for the extension ID
Returns
0 if found, a negative integer otherwise

◆ janus_rtp_header_extension_set_abs_send_time()

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-time)

Parameters
[in]bufThe packet data
[in]lenThe packet data length in bytes
[in]idThe extension ID to look for
[out]abs_tsAbsolute Send Time value to set
Returns
0 if found, -1 otherwise

◆ janus_rtp_header_extension_set_transport_wide_cc()

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-transport-wide-cc-extensions-01)

Parameters
[in]bufThe packet data
[in]lenThe packet data length in bytes
[in]idThe extension ID to look for
[out]transSeqNumTransport wide sequence number to set
Returns
0 if found, -1 otherwise

◆ janus_rtp_header_update()

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.

Parameters
[in]headerThe RTP header to update
[in]contextThe context to use as a reference
[in]videoWhether this is an audio or a video packet
[in]stepdeprecated The expected timestamp step

◆ janus_rtp_payload()

char * janus_rtp_payload ( char * buf,
int len,
int * plen )

Helper to quickly access the RTP payload, skipping header and extensions.

Parameters
[in]bufThe packet data
[in]lenThe packet data length in bytes
[out]plenThe payload data length in bytes
Returns
A pointer to where the payload data starts, or NULL otherwise; plen is also set accordingly

◆ janus_rtp_simulcasting_context_process_rtp()

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 accordingly.

Note
Calling this method resets the changed_substream , changed_temporal and need_pli properties, and updates them according to the decisions made after processing the packet
Parameters
[in]contextThe simulcasting context to use
[in]bufThe RTP packet to process
[in]lenThe length of the RTP packet (header, extension and payload)
[in]ssrcsThe simulcast SSRCs to refer to (may be updated if rids are involved)
[in]ridsThe simulcast rids to refer to, if any
[in]vcodecVideo codec of the RTP payload
[in]scRTP switching context to refer to, if any (only needed for VP8 and dropping temporal layers)
Returns
TRUE if the packet should be relayed, FALSE if it should be dropped instead

◆ janus_rtp_simulcasting_context_reset()

void janus_rtp_simulcasting_context_reset ( janus_rtp_simulcasting_context * context)

Set (or reset) the context fields to their default values.

Parameters
[in]contextThe context to (re)set

◆ janus_rtp_simulcasting_prepare()

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 passes to plugins for new PeerConnections.

Parameters
[in]simulcastJSON object containing SSRCs and rids
[in]rid_ext_idThe rid RTP extension ID to set, if any
[in]ssrcsThe list of simulcast SSRCs to update, if any
[in]ridsThe list of rids to update, if any (items will be allocated)

◆ janus_rtp_skew_compensate_audio()

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.

Parameters
[in]headerThe RTP header to update
[in]contextThe context to use as a reference
[in]nowThe packet arrival monotonic time
Returns
0 if no compensation is needed, -N if a N packets drop must be performed, N if a N sequence numbers jump has been performed

◆ janus_rtp_skew_compensate_video()

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.

Parameters
[in]headerThe RTP header to update
[in]contextThe context to use as a reference
[in]nowThe packet arrival monotonic time
Returns
0 if no compensation is needed, -N if a N packets drop must be performed, N if a N sequence numbers jump has been performed

◆ janus_rtp_switching_context_reset()

void janus_rtp_switching_context_reset ( janus_rtp_switching_context * context)

Set (or reset) the context fields to their default values.

Parameters
[in]contextThe context to (re)set

◆ janus_videocodec_from_name()

janus_videocodec janus_videocodec_from_name ( const char * name)

◆ janus_videocodec_name()

const char * janus_videocodec_name ( janus_videocodec vcodec)

◆ janus_videocodec_pt()

int janus_videocodec_pt ( janus_videocodec vcodec)