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	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

#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. More...

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, gboolean vad, 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_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) More...

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) More...

int	janus_rtp_header_extension_parse_framemarking (char buf, int len, int id, janus_videocodec codec, uint8_t tid)
	Helper to parse a frame-marking RTP extension (http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07) More...

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) More...

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) More...

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) 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...

void	janus_rtp_simulcasting_prepare (json_t simulcast, int rid_ext_id, int framemarking_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. More...

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. More...

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

◆ MULTIOPUS_PT

#define MULTIOPUS_PT OPUS_PT

◆ OPUS_PT

#define OPUS_PT 111

◆ 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_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_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_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_framemarking()

int janus_rtp_header_extension_parse_framemarking	(	char *	buf,
		int	len,
		int	id,
		janus_videocodec	codec,
		uint8_t *	tid
	)

Helper to parse a frame-marking RTP extension (http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07)

Note: This is currently only used to get temporal layers for H.264 simulcasting

Parameters

[in]	buf	The packet data
[in]	len	The packet data length in bytes
[in]	id	The extension ID to look for
[in]	codec	The video codec (as a janus_videocodec instance) the extension refers to
[out]	tid	Temporal layer ID of the frame

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_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_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 processinf 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)

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,
		int *	framemarking_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]	framemarking_ext_id	The frame marking 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 )

Macros

Functions

Detailed Description

Macro Definition Documentation

◆ AV1_PT

◆ G722_PT

◆ H264_PT

◆ H265_PT

◆ ISAC16_PT

◆ ISAC32_PT

◆ MULTIOPUS_PT

◆ OPUS_PT

◆ PCMA_PT

◆ PCMU_PT

◆ VP8_PT

◆ VP9_PT

Function Documentation

◆ janus_audiocodec_from_name()

◆ janus_audiocodec_name()

◆ janus_audiocodec_pt()

◆ janus_is_rtp()

◆ janus_rtp_header_extension_get_from_id()

◆ janus_rtp_header_extension_get_id()

◆ janus_rtp_header_extension_parse_audio_level()

◆ janus_rtp_header_extension_parse_framemarking()

◆ janus_rtp_header_extension_parse_mid()

◆ janus_rtp_header_extension_parse_playout_delay()

◆ janus_rtp_header_extension_parse_rid()

◆ janus_rtp_header_extension_parse_transport_wide_cc()

◆ janus_rtp_header_extension_parse_video_orientation()

◆ janus_rtp_header_extension_replace_id()

◆ janus_rtp_header_extension_set_transport_wide_cc()

◆ janus_rtp_header_update()

◆ janus_rtp_payload()

◆ janus_rtp_simulcasting_context_process_rtp()

◆ janus_rtp_simulcasting_context_reset()

◆ janus_rtp_simulcasting_prepare()

◆ janus_rtp_skew_compensate_audio()

◆ janus_rtp_skew_compensate_video()

◆ janus_rtp_switching_context_reset()

◆ janus_srtp_error_str()

◆ janus_videocodec_from_name()

◆ janus_videocodec_name()

◆ janus_videocodec_pt()