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Macros | Functions
rtp.c File Reference

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 ISAC32_PT   104
 
#define ISAC16_PT   103
 
#define PCMU_PT   0
 
#define PCMA_PT   8
 
#define G722_PT   9
 
#define VP8_PT   96
 
#define VP9_PT   101
 
#define H264_PT   107
 

Functions

char * janus_rtp_payload (char *buf, int len, int *plen)
 Helper to quickly access the RTP payload, skipping header and extensions. More...
 
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. More...
 
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. More...
 
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) More...
 
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) More...
 
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) More...
 
int janus_rtp_header_extension_parse_rtp_stream_id (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) More...
 
int janus_rtp_header_extension_parse_transport_wide_cc (char *buf, int len, int id, uint16_t *transSeqNum)
 Helper to parse a rtp-stream-id RTP extension (https://tools.ietf.org/html/draft-ietf-avtext-rid-09) More...
 
void janus_rtp_switching_context_reset (janus_rtp_switching_context *context)
 Set (or reset) the context fields to their default values. More...
 
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. More...
 
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. More...
 
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. More...
 
const char * janus_srtp_error_str (int error)
 Helper method to get a string representation of a libsrtp error code. More...
 
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. More...
 
gboolean janus_rtp_simulcasting_context_process_rtp (janus_rtp_simulcasting_context *context, char *buf, int len, uint32_t *ssrcs, 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. More...
 

Detailed Description

RTP processing.

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

#define G722_PT   9
#define H264_PT   107
#define ISAC16_PT   103
#define ISAC32_PT   104
#define OPUS_PT   111
#define PCMA_PT   8
#define PCMU_PT   0
#define VP8_PT   96
#define VP9_PT   101

Function Documentation

janus_audiocodec janus_audiocodec_from_name ( const char *  name)
const char* janus_audiocodec_name ( janus_audiocodec  acodec)
int janus_audiocodec_pt ( janus_audiocodec  acodec)
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
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
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)

Parameters
[in]bufThe packet data
[in]lenThe packet data length in bytes
[in]idThe extension ID to look for
[out]levelThe level value in dBov (0=max, 127=min)
Returns
0 if found, -1 otherwise
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
int janus_rtp_header_extension_parse_rtp_stream_id ( 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
int janus_rtp_header_extension_parse_transport_wide_cc ( char *  buf,
int  len,
int  id,
uint16_t *  transSeqNum 
)

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]transSeqNumtransport wide sequence number
Returns
0 if found, -1 otherwise
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
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
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
gboolean janus_rtp_simulcasting_context_process_rtp ( janus_rtp_simulcasting_context context,
char *  buf,
int  len,
uint32_t *  ssrcs,
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 processinf 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
[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
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
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
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
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
const char* janus_srtp_error_str ( int  error)

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

Parameters
[in]errorThe libsrtp error code
Returns
A string representation of the error code
janus_videocodec janus_videocodec_from_name ( const char *  name)
const char* janus_videocodec_name ( janus_videocodec  vcodec)
int janus_videocodec_pt ( janus_videocodec  vcodec)