Tools for real-time streaming over IP using RTP/RTCP/RTSP/SDP. More...

Collaboration diagram for RTP Streaming:

Data Structures
struct	GF_RTSPRange

struct	GF_RTSPTransport

struct	GF_RTSPCommand

struct	GF_RTPInfo

struct	GF_RTSPResponse

struct	GF_RTPHeader

struct	GF_RTPMap

struct	GF_X_Attribute

struct	GF_SDPBandwidth

struct	GF_SDPTiming

struct	GF_SDPConnection

struct	GF_SDP_FMTP

struct	GF_SDPMedia

struct	GF_SDPInfo

struct	GP_RTPSLMap

struct	GF_RTPStaticMap

Macros
#define	GF_RTSP_VERSION "RTSP/1.0"

#define	GF_RTSP_PROFILE_RTP_AVP "RTP/AVP"

#define	GF_RTSP_PROFILE_RTP_AVP_TCP "RTP/AVP/TCP"

#define	GF_RTSP_PROFILE_UDP "udp"

#define	GF_RTSP_DESCRIBE "DESCRIBE"

#define	GF_RTSP_SETUP "SETUP"

#define	GF_RTSP_PLAY "PLAY"

#define	GF_RTSP_PAUSE "PAUSE"

#define	GF_RTSP_RECORD "RECORD"

#define	GF_RTSP_TEARDOWN "TEARDOWN"

#define	GF_RTSP_GET_PARAMETER "GET_PARAMETER"

#define	GF_RTSP_SET_PARAMETER "SET_PARAMETER"

#define	GF_RTSP_OPTIONS "OPTIONS"

#define	GF_RTSP_ANNOUNCE "ANNOUNCE"

#define	GF_RTSP_REDIRECT "REDIRECT"

#define	GF_RTSP_TUNNEL_POST -1000

#define	GF_SDP_MAX_TIMEOFFSET 10

Typedefs
typedef GF_Err(*	gf_rtsp_interleave_callback) (GF_RTSPSession sess, void cbk_ptr, u8 *buffer, u32 bufferSize, Bool IsRTCP)

typedef GF_Err(*	gf_rtp_tcp_callback) (void cbk1, void cbk2, Bool is_rtcp, u8 *pck, u32 pck_size)

typedef void(*	gf_rtp_packet_cbk) (void udta, u8 payload, u32 size, GF_SLHeader *hdr, GF_Err e)

Enumerations
enum	{ NC_RTSP_Continue = 100 , NC_RTSP_OK = 200 , NC_RTSP_Created = 201 , NC_RTSP_Low_on_Storage_Space = 250 , NC_RTSP_Multiple_Choice = 300 , NC_RTSP_Moved_Permanently = 301 , NC_RTSP_Moved_Temporarily = 302 , NC_RTSP_See_Other = 303 , NC_RTSP_Use_Proxy = 305 , NC_RTSP_Bad_Request = 400 , NC_RTSP_Unauthorized = 401 , NC_RTSP_Payment_Required = 402 , NC_RTSP_Forbidden = 403 , NC_RTSP_Not_Found = 404 , NC_RTSP_Method_Not_Allowed = 405 , NC_RTSP_Not_Acceptable = 406 , NC_RTSP_Proxy_Authentication_Required = 407 , NC_RTSP_Request_Timeout = 408 , NC_RTSP_Gone = 410 , NC_RTSP_Length_Required = 411 , NC_RTSP_Precondition_Failed = 412 , NC_RTSP_Request_Entity_Too_Large = 413 , NC_RTSP_Request_URI_Too_Long = 414 , NC_RTSP_Unsupported_Media_Type = 415 , NC_RTSP_Invalid_parameter = 451 , NC_RTSP_Illegal_Conference_Identifier = 452 , NC_RTSP_Not_Enough_Bandwidth = 453 , NC_RTSP_Session_Not_Found = 454 , NC_RTSP_Method_Not_Valid_In_This_State = 455 , NC_RTSP_Header_Field_Not_Valid = 456 , NC_RTSP_Invalid_Range = 457 , NC_RTSP_Parameter_Is_ReadOnly = 458 , NC_RTSP_Aggregate_Operation_Not_Allowed = 459 , NC_RTSP_Only_Aggregate_Operation_Allowed = 460 , NC_RTSP_Unsupported_Transport = 461 , NC_RTSP_Destination_Unreachable = 462 , NC_RTSP_Internal_Server_Error = 500 , NC_RTSP_Not_Implemented = 501 , NC_RTSP_Bad_Gateway = 502 , NC_RTSP_Service_Unavailable = 503 , NC_RTSP_Gateway_Timeout = 504 , NC_RTSP_RTSP_Version_Not_Supported = 505 , NC_RTSP_Option_not_support = 551 }

enum	{ GF_RTSP_STATE_INIT = 0 , GF_RTSP_STATE_WAITING , GF_RTSP_STATE_WAIT_FOR_CONTROL , GF_RTSP_STATE_INVALIDATED }

enum	{ GP_RTP_PCK_FORCE_MPEG4 = (1) , GP_RTP_PCK_USE_MULTI = (1<<1) , GP_RTP_PCK_USE_INTERLEAVING = (1<<2) , GP_RTP_PCK_USE_STATIC_ID = (1<<3) , GP_RTP_PCK_SIGNAL_RAP = (1<<4) , GP_RTP_PCK_SIGNAL_AU_IDX = (1<<5) , GP_RTP_PCK_SIGNAL_SIZE = (1<<6) , GP_RTP_PCK_SIGNAL_TS = (1<<7) , GP_RTP_PCK_SYSTEMS_CAROUSEL = (1<<8) , GP_RTP_PCK_USE_LATM_AAC = (1<<9) , GP_RTP_PCK_SELECTIVE_ENCRYPTION = (1<<10) , GP_RTP_PCK_KEY_IDX_PER_AU = (1<<11) , GP_RTP_DIMS_COMPRESSED = (1<<12) , GP_RTP_PCK_FORCE_STATIC_ID = (1<<13) }

enum	{ GF_RTP_PAYT_PCMU = 0 , GF_RTP_PAYT_GSM , GF_RTP_PAYT_G723 , GF_RTP_PAYT_DVI4_8K , GF_RTP_PAYT_DVI4_16K , GF_RTP_PAYT_LPC , GF_RTP_PAYT_PCMA , GF_RTP_PAYT_G722 , GF_RTP_PAYT_L16_STEREO , GF_RTP_PAYT_L16_MONO , GF_RTP_PAYT_QCELP_BASIC , GF_RTP_PAYT_CN , GF_RTP_PAYT_MPEG12_AUDIO , GF_RTP_PAYT_G728 , GF_RTP_PAYT_DVI4_11K , GF_RTP_PAYT_DVI4_22K , GF_RTP_PAYT_G729 , GF_RTP_PAYT_CelB = 25 , GF_RTP_PAYT_JPEG = 26 , GF_RTP_PAYT_nv = 28 , GF_RTP_PAYT_H261 = 31 , GF_RTP_PAYT_MPEG12_VIDEO = 32 , GF_RTP_PAYT_MP2T = 33 , GF_RTP_PAYT_H263 = 34 , GF_RTP_PAYT_LAST_STATIC_DEFINED = 35 , GF_RTP_PAYT_UNKNOWN = 128 , GF_RTP_PAYT_MPEG4 , GF_RTP_PAYT_AMR , GF_RTP_PAYT_AMR_WB , GF_RTP_PAYT_QCELP , GF_RTP_PAYT_EVRC_SMV , GF_RTP_PAYT_3GPP_TEXT , GF_RTP_PAYT_H264_AVC , GF_RTP_PAYT_LATM , GF_RTP_PAYT_AC3 , GF_RTP_PAYT_EAC3 , GF_RTP_PAYT_H264_SVC , GF_RTP_PAYT_HEVC , GF_RTP_PAYT_LHVC , GF_RTP_PAYT_VVC , GF_RTP_PAYT_OPUS }

enum	{ GF_RTP_NEW_AU = (1) , GF_RTP_AMR_ALIGN = (1<<1) , GF_RTP_M4V_CHECK_RAP = (1<<2) , GF_RTP_AVC_WAIT_RAP = (1<<4) , GF_RTP_HAS_ISMACRYP = (1<<5) , GF_RTP_ISMA_SEL_ENC = (1<<6) , GF_RTP_ISMA_HAS_KEY_IDX = (1<<7) , GF_RTP_AVC_USE_ANNEX_B = (1<<8) }

Functions
const char *	gf_rtsp_nc_to_string (u32 ErrCode)

GF_RTSPRange *	gf_rtsp_range_parse (char *range_buf)

GF_RTSPRange *	gf_rtsp_range_new ()

void	gf_rtsp_range_del (GF_RTSPRange *range)

GF_RTSPTransport *	gf_rtsp_transport_clone (GF_RTSPTransport *transp)

void	gf_rtsp_transport_del (GF_RTSPTransport *transp)

GF_RTSPCommand *	gf_rtsp_command_new ()

void	gf_rtsp_command_del (GF_RTSPCommand *com)

void	gf_rtsp_command_reset (GF_RTSPCommand *com)

GF_RTSPResponse *	gf_rtsp_response_new ()

void	gf_rtsp_response_del (GF_RTSPResponse *rsp)

void	gf_rtsp_response_reset (GF_RTSPResponse *rsp)

GF_RTSPSession *	gf_rtsp_session_new (char *sURL, u16 DefaultPort)

void	gf_rtsp_session_del (GF_RTSPSession *sess)

GF_Err	gf_rtsp_set_buffer_size (GF_RTSPSession *sess, u32 BufferSize)

u32	gf_rtsp_session_reset (GF_RTSPSession *sess, Bool ResetConnection)

Bool	gf_rtsp_is_my_session (GF_RTSPSession sess, char url)

const char *	gf_rtsp_get_server_name (GF_RTSPSession *sess)

const char *	gf_rtsp_get_user (GF_RTSPSession *sess)

const char *	gf_rtsp_get_password (GF_RTSPSession *sess)

u16	gf_rtsp_get_session_port (GF_RTSPSession *sess)

Bool	gf_rtsp_use_tls (GF_RTSPSession *sess)

GF_Err	gf_rtsp_get_response (GF_RTSPSession sess, GF_RTSPResponse rsp)

u32	gf_rtsp_get_session_state (GF_RTSPSession *sess)

void	gf_rtsp_reset_aggregation (GF_RTSPSession *sess)

GF_Err	gf_rtsp_send_command (GF_RTSPSession sess, GF_RTSPCommand com)

GF_Err	gf_rtsp_check_connection (GF_RTSPSession *sess)

GF_Err	gf_rtsp_set_interleave_callback (GF_RTSPSession *sess, gf_rtsp_interleave_callback SignalData)

GF_Err	gf_rtsp_session_read (GF_RTSPSession *sess)

GF_Err	gf_rtsp_register_interleave (GF_RTSPSession sess, void the_ch, u8 LowInterID, u8 HighInterID)

u32	gf_rtsp_unregister_interleave (GF_RTSPSession *sess, u8 LowInterID)

GF_RTSPSession *	gf_rtsp_session_new_server (GF_Socket rtsp_listener, Bool allow_http_tunnel, void ssl_ctx)

GF_Err	gf_rtsp_get_command (GF_RTSPSession sess, GF_RTSPCommand com)

char *	gf_rtsp_generate_session_id (GF_RTSPSession *sess)

GF_Err	gf_rtsp_send_response (GF_RTSPSession sess, GF_RTSPResponse rsp)

GF_Err	gf_rtsp_get_session_ip (GF_RTSPSession *sess, char buffer[GF_MAX_IP_NAME_LEN])

GF_Err	gf_rtsp_get_remote_address (GF_RTSPSession sess, char buffer)

GF_Err	gf_rtsp_session_write_interleaved (GF_RTSPSession sess, u32 idx, u8 pck, u32 pck_size)

const char *	gf_rtsp_get_session_cookie (GF_RTSPSession *sess)

GF_Err	gf_rtsp_merge_tunnel (GF_RTSPSession sess, GF_RTSPSession post_sess)

void	gf_rtsp_session_set_netcap_id (GF_RTSPSession sess, const char netcap_id)

GF_RTPChannel *	gf_rtp_new ()

GF_RTPChannel *	gf_rtp_new_ex (const char *netcap_id)

void	gf_rtp_del (GF_RTPChannel *ch)

GF_Err	gf_rtp_setup_transport (GF_RTPChannel ch, GF_RTSPTransport trans_info, const char *remote_address)

GF_Err	gf_rtp_set_ports (GF_RTPChannel *ch, u16 first_port)

GF_Err	gf_rtp_setup_payload (GF_RTPChannel *ch, u32 PayloadType, u32 ClockRate)

void	gf_rtp_enable_nat_keepalive (GF_RTPChannel *ch, u32 nat_timeout)

GF_Err	gf_rtp_initialize (GF_RTPChannel ch, u32 UDPBufferSize, Bool IsSource, u32 PathMTU, u32 ReorederingSize, u32 MaxReorderDelay, char local_interface_ip)

GF_Err	gf_rtp_stop (GF_RTPChannel *ch)

GF_Err	gf_rtp_set_ssm (GF_RTPChannel ch, const char src_ip_inc, u32 nb_src_ip_inc, const char *src_ip_exc, u32 nb_src_ip_exc)

GF_Err	gf_rtp_set_info_rtp (GF_RTPChannel *ch, u32 seq_num, u32 rtp_time, u32 ssrc)

Double	gf_rtp_get_current_time (GF_RTPChannel *ch)

void	gf_rtp_reset_buffers (GF_RTPChannel *ch)

void	gf_rtp_reset_ssrc (GF_RTPChannel *ch)

u32	gf_rtp_read_rtp (GF_RTPChannel ch, u8 buffer, u32 buffer_size)

u32	gf_rtp_flush_rtp (GF_RTPChannel ch, u8 buffer, u32 buffer_size)

u32	gf_rtp_read_rtcp (GF_RTPChannel ch, u8 buffer, u32 buffer_size)

u32	gf_rtp_read_flush (GF_RTPChannel ch, u8 buffer, u32 buffer_size)

GF_Err	gf_rtp_decode_rtp (GF_RTPChannel ch, u8 pck, u32 pck_size, GF_RTPHeader rtp_hdr, u32 PayloadStart)

GF_Err	gf_rtp_decode_rtcp (GF_RTPChannel ch, u8 pck, u32 pck_size, Bool *has_sr)

GF_Err	gf_rtp_send_rtcp_report (GF_RTPChannel *ch)

void	gf_rtp_set_loss_rate (GF_RTPChannel *ch, u32 loss_rate)

GF_Err	gf_rtp_send_bye (GF_RTPChannel *ch)

GF_Err	gf_rtp_send_packet (GF_RTPChannel ch, GF_RTPHeader rtp_hdr, u8 *pck, u32 pck_size, Bool fast_send)

GF_Err	gf_rtp_set_interleave_callbacks (GF_RTPChannel ch, gf_rtp_tcp_callback tcp_callback, void cbk1, void *cbk2)

u32	gf_rtp_is_unicast (GF_RTPChannel *ch)

u32	gf_rtp_is_interleaved (GF_RTPChannel *ch)

u32	gf_rtp_get_clockrate (GF_RTPChannel *ch)

u8	gf_rtp_get_low_interleave_id (GF_RTPChannel *ch)

u8	gf_rtp_get_hight_interleave_id (GF_RTPChannel *ch)

const GF_RTSPTransport *	gf_rtp_get_transport (GF_RTPChannel *ch)

Float	gf_rtp_get_loss (GF_RTPChannel *ch)

u32	gf_rtp_get_tcp_bytes_sent (GF_RTPChannel *ch)

void	gf_rtp_get_ports (GF_RTPChannel ch, u16 rtp_port, u16 *rtcp_port)

GF_SDPInfo *	gf_sdp_info_new ()

void	gf_sdp_info_del (GF_SDPInfo *sdp)

void	gf_sdp_info_reset (GF_SDPInfo *sdp)

GF_Err	gf_sdp_info_parse (GF_SDPInfo sdp, char sdp_text, u32 text_size)

GF_SDPMedia *	gf_sdp_media_new ()

void	gf_sdp_media_del (GF_SDPMedia *media)

GF_SDPConnection *	gf_sdp_conn_new ()

void	gf_sdp_conn_del (GF_SDPConnection *conn)

GF_SDP_FMTP *	gf_sdp_fmtp_new ()

void	gf_sdp_fmtp_del (GF_SDP_FMTP *fmtp)

GP_RTPPacketizer *	gf_rtp_builder_new (u32 rtp_payt, GF_SLConfig slc, u32 flags, void cbk_obj, void(OnNewPacket)(void cbk, GF_RTPHeader header), void(OnPacketDone)(void cbk, GF_RTPHeader header), void(OnDataReference)(void cbk, u32 payload_size, u32 offset_from_orig), void(OnData)(void cbk, u8 *data, u32 data_size, Bool is_head))

void	gf_rtp_builder_del (GP_RTPPacketizer *builder)

void	gf_rtp_builder_init (GP_RTPPacketizer builder, u8 PayloadType, u32 MaxPayloadSize, u32 max_ptime, u32 StreamType, u32 codecid, u32 PL_ID, u32 avgSize, u32 maxSize, u32 avgTS, u32 maxDTS, u32 IV_length, u32 KI_length, char pref_mode)

void	gf_rtp_builder_set_cryp_info (GP_RTPPacketizer builder, u64 IV, char key_indicator, Bool is_encrypted)

GF_Err	gf_rtp_builder_process (GP_RTPPacketizer builder, u8 data, u32 data_size, u8 IsAUEnd, u32 FullAUSize, u32 duration, u8 descIndex)

GF_Err	gf_rtp_builder_format_sdp (GP_RTPPacketizer builder, char payload_name, char *out_sdp_line, char dsi, u32 dsi_size)

Bool	gf_rtp_builder_get_payload_name (GP_RTPPacketizer *builder, char payload_name[20], char media_name[20])

GF_RTPDepacketizer *	gf_rtp_depacketizer_new (GF_SDPMedia media, u32 hdr_payt, gf_rtp_packet_cbk sl_packet_cbk, void udta)

void	gf_rtp_depacketizer_del (GF_RTPDepacketizer *rtpd)

void	gf_rtp_depacketizer_reset (GF_RTPDepacketizer *rtpd, Bool full_reset)

void	gf_rtp_depacketizer_process (GF_RTPDepacketizer rtpd, GF_RTPHeader hdr, u8 *payload, u32 size)

Detailed Description

This section documents the tools used for real-time streaming over IP using RTP/RTCP/RTSP/SDP.

Data Structure Documentation

◆ GF_RTSPRange

struct GF_RTSPRange

RTSP Range information RTSP Session level only, although this is almost the same format as an SDP range, this is not used in the SDP lib as "a=range" is not part of SDP but part of RTSP

Data Fields
Double	start
Double	end
u32	UseSMPTE
Double	FPS

◆ GF_RTSPTransport

struct GF_RTSPTransport

RTSP transport structure

Data Fields
Bool	IsUnicast
char *	destination
char *	source
Bool	IsRecord
Bool	Append
Bool	IsInterleaved
u32	rtpID
u32	rtcpID
u32	MulticastLayers
u8	TTL
u16	port_first
u16	port_last
u16	client_port_first
u16	client_port_last
u32	SSRC
char *	Profile
Bool	is_sender

◆ GF_RTSPCommand

struct GF_RTSPCommand

RTSP command structure

Collaboration diagram for GF_RTSPCommand:

Data Fields
char *	Accept
char *	Accept_Encoding
char *	Accept_Language
char *	Authorization
u32	Bandwidth
u32	Blocksize
char *	Cache_Control
char *	Conference
char *	Connection
u32	Content_Length
u32	CSeq
char *	From
char *	Proxy_Authorization
char *	Proxy_Require
GF_RTSPRange *	Range
char *	Referer
Double	Scale
char *	Session
Double	Speed
GF_List *	Transports
char *	User_Agent
char *	method
GF_List *	Xtensions
char *	body
char *	ControlString
void *	user_data
u32	user_flags
char *	service_name
u32	StatusCode
Bool	is_resend

◆ GF_RTPInfo

struct GF_RTPInfo

RTP-Info for RTP channels.

There may be several RTP-Infos in one response
based on the server implementation (DSS/QTSS begaves this way)

Data Fields
char *	url
u32	seq
u32	rtp_time
u32	ssrc

◆ GF_RTSPResponse

struct GF_RTSPResponse

RTSP Response

Collaboration diagram for GF_RTSPResponse:

Data Fields
u32	ResponseCode
char *	ResponseInfo
char *	Accept
char *	Accept_Encoding
char *	Accept_Language
char *	Allow
char *	Authorization
u32	Bandwidth
u32	Blocksize
char *	Cache_Control
char *	Conference
char *	Connection
char *	Content_Base
char *	Content_Encoding
char *	Content_Language
u32	Content_Length
char *	Content_Location
char *	Content_Type
u32	CSeq
char *	Date
char *	Expires
char *	From
char *	Host
char *	If_Match
char *	If_Modified_Since
char *	Last_Modified
char *	Location
char *	Proxy_Authenticate
char *	Proxy_Require
char *	Public
GF_RTSPRange *	Range
char *	Referer
char *	Require
char *	Retry_After
GF_List *	RTP_Infos
Double	Scale
char *	Server
char *	Session
u32	SessionTimeOut
Double	Speed
u32	StreamID
char *	Timestamp
GF_List *	Transports
char *	Unsupported
char *	User_Agent
char *	Vary
char *	Via
char *	WWW_Authenticate
GF_List *	Xtensions
char *	body

◆ GF_RTPHeader

struct GF_RTPHeader

RTP header

Data Fields
u8	Version
u8	Padding
u8	Extension
u8	CSRCCount
u8	Marker
u8	PayloadType
u16	SequenceNumber
u32	TimeStamp
u32	SSRC
u32	CSRC[16]
u64	recomputed_ntp_ts

◆ GF_RTPMap

struct GF_RTPMap

RTPMap information

Data Fields
u32	PayloadType
char *	payload_name
u32	ClockRate
u32	AudioChannels

◆ GF_X_Attribute

struct GF_X_Attribute

Extension Attribute

All attributes x-ZZZZ are considered as extensions attributes. If no "x-" is found
the attributes in the RTSP response is SKIPPED. The "x-" radical is removed in the structure
when parsing commands / responses

Data Fields
char *	Name
char *	Value

◆ GF_SDPBandwidth

struct GF_SDPBandwidth

SDP bandwidth info

Data Fields
char *	name
u32	value

◆ GF_SDPTiming

struct GF_SDPTiming

SDP Timing information

Data Fields
u32	StartTime
u32	StopTime
u32	RepeatInterval
u32	ActiveDuration
u32	OffsetFromStart[GF_SDP_MAX_TIMEOFFSET]
u32	NbRepeatOffsets
u32	AdjustmentTime[GF_SDP_MAX_TIMEOFFSET]
s32	AdjustmentOffset[GF_SDP_MAX_TIMEOFFSET]
u32	NbZoneOffsets

◆ GF_SDPConnection

struct GF_SDPConnection

SDP Connection information

Data Fields
char *	net_type
char *	add_type
char *	host
s32	TTL
u32	add_count

◆ GF_SDP_FMTP

struct GF_SDP_FMTP

SDP FormatTypePayload

description of dynamic payload types. This is opaque at the SDP level.
Each attributes is assumed to be formatted as <param_name=param_val; ...>
If not the case the attribute will have an empty value string and only the
parameter name.

Data Fields
u32	PayloadType
GF_List *	Attributes

◆ GF_SDPMedia

struct GF_SDPMedia

SDP Media information

Data Fields
u32	Type
u32	PortNumber
u32	NumPorts
char *	Profile
GF_List *	Connections
GF_List *	RTPMaps
GF_List *	FMTP
char *	fmt_list
GF_List *	Attributes
char *	k_method
char *	k_key
GF_List *	Bandwidths
u32	PacketTime
u32	SendReceive
char *	orientation
char *	sdplang
char *	lang
Double	FrameRate
s32	Quality

◆ GF_SDPInfo

struct GF_SDPInfo

SDP information

Collaboration diagram for GF_SDPInfo:

Data Fields
u32	Version
char *	o_username
char *	o_session_id
char *	o_version
char *	o_address
char *	o_net_type
char *	o_add_type
char *	s_session_name
char *	i_description
char *	u_uri
char *	e_email
char *	p_phone
GF_SDPConnection *	c_connection
GF_List *	b_bandwidth
GF_List *	Timing
char *	k_method
char *	k_key
char *	a_cat
char *	a_keywds
char *	a_tool
u32	a_SendReceive
char *	a_type
char *	a_charset
char *	a_sdplang
char *	a_lang
GF_List *	Attributes
GF_List *	media_desc

◆ GP_RTPSLMap

struct GP_RTPSLMap

Mapping between RTP and GPAC / MPEG-4 Systems SyncLayer

Data Fields
char	mode[30]
u8 *	config
u32	configSize
u8	config_updated
u8	StreamType
u32	PL_ID
u16	rvc_predef
u8 *	rvc_config
u32	rvc_config_size
u32	ConstantSize
u32	ConstantDuration
u32	CodecID
u32	maxDisplacement
u32	deinterleaveBufferSize
u32	SizeLength
u32	IndexLength
u32	IndexDeltaLength
u32	DTSDeltaLength
u32	CTSDeltaLength
Bool	RandomAccessIndication
u32	StreamStateIndication
u32	AuxiliaryDataSizeLength
u8	IV_length
u8	IV_delta_length
u8	KI_length
u32	auh_first_min_len
u32	auh_min_len

◆ GF_RTPStaticMap

struct GF_RTPStaticMap

Static RTP map definition, for static payload types

Data Fields
u32	fmt
u32	clock_rate
u32	stream_type
u32	codec_id
const char *	mime

Macro Definition Documentation

◆ GF_RTSP_VERSION

#define GF_RTSP_VERSION "RTSP/1.0"

RTSP version supported by GPAC

◆ GF_RTSP_PROFILE_RTP_AVP

#define GF_RTSP_PROFILE_RTP_AVP "RTP/AVP"

RTSP AVP Transport Profile

◆ GF_RTSP_PROFILE_RTP_AVP_TCP

#define GF_RTSP_PROFILE_RTP_AVP_TCP "RTP/AVP/TCP"

RTSP AVP + TCP Transport Profile

◆ GF_RTSP_PROFILE_UDP

#define GF_RTSP_PROFILE_UDP "udp"

RTSP UDP Transport Profile

◆ GF_RTSP_DESCRIBE

#define GF_RTSP_DESCRIBE "DESCRIBE"

RTSP DESCRIBE method

◆ GF_RTSP_SETUP

#define GF_RTSP_SETUP "SETUP"

RTSP SETUP method

◆ GF_RTSP_PLAY

#define GF_RTSP_PLAY "PLAY"

RTSP PLAY method

◆ GF_RTSP_PAUSE

#define GF_RTSP_PAUSE "PAUSE"

RTSP PAUSE method

◆ GF_RTSP_RECORD

#define GF_RTSP_RECORD "RECORD"

RTSP RECORD method

◆ GF_RTSP_TEARDOWN

#define GF_RTSP_TEARDOWN "TEARDOWN"

RTSP TEARDOWN method

◆ GF_RTSP_GET_PARAMETER

#define GF_RTSP_GET_PARAMETER "GET_PARAMETER"

RTSP GET_PARAMETER method

◆ GF_RTSP_SET_PARAMETER

#define GF_RTSP_SET_PARAMETER "SET_PARAMETER"

RTSP SET_PARAMETER method

◆ GF_RTSP_OPTIONS

#define GF_RTSP_OPTIONS "OPTIONS"

RTSP OPTIONS method

◆ GF_RTSP_ANNOUNCE

#define GF_RTSP_ANNOUNCE "ANNOUNCE"

RTSP ANNOUCE method

◆ GF_RTSP_REDIRECT

#define GF_RTSP_REDIRECT "REDIRECT"

RTSP REDIRECT method

◆ GF_RTSP_TUNNEL_POST

#define GF_RTSP_TUNNEL_POST -1000

special error code for gf_rtsp_get_command

◆ GF_SDP_MAX_TIMEOFFSET

#define GF_SDP_MAX_TIMEOFFSET 10

SDP maximum time offset We do not support more than this offset / zone adjustment if more are needed, RFC recommends to use several entries rather than a big offset

Typedef Documentation

◆ gf_rtsp_interleave_callback

typedef GF_Err(* gf_rtsp_interleave_callback) (GF_RTSPSession *sess, void *cbk_ptr, u8 *buffer, u32 bufferSize, Bool IsRTCP)

callback function for interleaved RTSP/TCP transport

Parameters

sess	the target RTSP session
cbk_ptr	opaque data
buffer	RTP or RTCP packet
bufferSize	packet size in bytes
IsRTCP	set to GF_TRUE if the packet is an RTCP packet, GF_FALSE otherwise

Returns: error if any

◆ gf_rtp_tcp_callback

typedef GF_Err(* gf_rtp_tcp_callback) (void *cbk1, void *cbk2, Bool is_rtcp, u8 *pck, u32 pck_size)

callback used for writing rtp over TCP

Parameters

cbk1	opaque user data
cbk2	opaque user data
is_rtcp	indicates the data is an RTCP packet
pck	the RTP/RTCP buffer
pck_size	the RTP/RTCP size

Returns: error if any

◆ gf_rtp_packet_cbk

typedef void(* gf_rtp_packet_cbk) (void *udta, u8 *payload, u32 size, GF_SLHeader *hdr, GF_Err e)

RTP depacketizer callback

Parameters

udta	opaque user data
payload	depacketized payload data
size	payload size
hdr	MPEG-4 Sync Layer structure for the payload (with AU start/end flag, timestamps, etc)
e	error if any

Enumeration Type Documentation

◆ anonymous enum

anonymous enum

RTSP NOTIF CODES

Enumerator
NC_RTSP_Continue
NC_RTSP_OK
NC_RTSP_Created
NC_RTSP_Low_on_Storage_Space
NC_RTSP_Multiple_Choice
NC_RTSP_Moved_Permanently
NC_RTSP_Moved_Temporarily
NC_RTSP_See_Other
NC_RTSP_Use_Proxy
NC_RTSP_Bad_Request
NC_RTSP_Unauthorized
NC_RTSP_Payment_Required
NC_RTSP_Forbidden
NC_RTSP_Not_Found
NC_RTSP_Method_Not_Allowed
NC_RTSP_Not_Acceptable
NC_RTSP_Proxy_Authentication_Required
NC_RTSP_Request_Timeout
NC_RTSP_Gone
NC_RTSP_Length_Required
NC_RTSP_Precondition_Failed
NC_RTSP_Request_Entity_Too_Large
NC_RTSP_Request_URI_Too_Long
NC_RTSP_Unsupported_Media_Type
NC_RTSP_Invalid_parameter
NC_RTSP_Illegal_Conference_Identifier
NC_RTSP_Not_Enough_Bandwidth
NC_RTSP_Session_Not_Found
NC_RTSP_Method_Not_Valid_In_This_State
NC_RTSP_Header_Field_Not_Valid
NC_RTSP_Invalid_Range
NC_RTSP_Parameter_Is_ReadOnly
NC_RTSP_Aggregate_Operation_Not_Allowed
NC_RTSP_Only_Aggregate_Operation_Allowed
NC_RTSP_Unsupported_Transport
NC_RTSP_Destination_Unreachable
NC_RTSP_Internal_Server_Error
NC_RTSP_Not_Implemented
NC_RTSP_Bad_Gateway
NC_RTSP_Service_Unavailable
NC_RTSP_Gateway_Timeout
NC_RTSP_RTSP_Version_Not_Supported
NC_RTSP_Option_not_support

◆ anonymous enum

anonymous enum

RTSP States. The only non blocking mode is GF_RTSP_STATE_WAIT_FOR_CONTROL

Enumerator
GF_RTSP_STATE_INIT	Initialized (connection might be off, but all structures are in place) This is the default state between # requests (aka, DESCRIBE and SETUP or SETUP and PLAY ...)
GF_RTSP_STATE_WAITING	Waiting
GF_RTSP_STATE_WAIT_FOR_CONTROL	PLAY, PAUSE, RECORD. Aggregation is allowed for the same type, you can send several command in a row. However the session will return GF_SERVICE_ERROR if you do not have a valid SessionID in the command You cannot issue a SETUP / DESCRIBE while in this state
GF_RTSP_STATE_INVALIDATED	FATAL ERROR: session is invalidated by server. Call reset and restart from SETUP if needed

◆ anonymous enum

anonymous enum

packetizer config flags - some flags are dynamically re-assigned when detecting multiSL / B-Frames / ...

Enumerator
GP_RTP_PCK_FORCE_MPEG4
GP_RTP_PCK_USE_MULTI
GP_RTP_PCK_USE_INTERLEAVING
GP_RTP_PCK_USE_STATIC_ID
GP_RTP_PCK_SIGNAL_RAP
GP_RTP_PCK_SIGNAL_AU_IDX
GP_RTP_PCK_SIGNAL_SIZE
GP_RTP_PCK_SIGNAL_TS
GP_RTP_PCK_SYSTEMS_CAROUSEL
GP_RTP_PCK_USE_LATM_AAC
GP_RTP_PCK_SELECTIVE_ENCRYPTION
GP_RTP_PCK_KEY_IDX_PER_AU
GP_RTP_DIMS_COMPRESSED
GP_RTP_PCK_FORCE_STATIC_ID

◆ anonymous enum

anonymous enum

Supported payload types

Enumerator
GF_RTP_PAYT_PCMU
GF_RTP_PAYT_GSM
GF_RTP_PAYT_G723
GF_RTP_PAYT_DVI4_8K
GF_RTP_PAYT_DVI4_16K
GF_RTP_PAYT_LPC
GF_RTP_PAYT_PCMA
GF_RTP_PAYT_G722
GF_RTP_PAYT_L16_STEREO
GF_RTP_PAYT_L16_MONO
GF_RTP_PAYT_QCELP_BASIC
GF_RTP_PAYT_CN
GF_RTP_PAYT_MPEG12_AUDIO
GF_RTP_PAYT_G728
GF_RTP_PAYT_DVI4_11K
GF_RTP_PAYT_DVI4_22K
GF_RTP_PAYT_G729
GF_RTP_PAYT_CelB
GF_RTP_PAYT_JPEG
GF_RTP_PAYT_nv
GF_RTP_PAYT_H261
GF_RTP_PAYT_MPEG12_VIDEO
GF_RTP_PAYT_MP2T
GF_RTP_PAYT_H263
GF_RTP_PAYT_LAST_STATIC_DEFINED
GF_RTP_PAYT_UNKNOWN
GF_RTP_PAYT_MPEG4
GF_RTP_PAYT_AMR
GF_RTP_PAYT_AMR_WB
GF_RTP_PAYT_QCELP
GF_RTP_PAYT_EVRC_SMV
GF_RTP_PAYT_3GPP_TEXT
GF_RTP_PAYT_H264_AVC
GF_RTP_PAYT_LATM
GF_RTP_PAYT_AC3
GF_RTP_PAYT_EAC3
GF_RTP_PAYT_H264_SVC
GF_RTP_PAYT_HEVC
GF_RTP_PAYT_LHVC
GF_RTP_PAYT_VVC
GF_RTP_PAYT_OPUS

◆ anonymous enum

anonymous enum

rtp payload flags

Enumerator
GF_RTP_NEW_AU
GF_RTP_AMR_ALIGN
GF_RTP_M4V_CHECK_RAP
GF_RTP_AVC_WAIT_RAP
GF_RTP_HAS_ISMACRYP
GF_RTP_ISMA_SEL_ENC
GF_RTP_ISMA_HAS_KEY_IDX
GF_RTP_AVC_USE_ANNEX_B

Function Documentation

◆ gf_rtsp_nc_to_string()

const char* gf_rtsp_nc_to_string ( u32 ErrCode )

Gives string description of error code

Parameters

ErrCode the RTSP error code

Returns: the description of the RTSP error code

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◆ gf_rtsp_range_parse()

GF_RTSPRange* gf_rtsp_range_parse ( char * range_buf )

parses a Range line and returns range header structure. This can be used for RTSP extension of SDP

Note: Only support for npt for now

Parameters

range_buf the range string

Returns: a newly allocated RTSP range

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◆ gf_rtsp_range_new()

GF_RTSPRange* gf_rtsp_range_new ( )

creates a new RTSP range

Returns: a newly allocated RTSP range

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◆ gf_rtsp_range_del()

void gf_rtsp_range_del ( GF_RTSPRange * range )

destroys a RTSP range

Parameters

range the target RTSP range

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◆ gf_rtsp_transport_clone()

GF_RTSPTransport* gf_rtsp_transport_clone ( GF_RTSPTransport * transp )

clones a RTSP transport

Parameters

transp the target RTSP transport

Returns: an allocated copy RTSP transport

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◆ gf_rtsp_transport_del()

void gf_rtsp_transport_del ( GF_RTSPTransport * transp )

destroys a RTSP transport

Parameters

transp the target RTSP transport

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◆ gf_rtsp_command_new()

GF_RTSPCommand* gf_rtsp_command_new ( )

creates an RTSP command

Returns: the newly allocated RTSP command

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◆ gf_rtsp_command_del()

void gf_rtsp_command_del ( GF_RTSPCommand * com )

destroys an RTSP command

Parameters

com	the target RTSP command

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◆ gf_rtsp_command_reset()

void gf_rtsp_command_reset ( GF_RTSPCommand * com )

resets an RTSP command

Parameters

com	the target RTSP command

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◆ gf_rtsp_response_new()

GF_RTSPResponse* gf_rtsp_response_new ( )

creates an RTSP response

Returns: the newly allocated RTSP response

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◆ gf_rtsp_response_del()

void gf_rtsp_response_del ( GF_RTSPResponse * rsp )

deletes an RTSP response

Parameters

rsp	the target RTSP response

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◆ gf_rtsp_response_reset()

void gf_rtsp_response_reset ( GF_RTSPResponse * rsp )

resets an RTSP response

Parameters

rsp	the target RTSP response

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◆ gf_rtsp_session_new()

GF_RTSPSession* gf_rtsp_session_new	(	char *	sURL,
		u16	DefaultPort
	)

creates a new RTSP session

Parameters

sURL	the target RTSP session URL
DefaultPort	the target RTSP session port

Returns: a newly allocated RTSP session

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◆ gf_rtsp_session_del()

void gf_rtsp_session_del ( GF_RTSPSession * sess )

destroys an RTSP session

Parameters

sess	the target RTSP session

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◆ gf_rtsp_set_buffer_size()

GF_Err gf_rtsp_set_buffer_size	(	GF_RTSPSession *	sess,
		u32	BufferSize
	)

sets TCP buffer size of an RTSP session

Parameters

sess	the target RTSP session
BufferSize	desired buffer size in bytes

Returns: error if any

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◆ gf_rtsp_session_reset()

u32 gf_rtsp_session_reset	(	GF_RTSPSession *	sess,
		Bool	ResetConnection
	)

resets state machine, invalidate SessionID

Note: RFC2326 requires that the session is reseted when all RTP streams are closed. As this lib doesn't maintain the number of valid streams you MUST call reset when all your streams are shutdown (either requested through TEARDOWN or signaled through RTCP BYE packets for RTP, or any other signaling means for other protocols)

Parameters

sess	the target RTSP session
ResetConnection	if set, this will destroy the associated TCP socket. This is useful in case of timeouts, because some servers do not restart with the right CSeq.

Returns: number of retries for reset happening before first server reply, 0 otherwise

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◆ gf_rtsp_is_my_session()

Bool gf_rtsp_is_my_session	(	GF_RTSPSession *	sess,
		char *	url
	)

checks if an RTSP session matches an RTSP URL

Parameters

sess	the target RTSP session
url	the URL to test

Returns: GF_TRUE if the session matches the URL, GF_FALSE otherwise

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◆ gf_rtsp_get_server_name()

const char* gf_rtsp_get_server_name ( GF_RTSPSession * sess )

gets server name of an RTSP session

Parameters

sess	the target RTSP session

Returns: the server name

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◆ gf_rtsp_get_user()

const char* gf_rtsp_get_user ( GF_RTSPSession * sess )

gets user name of an RTSP session

Parameters

sess	the target RTSP session

Returns: the user name or NULL if none

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◆ gf_rtsp_get_password()

const char* gf_rtsp_get_password ( GF_RTSPSession * sess )

gets password of an RTSP session

Parameters

sess	the target RTSP session

Returns: the password or NULL if none

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◆ gf_rtsp_get_session_port()

u16 gf_rtsp_get_session_port ( GF_RTSPSession * sess )

gets server port of an RTSP session

Parameters

sess	the target RTSP session

Returns: the server port

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◆ gf_rtsp_use_tls()

Bool gf_rtsp_use_tls ( GF_RTSPSession * sess )

checks if RTSP connection is over TLS

Parameters

sess	the target RTSP session

Returns: GF_TRUE if connection is secured

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◆ gf_rtsp_get_response()

GF_Err gf_rtsp_get_response	(	GF_RTSPSession *	sess,
		GF_RTSPResponse *	rsp
	)

fetches an RTSP response from the server. the GF_RTSPResponse will be reseted before fetch

Parameters

sess	the target RTSP session
rsp	the response object to fill with the response. This will be reseted before TCP fetch

Returns: error if any

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◆ gf_rtsp_get_session_state()

u32 gf_rtsp_get_session_state ( GF_RTSPSession * sess )

gets the RTSP session state

Parameters

sess	the target RTSP session

Returns: the session state

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◆ gf_rtsp_reset_aggregation()

void gf_rtsp_reset_aggregation ( GF_RTSPSession * sess )

forces a reset of the state to GF_RTSP_STATE_INIT

Parameters

sess	the target RTSP session

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◆ gf_rtsp_send_command()

GF_Err gf_rtsp_send_command	(	GF_RTSPSession *	sess,
		GF_RTSPCommand *	com
	)

sends an RTSP request to the server.

Parameters

sess	the target RTSP session
com	the RTSP command to send

Returns: error if any

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◆ gf_rtsp_check_connection()

GF_Err gf_rtsp_check_connection ( GF_RTSPSession * sess )

checks connection status - should be called before processing any RTSP for non-blocking IO

Parameters

sess	the target RTSP session

Returns: GF_IP_NETWORK_EMPTY if connection is still pending or data cannot be flushed, GF_OK if connected and no more data to send or error if any

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◆ gf_rtsp_set_interleave_callback()

GF_Err gf_rtsp_set_interleave_callback	(	GF_RTSPSession *	sess,
		gf_rtsp_interleave_callback	SignalData
	)

assigns the callback function for interleaved RTSP/TCP transport

Parameters

sess	the target RTSP session
SignalData	the callback function on each interleaved packet

Returns: error if any

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◆ gf_rtsp_session_read()

GF_Err gf_rtsp_session_read ( GF_RTSPSession * sess )

reads RTSP session (response fetch and interleaved RTSP/TCP transport)

Parameters

sess	the target RTSP session

Returns: error if any

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◆ gf_rtsp_register_interleave()

GF_Err gf_rtsp_register_interleave	(	GF_RTSPSession *	sess,
		void *	the_ch,
		u8	LowInterID,
		u8	HighInterID
	)

registers a new interleaved RTP channel over an RTSP connection

Parameters

sess	the target RTSP session
the_ch	opaque data passed to gf_rtsp_interleave_callback
LowInterID	ID of the RTP interleave channel
HighInterID	ID of the RCTP interleave channel

Returns: error if any

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◆ gf_rtsp_unregister_interleave()

u32 gf_rtsp_unregister_interleave	(	GF_RTSPSession *	sess,
		u8	LowInterID
	)

unregisters a new interleaved RTP channel over an RTSP connection

Parameters

sess	the target RTSP session
LowInterID	ID of the RTP interleave channel

Returns: the numbers of registered interleaved channels remaining

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◆ gf_rtsp_session_new_server()

GF_RTSPSession* gf_rtsp_session_new_server	(	GF_Socket *	rtsp_listener,
		Bool	allow_http_tunnel,
		void *	ssl_ctx
	)

creates a new RTSP session from an existing socket in listen state. If no pending connection is detected, return NULL

Parameters

rtsp_listener	the listening server socket
allow_http_tunnel	indicate if HTTP tunnel should be enabled
ssl_ctx	OpenSSL context

Returns: the newly allocated RTSP session if any, NULL otherwise

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◆ gf_rtsp_get_command()

GF_Err gf_rtsp_get_command	(	GF_RTSPSession *	sess,
		GF_RTSPCommand *	com
	)

fetches an RTSP request

Parameters

sess	the target RTSP session
com	the RTSP command to fill with the command. This will be reseted before fetch

Returns: error if any or GF_RTSP_TUNNEL_POST if this is a POST on a HTPP tunnel

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◆ gf_rtsp_generate_session_id()

char* gf_rtsp_generate_session_id ( GF_RTSPSession * sess )

generates a session ID for the given session

Parameters

sess	the target RTSP session

Returns: an allocated string containing a session ID

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◆ gf_rtsp_send_response()

GF_Err gf_rtsp_send_response	(	GF_RTSPSession *	sess,
		GF_RTSPResponse *	rsp
	)

sends an RTSP response

Parameters

sess	the target RTSP session
rsp	the response to send

Returns: error if any

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◆ gf_rtsp_get_session_ip()

GF_Err gf_rtsp_get_session_ip	(	GF_RTSPSession *	sess,
		char	buffer[GF_MAX_IP_NAME_LEN]
	)

gets the IP address of the local host running the session

Parameters

sess	the target RTSP session
buffer	buffer to store the local host name

Returns: error if any

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◆ gf_rtsp_get_remote_address()

GF_Err gf_rtsp_get_remote_address	(	GF_RTSPSession *	sess,
		char *	buffer
	)

gets the IP address of the connected peer

Parameters

sess	the target RTSP session
buffer	buffer to store the connected peer name

Returns: error if any

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◆ gf_rtsp_session_write_interleaved()

GF_Err gf_rtsp_session_write_interleaved	(	GF_RTSPSession *	sess,
		u32	idx,
		u8 *	pck,
		u32	pck_size
	)

writes a packet on an interleaved channel over RTSP

Parameters

sess	the target RTSP session
idx	ID (RTP or RTCP) of the interleaved channel
pck	packet data (RTP or RTCP packet) to write
pck_size	packet size in bytes

Returns: error if any

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◆ gf_rtsp_get_session_cookie()

const char* gf_rtsp_get_session_cookie ( GF_RTSPSession * sess )

gets sessioncookie for HTTP tunnel

Parameters

sess	the target RTSP session

Returns: cookie or NULL if none

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◆ gf_rtsp_merge_tunnel()

GF_Err gf_rtsp_merge_tunnel	(	GF_RTSPSession *	sess,
		GF_RTSPSession *	post_sess
	)

move TCP connection of a POST HTTP tunnel link to main session

Parameters

sess	the target RTSP session
post_sess	the target RTSP POST http tunnel session - the session is not destroyed, only its connection is detached

Returns: error if any

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◆ gf_rtsp_session_set_netcap_id()

void gf_rtsp_session_set_netcap_id	(	GF_RTSPSession *	sess,
		const char *	netcap_id
	)

sets associated netcap rules

Parameters

sess	the target RTSP session
netcap_id	ID of netcap configuration to use, may be null (see gpac -h netcap)

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◆ gf_rtp_new()

GF_RTPChannel* gf_rtp_new ( )

creates a new RTP channel

Returns: a newly allocated RTP channel

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◆ gf_rtp_new_ex()

GF_RTPChannel* gf_rtp_new_ex ( const char * netcap_id )

creates a new RTP channel

Parameters

netcap_id ID of netcap configuration to use, may be null (see gpac -h netcap)

Returns: a newly allocated RTP channel

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◆ gf_rtp_del()

void gf_rtp_del ( GF_RTPChannel * ch )

destroys an RTP channel

Parameters

ch	the target RTP channel

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◆ gf_rtp_setup_transport()

GF_Err gf_rtp_setup_transport	(	GF_RTPChannel *	ch,
		GF_RTSPTransport *	trans_info,
		const char *	remote_address
	)

setup transport for an RTP channel A server channelis configured through the transport structure, with the same info as a client channel, the client_port_* info designing the REMOTE client and port_* designing the server channel

Parameters

ch	the target RTP channel
trans_info	the transport info for this channel
remote_address	the remote / destination address of the channel

Returns: error if any

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◆ gf_rtp_set_ports()

GF_Err gf_rtp_set_ports	(	GF_RTPChannel *	ch,
		u16	first_port
	)

setup of rtp/rtcp transport ports This only applies in unicast, non interleaved cases. For multicast port setup MUST be done through the above gf_rtp_setup_transport function This will take care of port reuse

Parameters

ch	the target RTP channel
first_port	RTP port number of the RTP channel

Returns: error if any

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◆ gf_rtp_setup_payload()

GF_Err gf_rtp_setup_payload	(	GF_RTPChannel *	ch,
		u32	PayloadType,
		u32	ClockRate
	)

init of RTP payload information. Only ONE payload per sync source is supported in this version of the library (a sender cannot switch payload types on a single media)

Parameters

ch	the target RTP channel
PayloadType	identifier of RTP payload
ClockRate	clock rate in (1/Hz) of RTP channel

Returns: error if any

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◆ gf_rtp_enable_nat_keepalive()

void gf_rtp_enable_nat_keepalive	(	GF_RTPChannel *	ch,
		u32	nat_timeout
	)

enables sending of NAT keep-alive packets for NAT traversal

Parameters

ch	the target RTP channel
nat_timeout	specifies the inactivity period in ms after which NAT keepalive packets are sent. If 0, disables NAT keep-alive packets

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◆ gf_rtp_initialize()

GF_Err gf_rtp_initialize	(	GF_RTPChannel *	ch,
		u32	UDPBufferSize,
		Bool	IsSource,
		u32	PathMTU,
		u32	ReorederingSize,
		u32	MaxReorderDelay,
		char *	local_interface_ip
	)

initializes the RTP channel.

Parameters

ch	the target RTP channel
UDPBufferSize	UDP stack buffer size if configurable by OS/ISP - ignored otherwise

Note

On WinCE devices, this is not configurable on an app bases but for the whole OS you must update the device registry with:

[HKEY_LOCAL_MACHINE\Comm\Afd]

DgramBuffer=dword:N

N

@ N

Definition: uni_bidi.c:68

    where N is the number of UDP datagrams a socket should be able to buffer. For multimedia

app you should set N as large as possible. The device MUST be reseted for the param to take effect

Parameters

IsSource	if true, the channel is a sender (media data, sender report, Receiver report processing) if source, you must specify the Path MTU size. The RTP lib won't send any packet bigger than this size your application shall perform payload size splitting if needed
PathMTU	desired path MTU (max packet size) in bytes
ReorederingSize	max number of packets to queue for reordering. 0 means no reordering
MaxReorderDelay	max time to wait in ms before releasing first packet in reoderer when only one packet is present. If 0 and reordering size is specified, defaults to 200 ms (usually enough).
local_interface_ip	local interface address to use for multicast. If NULL, default address is used

Returns: error if any

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◆ gf_rtp_stop()

GF_Err gf_rtp_stop ( GF_RTPChannel * ch )

stops the RTP channel. This destrpys RTP and RTCP sockets as well as packet reorderer

Parameters

ch	the target RTP channel

Returns: error if any

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◆ gf_rtp_set_ssm()

GF_Err gf_rtp_set_ssm	(	GF_RTPChannel *	ch,
		const char **	src_ip_inc,
		u32	nb_src_ip_inc,
		const char **	src_ip_exc,
		u32	nb_src_ip_exc
	)

sets source-specific IPs - this MUST be called before calling gf_rtp_initialize

Parameters

ch	the target RTP channel
src_ip_inc	IP of sources to receive from
nb_src_ip_inc	number of sources to receive from
src_ip_exc	IP of sources to exclude
nb_src_ip_exc	number of sources to exclude

Returns: error if any

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◆ gf_rtp_set_info_rtp()

GF_Err gf_rtp_set_info_rtp	(	GF_RTPChannel *	ch,
		u32	seq_num,
		u32	rtp_time,
		u32	ssrc
	)

inits the RTP info after a PLAY or PAUSE, rtp_time is the rtp TimeStamp of the RTP packet with seq_num sequence number. This info is needed to compute the CurrentTime of the RTP channel ssrc may not be known if sender hasn't indicated it (use 0 then)

Parameters

ch	the target RTP channel
seq_num	the seq num of the next packet to be received
rtp_time	the time in RTP timescale of the next packet to be received
ssrc	the SSRC identifier of the next packet to be received

Returns: error if any

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◆ gf_rtp_get_current_time()

Double gf_rtp_get_current_time ( GF_RTPChannel * ch )

retrieves current RTP time in sec. If rtp_time was unknown (not on demand media) the time is absolute. Otherwise this is the time in ms elapsed since the last PLAY range start value Not supported yet if played without RTSP (aka RTCP time not supported)

Parameters

ch	the target RTP channel

Returns: NTP clock in seconds

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◆ gf_rtp_reset_buffers()

void gf_rtp_reset_buffers ( GF_RTPChannel * ch )

resets both sockets and packet reorderer

Parameters

ch	the target RTP channel

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◆ gf_rtp_reset_ssrc()

void gf_rtp_reset_ssrc ( GF_RTPChannel * ch )

resets sender SSRC

Parameters

ch	the target RTP channel

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◆ gf_rtp_read_rtp()

u32 gf_rtp_read_rtp	(	GF_RTPChannel *	ch,
		u8 *	buffer,
		u32	buffer_size
	)

reads any RTP data on UDP only (not valid for TCP). Performs re-ordering if configured for it

Parameters

ch	the target RTP channel
buffer	the buffer where to store the RTP packet
buffer_size	the size of the buffer

Returns: amount of data read in bytes

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◆ gf_rtp_flush_rtp()

u32 gf_rtp_flush_rtp	(	GF_RTPChannel *	ch,
		u8 *	buffer,
		u32	buffer_size
	)

flushes any pending data in packet reorderer, but does not flush packet reorderer if reorderer timeout is not exceeded

Parameters

ch	the target RTP channel
buffer	the buffer where to store the data
buffer_size	the size of the buffer

Returns: amount of data read in bytes

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◆ gf_rtp_read_rtcp()

u32 gf_rtp_read_rtcp	(	GF_RTPChannel *	ch,
		u8 *	buffer,
		u32	buffer_size
	)

reads any RTCP data on UDP only (not valid for TCP). Performs re-ordering if configured for it

Parameters

ch	the target RTP channel
buffer	the buffer where to store the RTCP packet
buffer_size	the size of the buffer

Returns: amount of data read in bytes

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◆ gf_rtp_read_flush()

u32 gf_rtp_read_flush	(	GF_RTPChannel *	ch,
		u8 *	buffer,
		u32	buffer_size
	)

flushes any pending data in packet reorderer, and flushes packet reorderer if reorderer timeout is not exceeded. Typically called several times until returning 0.

Parameters

ch	the target RTP channel
buffer	the buffer where to store the data
buffer_size	the size of the buffer

Returns: amount of data read in bytes

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◆ gf_rtp_decode_rtp()

GF_Err gf_rtp_decode_rtp	(	GF_RTPChannel *	ch,
		u8 *	pck,
		u32	pck_size,
		GF_RTPHeader *	rtp_hdr,
		u32 *	PayloadStart
	)

decodes an RTP packet and gets the beginning of the RTP payload

Parameters

ch	the target RTP channel
pck	the RTP packet buffer
pck_size	the size of the RTP packet
rtp_hdr	filled with decoded RTP header information
PayloadStart	set to the offset in bytes of the start of the payload in the RTP packet

Returns: error if any

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◆ gf_rtp_decode_rtcp()

GF_Err gf_rtp_decode_rtcp	(	GF_RTPChannel *	ch,
		u8 *	pck,
		u32	pck_size,
		Bool *	has_sr
	)

decodes an RTCP packet and update timing info, send ReceiverReport too

Parameters

ch	the target RTP channel
pck	the RTP packet buffer
pck_size	the size of the RTP packet
has_sr	set to GF_TRUE if the RTCP packet contained an SenderReport

Returns: error if any

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◆ gf_rtp_send_rtcp_report()

GF_Err gf_rtp_send_rtcp_report ( GF_RTPChannel * ch )

computes and send Receiver report. If the channel is a TCP channel, you must specify the callback function.

Note: Many RTP implementation do NOT process RTCP info received on TCP... the lib will decide whether the report shall be sent or not, therefore you should call this function at regular times

Parameters

ch	the target RTP channel

Returns: error if any

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◆ gf_rtp_set_loss_rate()

void gf_rtp_set_loss_rate	(	GF_RTPChannel *	ch,
		u32	loss_rate
	)

forces loss rate for next Receiver report

Parameters

ch	the target RTP channel
loss_rate	loss rate in per-thousand

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◆ gf_rtp_send_bye()

GF_Err gf_rtp_send_bye ( GF_RTPChannel * ch )

sends a BYE info (leaving the session)

Parameters

ch	the target RTP channel

Returns: error if any

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◆ gf_rtp_send_packet()

GF_Err gf_rtp_send_packet	(	GF_RTPChannel *	ch,
		GF_RTPHeader *	rtp_hdr,
		u8 *	pck,
		u32	pck_size,
		Bool	fast_send
	)

sends an RTP packet. In fast_send mode,

Parameters

ch	the target RTP channel
rtp_hdr	the RTP header of the packet
pck	the RTP payload buffer
pck_size	the RTP payload size
fast_send	if set, the payload buffer contains 12 bytes available BEFORE its indicated start where the RTP header is written in place

Returns: error if any

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◆ gf_rtp_set_interleave_callbacks()

GF_Err gf_rtp_set_interleave_callbacks	(	GF_RTPChannel *	ch,
		gf_rtp_tcp_callback	tcp_callback,
		void *	cbk1,
		void *	cbk2
	)

sets RTP interleaved callback on the RTP channel

Parameters

ch	the target RTP channel
tcp_callback	the callback function
cbk1	user data for the callback function
cbk2	user data for the callback function

Returns: error if any

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◆ gf_rtp_is_unicast()

u32 gf_rtp_is_unicast ( GF_RTPChannel * ch )

checks if an RTP channel is unicast

Parameters

ch	the target RTP channel

Returns: GF_TRUE if unicast, GF_FALSE otherwise

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◆ gf_rtp_is_interleaved()

u32 gf_rtp_is_interleaved ( GF_RTPChannel * ch )

checks if an RTP channel is interleaved

Parameters

ch	the target RTP channel

Returns: GF_TRUE if interleaved, GF_FALSE otherwise

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◆ gf_rtp_get_clockrate()

u32 gf_rtp_get_clockrate ( GF_RTPChannel * ch )

gets clockrate/timescale of an RTP channel

Parameters

ch	the target RTP channel

Returns: the RTP clock rate

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◆ gf_rtp_get_low_interleave_id()

u8 gf_rtp_get_low_interleave_id ( GF_RTPChannel * ch )

gets the low interleave ID of an RTP channel

Parameters

ch	the target RTP channel

Returns: the low (RTP) interleave ID of the channel, or 0 if not interleaved

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◆ gf_rtp_get_hight_interleave_id()

u8 gf_rtp_get_hight_interleave_id ( GF_RTPChannel * ch )

gets the high interleave ID of an RTP channel

Parameters

ch	the target RTP channel

Returns: the high (RTCP) interleave ID of the channel, or 0 if not interleaved

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◆ gf_rtp_get_transport()

const GF_RTSPTransport* gf_rtp_get_transport ( GF_RTPChannel * ch )

gets the transport associated with an RTP channel

Parameters

ch	the target RTP channel

Returns: the RTSP transport information

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◆ gf_rtp_get_loss()

Float gf_rtp_get_loss ( GF_RTPChannel * ch )

gets loss rate of the RTP channel

Parameters

ch	the target RTP channel

Returns: the loss rate of the channel, between 0 and 100 %

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◆ gf_rtp_get_tcp_bytes_sent()

u32 gf_rtp_get_tcp_bytes_sent ( GF_RTPChannel * ch )

gets number of TCP bytes send for an interleaved channel

Parameters

ch	the target RTP channel

Returns: the number of bytes sent

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◆ gf_rtp_get_ports()

void gf_rtp_get_ports	(	GF_RTPChannel *	ch,
		u16 *	rtp_port,
		u16 *	rtcp_port
	)

gets ports of an non-interleaved RTP channel

Parameters

ch	the target RTP channel
rtp_port	the RTP port number
rtcp_port	the RCTP port number

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◆ gf_sdp_info_new()

GF_SDPInfo* gf_sdp_info_new ( )

creates a new SDP info

Returns: the newly allocated SDP

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◆ gf_sdp_info_del()

void gf_sdp_info_del ( GF_SDPInfo * sdp )

destrucs an SDP info Memory Consideration: the destructors free all non-NULL string. You should therefore be careful while (de-)assigning the strings. The function gf_sdp_info_parse() performs a complete reset of the GF_SDPInfo

Parameters

sdp	the target SDP to destroy

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◆ gf_sdp_info_reset()

void gf_sdp_info_reset ( GF_SDPInfo * sdp )

resets all structures and destroys substructures too

Parameters

sdp	the target SDP to destroy

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◆ gf_sdp_info_parse()

GF_Err gf_sdp_info_parse	(	GF_SDPInfo *	sdp,
		char *	sdp_text,
		u32	text_size
	)

parses a memory SDP buffer

Parameters

sdp	the target SDP to destroy
sdp_text	the encoded SDP buffer
text_size	sizeo if the encoded SDP buffer

Returns: error if any

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◆ gf_sdp_media_new()

GF_SDPMedia* gf_sdp_media_new ( )

creates a new SDP media

Returns: the newly allocated SDP media

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◆ gf_sdp_media_del()

void gf_sdp_media_del ( GF_SDPMedia * media )

destroys an SDP media

Parameters

media the target SDP media

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◆ gf_sdp_conn_new()

GF_SDPConnection* gf_sdp_conn_new ( )

creates a new SDP connection

Returns: the newly allocated SDP connection

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◆ gf_sdp_conn_del()

void gf_sdp_conn_del ( GF_SDPConnection * conn )

destroys an SDP connection

Parameters

conn	the target SDP connection

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◆ gf_sdp_fmtp_new()

GF_SDP_FMTP* gf_sdp_fmtp_new ( )

creates a new SDP payload

Returns: the newly allocated SDP payload

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◆ gf_sdp_fmtp_del()

void gf_sdp_fmtp_del ( GF_SDP_FMTP * fmtp )

destroys an SDP payload

Parameters

fmtp	the target SDP payload

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◆ gf_rtp_builder_new()

GP_RTPPacketizer* gf_rtp_builder_new	(	u32	rtp_payt,
		GF_SLConfig *	slc,
		u32	flags,
		void *	cbk_obj,
		void()(void cbk, GF_RTPHeader *header)	OnNewPacket,
		void()(void cbk, GF_RTPHeader *header)	OnPacketDone,
		void()(void cbk, u32 payload_size, u32 offset_from_orig)	OnDataReference,
		void()(void cbk, u8 *data, u32 data_size, Bool is_head)	OnData
	)

creates a new builder

Parameters

rtp_payt	rtp payload format, one of the above GF_RTP_PAYT_*
slc	user-given SL config to use. If none specified, default RFC config is used
flags	packetizer flags, one of the above GP_RTP_PCK_*
cbk_obj	callback object passed back in functions
OnNewPacket	callback function starting new RTP packet header: rtp header for new packet - note that RTP header flags are not used until PacketDone is called
OnPacketDone	callback function closing current RTP packet header: final rtp header for packet
OnDataReference	optional, to call each time data from input buffer is added to current RTP packet. If not set, data must be added through OnData payload_size: size of reference data offset_from_orig: start offset in input buffer
OnData	to call each time data is added to current RTP packet (either extra data from payload or data from input when not using referencing) is_head: signal the data added MUST be inserted at the beginning of the payload. Otherwise data is concatenated as received

Returns: a new RTP packetizer object

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◆ gf_rtp_builder_del()

void gf_rtp_builder_del ( GP_RTPPacketizer * builder )

destroys an RTP packetizer

Parameters

builder the target RTP packetizer

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◆ gf_rtp_builder_init()

void gf_rtp_builder_init	(	GP_RTPPacketizer *	builder,
		u8	PayloadType,
		u32	MaxPayloadSize,
		u32	max_ptime,
		u32	StreamType,
		u32	codecid,
		u32	PL_ID,
		u32	avgSize,
		u32	maxSize,
		u32	avgTS,
		u32	maxDTS,
		u32	IV_length,
		u32	KI_length,
		char *	pref_mode
	)

inits the RTP packetizer

Parameters

builder	the target RTP packetizer
PayloadType	the payload type to use
MaxPayloadSize	maximum payload size of RTP packets (eg MTU minus IP/UDP/RTP headers)
max_ptime	maximum packet duration IN RTP TIMESCALE
StreamType	MPEG-4 system stream type - MUST always be provided for payloads format specifying audio or video streams
codecid	ID of the media codec
PL_ID	profile and level identifier for the stream all other params are for RFC 3640 ***
avgSize	average size of an AU. This is not always known (real-time encoding). In this case you should specify a rough compute indicating how many packets could be stored per RTP packet. for ex AAC stereo at 44100 k / 64kbps , one AU ~= 380 bytes so 3 AUs for 1500 MTU is ok - BE CAREFUL: MultiSL adds some SL info on top of the 12 byte RTP header so you should specify a smaller size The packetizer will ALWAYS make sure there's no pb storing the packets so specifying more will result in a slight overhead in the SL mapping but the gain to singleSL will still be worth it. -Nota: at init, the packetizer can decide to switch to SingleSL if the average size specified is too close to the PathMTU
maxSize	max size of an AU. If unknown (real-time) set to 0
avgTS	average CTS progression (1000/FPS for video)
maxDTS	maximum DTS offset in case of bidirectional coding.
IV_length	size (in bytes) of IV when ISMACrypted
KI_length	size (in bytes) of key indicator when ISMACrypted
pref_mode	MPEG-4 generic only, specifies the payload mode - can be NULL (mode generic)

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◆ gf_rtp_builder_set_cryp_info()

void gf_rtp_builder_set_cryp_info	(	GP_RTPPacketizer *	builder,
		u64	IV,
		char *	key_indicator,
		Bool	is_encrypted
	)

sets frame crypto info (ISMA E&A) for frame starting in next RTP packet

Parameters

builder	the target RTP packetizer
IV	initialization vector
key_indicator	key indicator
is_encrypted	encrypted flag

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◆ gf_rtp_builder_process()

GF_Err gf_rtp_builder_process	(	GP_RTPPacketizer *	builder,
		u8 *	data,
		u32	data_size,
		u8	IsAUEnd,
		u32	FullAUSize,
		u32	duration,
		u8	descIndex
	)

packetizes input buffer

Parameters

builder	the target RTP packetizer
data	input buffer
data_size	input buffer size
IsAUEnd	set to one if this buffer is the last of the AU
FullAUSize	complete access unit size if known, 0 otherwise
duration	sample duration in rtp timescale (mostly needed for 3GPP text streams)
descIndex	sample description index (mostly needed for 3GPP text streams)

Returns: error if any

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◆ gf_rtp_builder_format_sdp()

GF_Err gf_rtp_builder_format_sdp	(	GP_RTPPacketizer *	builder,
		char *	payload_name,
		char **	out_sdp_line,
		char *	dsi,
		u32	dsi_size
	)

formats the "fmtp: " attribute for the MPEG-4 generic packetizer. sdpline shall be at least 2000 char

Parameters

builder	the target RTP packetizer
payload_name	name of the payload to use (profile of RFC 3640)
out_sdp_line	SDP line buffer produced - must be freed by caller
dsi	decoder config of stream if any, or NULL
dsi_size	size of the decoder config

Returns: error if any

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◆ gf_rtp_builder_get_payload_name()

Bool gf_rtp_builder_get_payload_name	(	GP_RTPPacketizer *	builder,
		char	payload_name[20],
		char	media_name[20]
	)

formats SDP payload name and media name

Parameters

builder	the target RTP packetizer
payload_name	the buffer to fill with the payload name
media_name	the buffer to fill with the payload name

Returns: GF_TRUE if success, GF_FALSE otherwise

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◆ gf_rtp_depacketizer_new()

GF_RTPDepacketizer* gf_rtp_depacketizer_new	(	GF_SDPMedia *	media,
		u32	hdr_payt,
		gf_rtp_packet_cbk	sl_packet_cbk,
		void *	udta
	)

creates a new depacketizer

Parameters

media	the SDP media structure describing the RTP stream - can be NULL for static payload types
hdr_payt	the static RTP payload type when no SDP is used
sl_packet_cbk	callback function of the depacketizer to retrieves payload
udta	opaque data for the callback function

Returns: a newly allocated RTP depacketizer, or NULL if not supported

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◆ gf_rtp_depacketizer_del()

void gf_rtp_depacketizer_del ( GF_RTPDepacketizer * rtpd )

destroys an RTP depacketizer

Parameters

rtpd	the target RTP depacketizer

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◆ gf_rtp_depacketizer_reset()

void gf_rtp_depacketizer_reset	(	GF_RTPDepacketizer *	rtpd,
		Bool	full_reset
	)

resets an RTP depacketizer, assuming next packet should be an AU start

Parameters

rtpd	the target RTP depacketizer
full_reset	if set, completely reset the SL header

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◆ gf_rtp_depacketizer_process()

void gf_rtp_depacketizer_process	(	GF_RTPDepacketizer *	rtpd,
		GF_RTPHeader *	hdr,
		u8 *	payload,
		u32	size
	)

process an RTP depacketizer

Parameters

rtpd	the target RTP depacketizer
hdr	the RTP packet header
payload	the RTP packet payload
size	the RTP packet payload size

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Data Structures

Macros

Typedefs

Enumerations

Functions

Detailed Description

Data Structure Documentation

◆ GF_RTSPRange

◆ GF_RTSPTransport

◆ GF_RTSPCommand

◆ GF_RTPInfo

◆ GF_RTSPResponse

◆ GF_RTPHeader

◆ GF_RTPMap

◆ GF_X_Attribute

◆ GF_SDPBandwidth

◆ GF_SDPTiming

◆ GF_SDPConnection

◆ GF_SDP_FMTP

◆ GF_SDPMedia

◆ GF_SDPInfo

◆ GP_RTPSLMap

◆ GF_RTPStaticMap

Macro Definition Documentation

◆ GF_RTSP_VERSION

◆ GF_RTSP_PROFILE_RTP_AVP

◆ GF_RTSP_PROFILE_RTP_AVP_TCP

◆ GF_RTSP_PROFILE_UDP

◆ GF_RTSP_DESCRIBE

◆ GF_RTSP_SETUP

◆ GF_RTSP_PLAY

◆ GF_RTSP_PAUSE

◆ GF_RTSP_RECORD

◆ GF_RTSP_TEARDOWN

◆ GF_RTSP_GET_PARAMETER

◆ GF_RTSP_SET_PARAMETER

◆ GF_RTSP_OPTIONS

◆ GF_RTSP_ANNOUNCE

◆ GF_RTSP_REDIRECT

◆ GF_RTSP_TUNNEL_POST

◆ GF_SDP_MAX_TIMEOFFSET

Typedef Documentation

◆ gf_rtsp_interleave_callback

◆ gf_rtp_tcp_callback

◆ gf_rtp_packet_cbk

Enumeration Type Documentation

◆ anonymous enum

◆ anonymous enum

◆ anonymous enum

◆ anonymous enum

◆ anonymous enum

Function Documentation

◆ gf_rtsp_nc_to_string()

◆ gf_rtsp_range_parse()

◆ gf_rtsp_range_new()

◆ gf_rtsp_range_del()

◆ gf_rtsp_transport_clone()

◆ gf_rtsp_transport_del()

◆ gf_rtsp_command_new()

◆ gf_rtsp_command_del()

◆ gf_rtsp_command_reset()

◆ gf_rtsp_response_new()

◆ gf_rtsp_response_del()

◆ gf_rtsp_response_reset()

◆ gf_rtsp_session_new()

◆ gf_rtsp_session_del()

◆ gf_rtsp_set_buffer_size()

◆ gf_rtsp_session_reset()

◆ gf_rtsp_is_my_session()

◆ gf_rtsp_get_server_name()

◆ gf_rtsp_get_user()

◆ gf_rtsp_get_password()

◆ gf_rtsp_get_session_port()

◆ gf_rtsp_use_tls()

◆ gf_rtsp_get_response()

◆ gf_rtsp_get_session_state()

◆ gf_rtsp_reset_aggregation()

◆ gf_rtsp_send_command()

◆ gf_rtsp_check_connection()

◆ gf_rtsp_set_interleave_callback()