rtp.c File Reference

RTP processing. More...

#include <string.h>
#include "rtp.h"
#include "rtpsrtp.h"
#include "debug.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_send_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_abs_capture_time (char *buf, int len, int id, uint64_t *abs_ts)
	Helper to parse an abs-capture-time RTP extension (http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time)
int	janus_rtp_header_extension_set_abs_capture_time (char *buf, int len, int id, uint64_t abs_ts)
	Helper to set an abs-capture-time RTP extension (http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-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, uint8_t *dd_content, int dd_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_rtp_svc_context_reset (janus_rtp_svc_context *context)
	Set (or reset) the context fields to their default values.
gboolean	janus_rtp_svc_context_process_rtp (janus_rtp_svc_context *context, char *buf, int len, uint8_t *dd_content, int dd_len, janus_videocodec vcodec, janus_vp9_svc_info *info, janus_rtp_switching_context *sc)
	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, uint8_t *ebit)
	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,
		uint8_t *	ebit )

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
[out]	ebit	Whether this packet is an end of frame or not

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_capture_time()

int janus_rtp_header_extension_parse_abs_capture_time	(	char *	buf,
		int	len,
		int	id,
		uint64_t *	abs_ts )

Helper to parse an abs-capture-time RTP extension (http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-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-capture-time value will be stored

Returns

0 if found, -1 otherwise

janus_rtp_header_extension_parse_abs_send_time()

int janus_rtp_header_extension_parse_abs_send_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_capture_time()

int janus_rtp_header_extension_set_abs_capture_time	(	char *	buf,
		int	len,
		int	id,
		uint64_t	abs_ts )

Helper to set an abs-capture-time RTP extension (http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-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_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,
		uint8_t *	dd_content,
		int	dd_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]	dd_content	The Dependency Descriptor RTP extension data, if available
[in]	dd_len	Length of the Dependency Descriptor data, if available
[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_svc_context_process_rtp()

gboolean janus_rtp_svc_context_process_rtp	(	janus_rtp_svc_context *	context,
		char *	buf,
		int	len,
		uint8_t *	dd_content,
		int	dd_len,
		janus_videocodec	vcodec,
		janus_vp9_svc_info *	info,
		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_spatial , changed_temporal and need_pli properties, and updates them according to the decisions made after processing the packet

Parameters

[in]	context	The VP9 SVC context to use
[in]	buf	The RTP packet to process
[in]	len	The length of the RTP packet (header, extension and payload)
[in]	dd_content	The Dependency Descriptor RTP extension data, if available
[in]	dd_len	Length of the Dependency Descriptor data, if available
[in]	vcodec	Video codec of the RTP payload
[in]	info	Parsed info on VP9-SVC, if any
[in]	sc	RTP switching context to refer to, if any

Returns

TRUE if the packet should be relayed, FALSE if it should be dropped instead

janus_rtp_svc_context_reset()

void janus_rtp_svc_context_reset

(

janus_rtp_svc_context *

context

)

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

Parameters

[in]

context

The context to (re)set

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

)