Loading...
Searching...
No Matches
rtp.c File Reference
Core » Protocols

RTP processing. More...

`#include <string.h>`
`#include "rtp.h"`
`#include "rtpsrtp.h"`
`#include "debug.h"`
`#include "utils.h"`
Include dependency graph for rtp.c:

## Macros

#define | OPUS_PT 111
#define | MULTIOPUS_PT OPUS_PT
#define | OPUSRED_PT 120
#define | ISAC32_PT 104
#define | ISAC16_PT 103
#define | PCMU_PT 0
#define | PCMA_PT 8
#define | G722_PT 9
#define | L16_48_PT 105
#define | L16_PT 106
#define | VP8_PT 96
#define | VP9_PT 101
#define | H264_PT 107
#define | AV1_PT 98
#define | H265_PT 100

## Functions

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)
int | janus_rtp_extension_id (const char *type)
void | janus_rtp_switching_context_reset (janus_rtp_switching_context *context)
| Set (or reset) the context fields to their default values.
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.
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.
const char * | janus_srtp_error_str (int error)
| Helper method to get a string representation of a libsrtp error code.
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)
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.
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 prepared before via janus_rtp_simulcasting_prepare.
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 accordingly.
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.

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

Copyright
  GNU General Public License v3

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

## AV1_PT

#define AV1_PT 98

## G722_PT

#define G722_PT 9

## H264_PT

#define H264_PT 107

## H265_PT

#define H265_PT 100

## ISAC16_PT

#define ISAC16_PT 103

## ISAC32_PT

#define ISAC32_PT 104

## L16_48_PT

#define L16_48_PT 105

## L16_PT

#define L16_PT 106

## MULTIOPUS_PT

#define MULTIOPUS_PT OPUS_PT

## OPUS_PT

#define OPUS_PT 111

## OPUSRED_PT

#define OPUSRED_PT 120

## PCMA_PT

#define PCMA_PT 8

## PCMU_PT

#define PCMU_PT 0

## VP8_PT

#define VP8_PT 96

## VP9_PT

#define VP9_PT 101

## 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] | context | The av1svc context to use
  [in] | dd | Pointer to the Dependency Descriptor data
  [in] | dd_len | The length of the Dependendy Descriptor data
  [out] | template_id | Pointer 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] | context | The 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] | buf | Buffer to inspect
  [in] | len | Length of the buffer to inspect

## janus_rtp_extension_id()

int janus_rtp_extension_id | ( | const char * | type | ) |

## 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

  sdp | The SDP to parse
  id | The 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

  sdp | The SDP to parse
  extension | The 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] | buf | The packet data
  [in] | len | The packet data length in bytes
  [in] | id | The extension ID to look for
  [out] | abs_ts | Variable 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] | buf | The packet data
  [in] | len | The packet data length in bytes
  [in] | id | The extension ID to look for
  [out] | vad | Whether the encoder thinks there's voice activity
  [out] | level | The 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] | buf | The packet data
  [in] | len | The packet data length in bytes
  [in] | id | The extension ID to look for
  [out] | dd_item | Buffer where the dependency descriptor will be written
  [out] | dd_len | Size 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] | buf | The packet data
  [in] | len | The packet data length in bytes
  [in] | id | The extension ID to look for
  [out] | sdes_item | Buffer where the RTP stream ID will be written
  [in] | sdes_len | Size 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] | buf | The packet data
  [in] | len | The packet data length in bytes
  [in] | id | The extension ID to look for
  [out] | min_delay | The minimum delay value
  [out] | max_delay | The 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] | buf | The packet data
  [in] | len | The packet data length in bytes
  [in] | id | The extension ID to look for
  [out] | sdes_item | Buffer where the RTP stream ID will be written
  [in] | sdes_len | Size 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] | buf | The packet data
  [in] | len | The packet data length in bytes
  [in] | id | The extension ID to look for
  [out] | transSeqNum | Variable 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] | buf | The packet data
  [in] | len | The packet data length in bytes
  [in] | id | The extension ID to look for
  [out] | c | The value of the Camera (C) bit
  [out] | f | The value of the Flip (F) bit
  [out] | r1 | The value of the first Rotation (R1) bit
  [out] | r0 | The 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] | buf | The packet data
  [in] | len | The packet data length in bytes
  [in] | id | The extension ID to look for and replace
  [in] | new_id | The 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] | buf | The packet data
  [in] | len | The packet data length in bytes
  [in] | id | The extension ID to look for
  [out] | abs_ts | Absolute 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] | buf | The packet data
  [in] | len | The packet data length in bytes
  [in] | id | The extension ID to look for
  [out] | transSeqNum | Transport 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] | header | The RTP header to update
  [in] | context | The context to use as a reference
  [in] | video | Whether this is an audio or a video packet
  [in] | step | deprecated 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] | buf | The packet data
  [in] | len | The packet data length in bytes
  [out] | plen | The 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_cleanup()

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 prepared before via janus_rtp_simulcasting_prepare.

Parameters

  [in] | rid_ext_id | The rid RTP extension ID to cleanup, if any
  [in] | ssrcs | The list of simulcast SSRCs to cleanup, if any
  [in] | rids | The list of rids to cleanup, if any (items will be freed and NULL-ed)
  [in] | rid_mutex | A mutex that must be acquired before cleaning up, if any

## 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,
|  | janus_mutex * | rid_mutex
| ) |  |

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] | context | The simulcasting context to use
  [in] | buf | The RTP packet to process
  [in] | len | The length of the RTP packet (header, extension and payload)
  [in] | ssrcs | The simulcast SSRCs to refer to (may be updated if rids are involved)
  [in] | rids | The simulcast rids to refer to, if any
  [in] | vcodec | Video codec of the RTP payload
  [in] | sc | RTP switching context to refer to, if any (only needed for VP8 and dropping temporal layers)
  [in] | rid_mutex | A mutex that must be acquired before reading the rids array, if any

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] | context | The 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] | simulcast | JSON object containing SSRCs and rids
  [in] | rid_ext_id | The rid RTP extension ID to set, if any
  [in] | ssrcs | The list of simulcast SSRCs to update, if any
  [in] | rids | The 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] | header | The RTP header to update
  [in] | context | The context to use as a reference
  [in] | now | The 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] | header | The RTP header to update
  [in] | context | The context to use as a reference
  [in] | now | The 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] | context | The context to (re)set

## janus_srtp_error_str()

const char * janus_srtp_error_str | ( | int | error | ) |

Helper method to get a string representation of a libsrtp error code.

Parameters

  [in] | error | The libsrtp error code

Returns
  A string representation of the error code

## 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 | ) |