API-socket-options.md

SRT API Socket Options

There is a general method of setting options on a socket in the SRT C API, similar to the system setsockopt/getsockopt functions.

• Types Used in Socket Options
• Getting and Setting Options
• List of Options

Types Used in Socket Options

Possible types of socket options are:

• int32_t - a 32-bit integer. On most systems similar to int. In some cases the value is expressed using an enumeration type (see Enumeration types... section below).

• int64_t - a 64-bit integer.

• bool - a Boolean type (<stdbool.h> for C, or built-in for C++). When setting an option, passing the value through an int type is also properly recognized. When getting an option, however, thebool type should be used. It is also possible to pass a variable of int type initialized with 0 and then comparing the resulting value with 0 (just don't compare the result with 1 or true).

• string - a C string. When setting an option, a const char* character array pointer is expected to be passed in optval and the array length in optlen without the terminating NULL character. When getting, an array is expected to be passed in optval with a sufficient size with an extra space for the terminating NULL character provided in optlen. The return value of optlen does not count the terminating NULL character.

• linger - Linger structure. Used exclusively with SRTO_LINGER.

Enumeration Types Used in Options

SRT_TRANSTYPE

Used by SRTO_TRANSTYPE option:

• SRTT_LIVE: Live mode.
• SRTT_FILE: File mode.

See Transmission Types for details.

SRT_KM_STATE

The defined encryption state as performed by the Key Material Exchange, used by SRTO_RCVKMSTATE, SRTO_SNDKMSTATE and SRTO_KMSTATE options:

• SRT_KM_S_UNSECURED (0): no encryption/decryption. If this state is only on the receiver, received encrypted packets will be dropped.

• SRT_KM_S_SECURING(1): pending security (HSv4 only). This is a temporary state used only if the connection uses HSv4 and the Key Material Exchange is not finished yet. On HSv5 this is not possible because the Key Material Exchange for the initial key is done in the handshake.

• SRT_KM_S_SECURED (2): KM exchange was successful and the data will be sent encrypted and will be decrypted by the receiver. This state is only possible on both sides in both directions simultaneously. Any unencrypted packet will be dropped by the receiver.

• SRT_KM_S_NOSECRET (3): If this state is in the sending direction (SRTO_SNDKMSTATE), then it means that the sending party has set a passphrase, but the peer did not. In this case the sending party can receive unencrypted packets from the peer, but packets it sends to the peer will be encrypted and the peer will not be able to decrypt them. This state is only possible in HSv5.

• SRT_KM_S_BADSECRET (4): The password is wrong (set differently on each party); encrypted payloads won't be decrypted in either direction.

• SRT_KM_S_BADCRYPTOMODE (5): The crypto mode mode configuration is either not supported or mismatches the configuration of the peer.

Note that with the default value of SRTO_ENFORCEDENCRYPTION option (true), the state is equal on both sides in both directions, and it can be only SRT_KM_S_UNSECURED or SRT_KM_S_SECURED (in other cases the connection is rejected). Otherwise it may happen that either both sides have different passwords and the state is SRT_KM_S_BADSECRET in both directions, or only one party has set a password, in which case the KM state is as follows:

	SRTO_RCVKMSTATE	SRTO_SNDKMSTATE
Party with no password:	SRT_KM_S_NOSECRET	SRT_KM_S_UNSECURED
Party with password:	SRT_KM_S_UNSECURED	SRT_KM_S_NOSECRET

Getting and Setting Options

Legacy version:

int srt_getsockopt(SRTSOCKET socket, int level, SRT_SOCKOPT optName, void* optval, int& optlen);
int srt_setsockopt(SRTSOCKET socket, int level, SRT_SOCKOPT optName, const void* optval, int optlen);

New version:

int srt_getsockflag(SRTSOCKET socket, SRT_SOCKOPT optName, void* optval, int& optlen);
int srt_setsockflag(SRTSOCKET socket, SRT_SOCKOPT optName, const void* optval, int optlen);

In the legacy version, there's an additional unused level parameter. It was there in the original UDT API just to mimic the system setsockopt function, but it's ignored.

Some options require a value of type bool while others require type integer, which is not the same -- they differ in size, and mistaking them may end up causing a crash. This must be kept in mind especially in any C wrapper. For convenience, the setting option function may accept both int32_t and bool types, but this is not so in the case of getting an option value.

UDT project legacy note: Almost all options from the UDT library are derived (there are a few deleted, including some deprecated already in UDT). Many new SRT options have been added. All options are available exclusively with the SRTO_ prefix. Old names are provided as alias names in the udt.h legacy/C++ API file. Note the translation rules:

• UDT_ prefix from UDT options was changed to the prefix SRTO_
• UDP_ prefix from UDT options was changed to the prefix SRTO_UDP_
• SRT_ prefix in older SRT versions was changed to SRTO_

The table below provides a complete list of SRT options and their characteristics according to the following legend:

1. Since: Defines the SRT version when this option was first introduced. If this field is empty, it's an option derived from UDT. "Version 0.0.0" is the oldest version of SRT ever created and put into use.

2. Restrict: Defines restrictions on setting the option. The field is empty if the option is not settable (see Dir column):

   • pre-bind: The option cannot be altered on a socket that is already bound (by calling srt_bind() or any other function doing this, including automatic binding when trying to connect, as well as accepted sockets). In other words, once an SRT socket has transitioned from SRTS_INIT to SRTS_OPENED socket state.

   • pre: The option cannot be altered on a socket that is in SRTS_LISTENING, SRTS_CONNECTING or SRTS_CONNECTED state. If an option was set on a listener socket, it will be inherited by a socket returned by srt_accept() (except for SRTO_STREAMID).

   • post: The option is unrestricted and can be altered at any time, including when the socket is connected, as well as on an accepted socket. The setting of this flag on a listening socket is usually derived by the accepted socket, but this isn't a rule for all options. Note though that there are some unrestricted options that have an important meaning when set prior to connecting (different one than for a connected socket).

   NOTE: The pre-bind characteristic applies exclusively to options that:

   • Change the behavior and functionality of the srt_bind call
   • Concern or set an option on the internally used UDP socket
   • Concern any kind of resource used by the multiplexer

3. Type: The data type of the option (see above).

4. Units: Roughly specified unit, if the value defines things like length or time. It can also define more precisely what kind of specialization can be used when the type is integer:

   • enum: the possible values are defined in an enumeration type
   • flags: the integer value is a collection of bit flags
   • B/s - bytes per second.

5. Default: The exact default value, if it can be easily specified. A more complicated default state of a particular option will be explained in the description (when marked by asterisk). For non-settable options this field is empty.

6. Range: If a value of an integer type has a limited range, or only a certain value allowed, it will be specified here (otherwise empty). A range value can be specified as:

    - `X-... `: specifies only a minimum value
    - `X-Y,Z `: values between X and Y are allowed, and additionally Z
   

   • If the value is of string type, this field will contain its maximum size in square brackets.

   • If the range contains additionally an asterisk, it means that more elaborate restrictions on the value apply, as explained in the description.

7. Dir: Option direction: W if can be set, R if can be retrieved, RW if both.

8. Entity: This describes whether the option can be set on the socket or the group. The G and S options may appear together, in which case both possibilities apply. The D and I options, mutually exclusive, appear always with G. The + marker can only coexist with GS. Possible specifications are:

   • S: This option can be set on a single socket (exclusively, if not GS)

   • G: This option can be set on a group (exclusively, if not GS)

   • D: If set on a group, it will be derived by the member socket

   • I: If set on a group, it will be taken and managed exclusively by the group

   • +: This option is also allowed to be set individually on a group member socket through a configuration object in SRT_SOCKGROUPCONFIG prepared by srt_create_config. Note that this setting may override the setting derived from the group.

List of Options

The following table lists SRT API socket options in alphabetical order. Option details are given further below.

Option Name	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_BINDTODEVICE	1.4.2	pre-bind	string				RW	GSD+
SRTO_CONGESTION	1.3.0	pre	string		"live"	*	W	S
SRTO_CONNTIMEO	1.1.2	pre	int32_t	ms	3000	0..	W	GSD+
SRTO_CRYPTOMODE	1.5.2	pre	int32_t		0 (Auto)	[0, 2]	W	GSD
SRTO_DRIFTTRACER	1.4.2	post	bool		true		RW	GSD
SRTO_ENFORCEDENCRYPTION	1.3.2	pre	bool		true		W	GSD
SRTO_EVENT			int32_t	flags			R	S
SRTO_FC		pre	int32_t	pkts	25600	32..	RW	GSD
SRTO_GROUPCONNECT	1.5.0	pre	int32_t		0	0...1	W	S
SRTO_GROUPMINSTABLETIMEO	1.5.0	pre	int32_t	ms	60	60-...	W	GDI+
SRTO_GROUPTYPE	1.5.0		int32_t	enum			R	S
SRTO_INPUTBW	1.0.5	post	int64_t	B/s	0	0..	RW	GSD
SRTO_IPTOS	1.0.5	pre-bind	int32_t		(system)	0..255	RW	GSD
SRTO_IPTTL	1.0.5	pre-bind	int32_t	hops	(system)	1..255	RW	GSD
SRTO_IPV6ONLY	1.4.0	pre-bind	int32_t		(system)	-1..1	RW	GSD
SRTO_ISN	1.3.0		int32_t				R	S
SRTO_KMPREANNOUNCE	1.3.2	pre	int32_t	pkts	0: 212	0.. *	RW	GSD
SRTO_KMREFRESHRATE	1.3.2	pre	int32_t	pkts	0: 224	0..	RW	GSD
SRTO_KMSTATE	1.0.2		int32_t	enum			R	S
SRTO_LATENCY	1.0.2	pre	int32_t	ms	120 *	0..	RW	GSD
SRTO_LINGER		post	linger	s	off *	0..	RW	GSD
SRTO_LOSSMAXTTL	1.2.0	post	int32_t	packets	0	0..	RW	GSD+
SRTO_MAXBW		post	int64_t	B/s	-1	-1..	RW	GSD
SRTO_MAXREXMITBW	1.5.3	post	int64_t	B/s	-1	-1..	RW	GSD
SRTO_MESSAGEAPI	1.3.0	pre	bool		true		W	GSD
SRTO_MININPUTBW	1.4.3	post	int64_t	B/s	0	0..	RW	GSD
SRTO_MINVERSION	1.3.0	pre	int32_t	version	0x010000	*	RW	GSD
SRTO_MSS		pre-bind	int32_t	bytes	1500	76..	RW	GSD
SRTO_NAKREPORT	1.1.0	pre	bool		*		RW	GSD+
SRTO_OHEADBW	1.0.5	post	int32_t	%	25	5..100	RW	GSD
SRTO_PACKETFILTER	1.4.0	pre	string		""	[512]	RW	GSD
SRTO_PASSPHRASE	0.0.0	pre	string		""	[10..80]	W	GSD
SRTO_PAYLOADSIZE	1.3.0	pre	int32_t	bytes	*	0.. *	W	GSD
SRTO_PBKEYLEN	0.0.0	pre	int32_t	bytes	0	*	RW	GSD
SRTO_PEERIDLETIMEO	1.3.3	pre	int32_t	ms	5000	0..	RW	GSD+
SRTO_PEERLATENCY	1.3.0	pre	int32_t	ms	0	0..	RW	GSD
SRTO_PEERVERSION	1.1.0		int32_t	*			R	GS
SRTO_RCVBUF		pre-bind	int32_t	bytes	8192 payloads	*	RW	GSD+
SRTO_RCVDATA			int32_t	pkts			R	S
SRTO_RCVKMSTATE	1.2.0		int32_t	enum			R	S
SRTO_RCVLATENCY	1.3.0	pre	int32_t	msec	*	0..	RW	GSD
SRTO_RCVSYN		post	bool		true		RW	GSI
SRTO_RCVTIMEO		post	int32_t	ms	-1	-1, 0..	RW	GSI
SRTO_RENDEZVOUS		pre	bool		false		RW	S
SRTO_RETRANSMITALGO	1.4.2	pre	int32_t		1	[0, 1]	RW	GSD
SRTO_REUSEADDR		pre-bind	bool		true		RW	GSD
SRTO_SENDER	1.0.4	pre	bool		false		W	S
SRTO_SNDBUF		pre-bind	int32_t	bytes	8192 payloads	*	RW	GSD+
SRTO_SNDDATA			int32_t	pkts			R	S
SRTO_SNDDROPDELAY	1.3.2	post	int32_t	ms	*	-1..	W	GSD+
SRTO_SNDKMSTATE	1.2.0		int32_t	enum			R	S
SRTO_SNDSYN		post	bool		true		RW	GSI
SRTO_SNDTIMEO		post	int32_t	ms	-1	-1..	RW	GSI
SRTO_STATE			int32_t	enum			R	S
SRTO_STREAMID	1.3.0	pre	string		""	[512]	RW	GSD
SRTO_TLPKTDROP	1.0.6	pre	bool		*		RW	GSD
SRTO_TRANSTYPE	1.3.0	pre	int32_t	enum	SRTT_LIVE	*	W	S
SRTO_TSBPDMODE	0.0.0	pre	bool		*		W	S
SRTO_UDP_RCVBUF		pre-bind	int32_t	bytes	8192 payloads	*	RW	GSD+
SRTO_UDP_SNDBUF		pre-bind	int32_t	bytes	65536	*	RW	GSD+
SRTO_VERSION	1.1.0		int32_t				R	S

Option Descriptions

SRTO_BINDTODEVICE

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_BINDTODEVICE	1.4.2	pre-bind	string				RW	GSD+

Refers to the SO_BINDTODEVICE system socket option for SOL_SOCKET level. This effectively limits the packets received by this socket to only those that are targeted to that device. The device is specified by name passed as string. The setting becomes effective after binding the socket (including default-binding when connecting).

NOTE: This option is only available on Linux and available there by default. On all other platforms setting this option will always fail.

NOTE: With the default system configuration, this option is only available for a process that runs as root. Otherwise the function that applies the setting (srt_bind, srt_connect etc.) will fail.

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SRTO_CONGESTION

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_CONGESTION	1.3.0	pre	string		"live"	*	W	S

The type of congestion controller used for the transmission for that socket.

Its type must be exactly the same on both connecting parties, otherwise the connection is rejected - however you may also change the value of this option for the accepted socket in the listener callback (see srt_listen_callback) if an appropriate instruction was given in the Stream ID.

Currently supported congestion controllers are designated as "live" and "file", which correspond to the Live and File modes.

Note that it is not recommended to change this option manually, but you should rather change the whole set of options using the SRTO_TRANSTYPE option.

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SRTO_CONNTIMEO

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_CONNTIMEO	1.1.2	pre	int32_t	msec	3000	0..	W	GSD+

Connect timeout. This option applies to the caller and rendezvous connection modes. For the rendezvous mode (see SRTO_RENDEZVOUS) the effective connection timeout will be 10 times the value set with SRTO_CONNTIMEO.

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SRTO_CRYPTOMODE

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_CRYPTOMODE	1.5.2	pre	int32_t		0 (Auto)	[0, 2]	RW	GSD

The encryption mode to be used if the SRTO_PASSPHRASE is set.

The feature is a part of the AEAD Preview API of SRT v1.5.2, and is disabled by default. To enable use ENABLE_AEAD_API_PREVIEW build option.

Crypto modes:

• 0: SRT listener accepts any mode from the caller. SRT Caller or SRT Rendezvous effectively negotiate AES-CTR (1).
• 1: regular AES-CTR (without message integrity authentication).
• 2: AES-GCM mode with message integrity authentication (AEAD).

The AES-GCM mode (2) is only allowed if TSBPD is enabled (SRTO_TSBPDMODE). Auto (0) is equivalent to AES-CTR (1) in Rendezvous and for a Caller.

Once a connection is established, reading SRTO_CRYPTOMODE shows the negotiated crypto mode: AES-CTR (1) or AES-GCM (2).

When Listener callback is used, the value of SRTO_CRYPTOMODE read on the new SRT socket to be accepted is not yet the negotiated one. It is the value to be negotiated, and is inherited from the listener SRT socket. If a specific behavior for each individual connection request is desired based on the user ID or anything else, the intended behavior can be achieved by setting the SRTO_CRYPTOMODE on the new SRT socket to a specific value.

For example, let's say the initial value set on the listener socket is Auto (0). The listener callback is called, signalling the new connection request. The user ID is extracted, and the server wants to force AES-GCM only for the connection from this specific user. In this case AES-GCM (2) can be set on the new socket.

There is no way to check the crypto mode being requested by the SRT caller at this point.

Caller	Listener	Negotiated		Rdv In-tor	Rdv Res-der	Negotiated
0 (auto)	0 (auto)	AES-CTR (1)		0 (auto)	0 (auto)	AES-CTR (1)
0 (auto)	AES-CTR (1)	AES-CTR (1)		0 (auto)	AES-CTR (1)	AES-CTR (1)
0 (auto)	AES-GCM (2)	reject		0 (auto)	AES-GCM (2)	reject
AES-CTR (1)	0 (auto)	AES-CTR (1)		AES-CTR (1)	0 (auto)	AES-CTR (1)
AES-CTR (1)	AES-CTR (1)	AES-CTR (1)		AES-CTR (1)	AES-CTR (1)	AES-CTR (1)
AES-CTR (1)	AES-GCM (2)	reject		AES-CTR (1)	AES-GCM (2)	reject
AES-GCM (2)	0 (auto)	AES-GCM (2)		AES-GCM (2)	0 (auto)	reject
AES-GCM (2)	AES-CTR (1)	reject		AES-GCM (2)	AES-CTR (1)	reject
AES-GCM (2)	AES-GCM (2)	AES-GCM (2)		AES-GCM (2)	AES-GCM (2)	AES-GCM (2)

• Rdv - Rendezvous; In-tor - initiator; Res-der - responder.

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SRTO_DRIFTTRACER

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_DRIFTTRACER	1.4.2	post	bool		true		RW	GSD

Enables or disables time drift tracer (receiver).

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SRTO_ENFORCEDENCRYPTION

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_ENFORCEDENCRYPTION	1.3.2	pre	bool		true		W	GSD

This option enforces that both connection parties have the same passphrase set, or both do not set the passphrase, otherwise the connection is rejected.

When this option is set to FALSE on both connection parties, the connection is allowed even if the passphrase differs on both parties, or it was set only on one party. Note that the party that has set a passphrase is still allowed to send data over the network. However, the receiver will not be able to decrypt that data and will not deliver it to the application. The party that has set no passphrase can send (unencrypted) data that will be successfully received by its peer.

This option can be used in some specific situations when the user knows both parties of the connection, so there's no possible situation of a rogue sender and can be useful in situations where it is important to know whether a connection is possible. The inability to decrypt an incoming transmission can be then reported as a different kind of problem.

IMPORTANT: There is unusual and unobvious behavior when this flag is TRUE on the caller and FALSE on the listener, and the passphrase was mismatched. On the listener side the connection will be established and broken right after, resulting in a short-lived "spurious" connection report on the listener socket. This way, a socket will be available for retrieval from an srt_accept call for a very short time, after which it will be removed from the listener backlog just as if no connection attempt was made at all. If the application is fast enough to react on an incoming connection, it will retrieve it, only to learn that it is already broken. This also makes possible a scenario where SRT_EPOLL_IN is reported on a listener socket, but then an srt_accept call reports an SRT_EASYNCRCV error. How fast the connection gets broken depends on the network parameters -- in particular, whether the UMSG_SHUTDOWN message sent by the caller is delivered (which takes one RTT in this case) or missed during the interval from its creation up to the connection timeout (default = 5 seconds). It is therefore strongly recommended that you only set this flag to FALSE on the listener when you are able to ensure that it is also set to FALSE on the caller side.

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SRTO_EVENT

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_EVENT			int32_t	flags			R	S

Returns bit flags set according to the current active events on the socket.

Possible values are those defined in SRT_EPOLL_OPT enum (a combination of SRT_EPOLL_IN, SRT_EPOLL_OUT and SRT_EPOLL_ERR).

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SRTO_FC

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_FC		pre	int32_t	pkts	25600	32..	RW	GSD

Flow Control limits the maximum number of packets "in flight" - payload (data) packets that were sent but reception is not yet acknowledged with an ACK control packet. It also includes data packets already received, but that can't be acknowledged due to loss of preceding data packet(s). In other words, if a data packet with sequence number A was lost, then acknowledgement of the following SRTO_FC packets is blocked until packet A is either successfully retransmitted or dropped by the Too-Late Packet Drop mechanism. Thus the sender will have SRTO_FC packets in flight, and will not be allowed to send further data packets. Therefore, when establishing the value of SRTO_FC, it is recommend taking into consideration possible delays due to packet loss and retransmission.

There is a restriction that the receiver buffer size (SRTO_RCVBUF) must not be greater than SRTO_FC (#700). Therefore, it is recommended to set the value of SRTO_FC first, and then the value of SRTO_RCVBUF.

The default flow control window size is 25600 packets. It is approximately:

• 270 Mbits of payload in the default live streaming configuration with an SRT payload size of 1316 bytes;
• 300 Mbits of payload with an SRT payload size of 1456 bytes.

The minimum number of packets in flight should be (assuming max payload size): FCmin = bps / 8 × RTTsec / (MSS - 44), where

• bps - is the payload bitrate of the stream in bits per second;
• RTTsec - RTT of the network connection in seconds;
• MSS - Maximum segment size (aka MTU), see SRTO_MSS;
• 44 - size of headers (20 bytes IPv4 + 8 bytes of UDP + 16 bytes of SRT packet header).

To avoid blocking the sending of further packets in case of packet loss, the recommended flow control window is FC = bps / 8 × (RTTsec + latency_sec) / (MSS - 44), where latency_sec is the receiver buffering delay (SRTO_RCVLATENCY) in seconds.

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SRTO_GROUPCONNECT

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_GROUPCONNECT	1.5.0	pre	int32_t		0	0...1	W	S

When this flag is set to 1 on a listener socket, it allows this socket to accept group connections. When set to the default 0, group connections will be rejected. Keep in mind that if the SRTO_GROUPCONNECT flag is set to 1 (i.e. group connections are allowed) srt_accept may return a socket or a group ID. A call to srt_accept on a listener socket that has group connections allowed must take this into consideration. It's up to the caller of this function to make this distinction and to take appropriate action depending on the type of entity returned.

When this flag is set to 1 on an accepted socket that is passed to the listener callback handler, it means that this socket is created for a group connection and it will become a member of a group. Note that in this case only the first connection within the group will result in reporting from srt_accept (further connections are handled in the background), and this function will return the group, not this socket ID.

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SRTO_GROUPMINSTABLETIMEO

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_GROUPMINSTABLETIMEO	1.5.0	pre	int32_t	ms	60	60-...	W	GDI+

The option is used for groups of type SRT_GTYPE_BACKUP. It defines the minimum value of the stability timeout for all active member sockets in a group. The actual timeout value is determined in runtime based on the RTT estimate of an individual member socket. If there is no response from the peer for the calculated timeout, the member is considered unstable, triggering activation of an idle backup member.

The smaller the value is, the earlier a backup member might be activated to prepare transition to that path. However, it may also lead to spurious activations of backup paths. The higher the value is, the later the backup link would be activated. All unacknowledged payload packets have to be retransmitted through the backup path. If they don't reach the receiver in time, they would be dropped. Therefore, an appropriate adjustment of the SRT buffering delay (SRTO_PEERLATENCY on sender, SRTO_RCVLATENCY on receiver) should also be considered.

Normally the receiver should send an ACK back to the sender every 10 ms. In the case of congestion, in the live streaming configuration of SRT a loss report is expected to be sent every RTT/2. The network jitter and increase of RTT on the public internet causes these intervals to be stretched. The default minimum value of 60 ms is selected as a general fit for most of the use cases.

Please refer to the SRT Connection Bonding: Main/Backup document for more details.

Note that the value of this option is not allowed to exceed the value of SRTO_PEERIDLETIMEO, which determines the timeout to actually break an idle (irresponsive) connection.

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SRTO_GROUPTYPE

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_GROUPTYPE	1.5.0		int32_t	enum			R	S

This option is read-only and it is intended to be called inside the listener callback handler (see srt_listen_callback). Possible values are defined in the SRT_GROUP_TYPE enumeration type.

This option returns the group type that is declared in the incoming connection. If the incoming connection is not going to make a group-member connection, then the value returned is SRT_GTYPE_UNDEFINED. If this option is read in any other context than inside the listener callback handler, the value is undefined.

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SRTO_INPUTBW

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_INPUTBW	1.0.5	post	int64_t	B/s	0	0..	RW	GSD

This option is effective only if SRTO_MAXBW is set to 0 (relative). It controls the maximum bandwidth together with SRTO_OHEADBW option according to the formula: MAXBW = INPUTBW * (100 + OHEADBW) / 100. When this option is set to 0 (automatic) then the real INPUTBW value will be estimated from the rate of the input (cases when the application calls the srt_send* function) during transmission. The minimum allowed estimate value is restricted by SRTO_MININPUTBW, meaning INPUTBW = MAX(INPUTBW_ESTIMATE; MININPUTBW).

Recommended: set this option to the anticipated bitrate of your live stream and keep the default 25% value for SRTO_OHEADBW.

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SRTO_MININPUTBW

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_MININPUTBW	1.4.3	post	int64_t	B/s	0	0..	RW	GSD

This option is effective only if both SRTO_MAXBW and SRTO_INPUTBW are set to 0. It controls the minimum allowed value of the input bitrate estimate.

See SRTO_INPUTBW.

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---

SRTO_IPTOS

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_IPTOS	1.0.5	pre-bind	int32_t		(system)	0..255	RW	GSD

IPv4 Type of Service (see IP_TOS option for IP) or IPv6 Traffic Class (see IPV6_TCLASS of IPv6) depending on socket address family. Applies to sender only.

NOTE: This option has been tested to work correctly on Linux only.

When getting, the returned value is the user preset for non-connected sockets and the actual value for connected sockets.

Sender: user configurable, default: 0xB8.

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SRTO_IPTTL

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_IPTTL	1.0.5	pre-bind	int32_t	hops	(system)	1..255	RW	GSD

IPv4 Time To Live (see IP_TTL option for IP) or IPv6 unicast hops (see IPV6_UNICAST_HOPS for IPv6) depending on socket address family. Applies to sender only.

NOTE: This option has been tested to work correctly on Linux only.

When getting, the returned value is the user preset for non-connected sockets and the actual value for connected sockets.

Sender: user configurable, default: 64

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SRTO_IPV6ONLY

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_IPV6ONLY	1.4.0	pre-bind	int32_t		(system)	-1..1	RW	GSD

Set system socket option level IPPROTO_IPV6 named IPV6_V6ONLY. This is meaningful only when the socket is going to be bound to the IPv6 wildcard address in6addr_any (known also as ::). If you bind to a wildcard address, you have the following possibilities:

• IPv4 only: bind to an IPv4 wildcard address
• IPv6 only: bind to an IPv6 wildcard address and set this option to 1
• IPv4 and IPv6: bind to an IPv6 wildcard address and set this option to 0

This option's default value is -1 because it is not possible to determine the default value on the current platform, and if you bind to an IPv6 wildcard address, this value is required prior to binding. When you bind implicitly by calling srt_connect on the socket, this isn't a problem -- binding will be done using the system-default value and then extracted afterwards. But if you want to bind explicitly using srt_bind, this option must be set explicitly to 0 or 1 because this information is vital for determining any potential bind conflicts with other sockets.

Possible values are:

• -1: (default) use system-default value (can be used when not binding to IPv6 wildcard ::)
• 0: The binding to in6addr_any will bind to both IPv6 and IPv4 wildcard address
• 1: The binding to in6addr_any will bind only to IPv6 and not IPv4 wildcard address

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SRTO_ISN

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_ISN	1.3.0		int32_t				R	S

The value of the ISN (Initial Sequence Number), which is the first sequence number put on the first UDP packets sent that are carrying an SRT data payload.

This value is useful for developers of some more complicated methods of flow control, possibly with multiple SRT sockets at a time. It is not intended to be used in any regular development.

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SRTO_KMPREANNOUNCE

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_KMPREANNOUNCE	1.3.2	pre	int32_t	pkts	0: 212	0.. *	RW	GSD

The interval (defined in packets) between when a new Stream Encrypting Key (SEK) is sent and when switchover occurs. This value also applies to the subsequent interval between when switchover occurs and when the old SEK is decommissioned.

At SRTO_KMPREANNOUNCE packets before switchover the new key is sent (repeatedly, if necessary, until it is confirmed by the receiver).

At the switchover point (see SRTO_KMREFRESHRATE), the sender starts encrypting and sending packets using the new key. The old key persists in case it is needed to decrypt packets that were in the flight window, or retransmitted packets.

The old key is decommissioned at SRTO_KMPREANNOUNCE packets after switchover.

The allowed range for this value is between 1 and half of the current value of SRTO_KMREFRESHRATE. The minimum value should never be less than the flight window SRTO_FC (i.e. the number of packets that have already left the sender but have not yet arrived at the receiver).

The value of SRTO_KMPREANNOUNCE must not exceed (SRTO_KMREFRESHRATE - 1) / 2`.

Default value: 0 - corresponds to 4096 packets (2¹² or 0x1000).

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SRTO_KMREFRESHRATE

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_KMREFRESHRATE	1.3.2	pre	int32_t	pkts	0: 224	0..	RW	GSD

The number of packets to be transmitted after which the Stream Encryption Key (SEK), used to encrypt packets, will be switched to the new one. Note that the old and new keys live in parallel for a certain period of time (see SRTO_KMPREANNOUNCE) before and after the switchover.

Having a preannounce period before switchover ensures the new SEK is installed at the receiver before the first packet encrypted with the new SEK is received. The old key remains active after switchover in order to decrypt packets that might still be in flight, or packets that have to be retransmitted.

Default value: 0 - corresponds to 16777216 packets (2²⁴ or 0x1000000).

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SRTO_KMSTATE

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_KMSTATE	1.0.2		int32_t	enum			R	S

Keying Material state. This is a legacy option that is equivalent to SRTO_SNDKMSTATE, if the socket has set SRTO_SENDER to true, and SRTO_RCVKMSTATE otherwise. This option is then equal to SRTO_RCVKMSTATE always if your application disregards possible cooperation with a peer older than 1.3.0, but then with the default value of SRTO_ENFORCEDENCRYPTION the value returned by both options is always the same. See SRT_KM_STATE for more details.

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SRTO_LATENCY

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_LATENCY	1.0.2	pre	int32_t	ms	120 *	0..	RW	GSD

This option sets both SRTO_RCVLATENCY and SRTO_PEERLATENCY to the same value specified. Note that the default value for SRTO_RCVLATENCY is modified by the SRTO_TRANSTYPE option.

Prior to SRT version 1.3.0 SRTO_LATENCY was the only option to set the latency. However it is effectively equivalent to setting SRTO_PEERLATENCY in the sending direction (see SRTO_SENDER), and SRTO_RCVLATENCY in the receiving direction. SRT version 1.3.0 and higher support bidirectional transmission, so that each side can be sender and receiver at the same time, and SRTO_SENDER became redundant.

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SRTO_LINGER

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_LINGER		pre	linger	s	off *	0..	RW	GSD

SRT socket linger time on close (similar to SO_LINGER). The default value in the live streaming configuration is OFF. In this type of workflow there is no point for wait for all the data to be delivered after a connection is closed. The default value in the file transfer configuration is 180 s.

SRT recommended value: off (0).

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SRTO_LOSSMAXTTL

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_LOSSMAXTTL	1.2.0	post	int32_t	packets	0	0..	RW	GSD+

The value up to which the Reorder Tolerance may grow. The Reorder Tolerance is the number of packets that must follow the experienced "gap" in sequence numbers of incoming packets so that the loss report is sent (in the hope that the gap is due to packet reordering rather than because of loss). The value of Reorder Tolerance starts from 0 and is set to a greater value when packet reordering is detected This happens when a "belated" packet, with sequence number older than the latest received, has been received, but without retransmission flag. When this is detected the Reorder Tolerance is set to the value of the interval between latest sequence and this packet's sequence, but not more than the value set by SRTO_LOSSMAXTTL. By default this value is set to 0, which means that this mechanism is off.

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SRTO_MAXBW

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_MAXBW	1.0.5	post	int64_t	B/s	-1	-1..	RW	GSD

Maximum send bandwidth:

• -1: infinite (the limit in Live Mode is 1 Gbps);
• 0: relative to input rate (see SRTO_INPUTBW);
• >0: absolute limit in B/s.

NOTE: This option has a default value of -1, regardless of the mode. For live streams it is typically recommended to set the value 0 here and rely on SRTO_INPUTBW and SRTO_OHEADBW options. However, if you want to do so, you should make sure that your stream has a fairly constant bitrate, or that changes are not abrupt, as high bitrate changes may work against the measurement. SRT cannot ensure that this is always the case for a live stream, therefore the default -1 remains even in live mode.

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SRTO_MAXREXMITBW

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_MAXREXMITBW	1.5.3	post	int64_t	B/s	-1	-1..	RW	GSD

Maximum BW limit for retransmissions:

• -1: unlimited;
• 0: do not allow retransmissions.
• >0: BW usage limit in Bytes/sec for packet retransmissions (including 16 bytes of SRT header).

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SRTO_MESSAGEAPI

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_MESSAGEAPI	1.3.0	pre	bool		true		W	GSD

When set, this socket uses the Message API[*], otherwise it uses the Stream API. Note that in live mode (see SRTO_TRANSTYPE option) only the Message API is available. In File mode you can chose to use one of two modes (note that the default for this option is changed with SRTO_TRANSTYPE option):

• Stream API (default for file mode): In this mode you may send as many data as you wish with one sending instruction, or even use dedicated functions that operate directly on a file. The internal facility will take care of any speed and congestion control. When receiving, you can also receive as many data as desired. The data not extracted will be waiting for the next call. There is no boundary between data portions in Stream mode.

• Message API: In this mode your single sending instruction passes exactly one piece of data that has boundaries (a message). Contrary to Live mode, this message may span multiple UDP packets, and the only size limitation is that it shall fit as a whole in the sending buffer. The receiver shall use as large a buffer as necessary to receive the message, otherwise reassembling and delivering the message might not be possible. When the message is not complete (not all packets received or there was a packet loss) it will not be copied to the application's buffer. Messages that are sent later, but were earlier reassembled by the receiver, will be delivered once ready, if the inorder flag was set to false. See srt_sendmsg.

As a comparison to the standard system protocols, the Stream API does transmission similar to TCP, whereas the Message API functions like the SCTP protocol.

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SRTO_MINVERSION

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_MINVERSION	1.3.0	pre	int32_t	version	0x010000	*	RW	GSD

The minimum SRT version that is required from the peer. A connection to a peer that does not satisfy the minimum version requirement will be rejected. See SRTO_VERSION for the version format.

The default value is 0x010000 (SRT v1.0.0).

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SRTO_MSS

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_MSS		pre-bind	int32_t	bytes	1500	76..	RW	GSD

Maximum Segment Size. Used for buffer allocation and rate calculation using packet counter assuming fully filled packets. Each party can set its own MSS value independently. During a handshake the parties exchange MSS values, and the lowest is used.

Generally on the internet MSS is 1500 by default. This is the maximum size of a UDP packet and can be only decreased, unless you have some unusual dedicated network settings. MSS is not to be confused with the size of the UDP payload or SRT payload - this size is the size of the IP packet, including the UDP and SRT headers

THe value of SRTO_MSS must not exceed SRTO_UDP_SNDBUF or SRTO_UDP_RCVBUF.

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SRTO_NAKREPORT

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_NAKREPORT	1.1.0	pre	bool		*		RW	GSD+

When set to true, every report for a detected loss will be repeated when the timeout for the expected retransmission of this loss has expired and the missing packet still wasn't recovered, or wasn't conditionally dropped (see SRTO_TLPKTDROP).

The default is true for Live mode, and false for File mode (see SRTO_TRANSTYPE).

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SRTO_OHEADBW

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_OHEADBW	1.0.5	post	int32_t	%	25	5..100	RW	GSD

Recovery bandwidth overhead above input rate (see SRTO_INPUTBW), in percentage of the input rate. It is effective only if SRTO_MAXBW is set to 0.

Sender: user configurable, default: 25%.

Recommendations:

• Overhead is intended to give you extra bandwidth for the case when a packet has taken part of the bandwidth, but then was lost and has to be retransmitted. Therefore the effective maximum bandwidth should be appropriately higher than your stream's bitrate so that there's some room for retransmission, but still limited so that the retransmitted packets don't cause the bandwidth usage to skyrocket when larger groups of packets are lost

• Don't configure it too low and avoid 0 in the case when you have the SRTO_INPUTBW option set to 0 (automatic). Otherwise your stream will choke and break quickly at any rise in packet loss.

• To do: set-only; get should be supported.

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SRTO_PACKETFILTER

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_PACKETFILTER	1.4.0	pre	string		""	[512]	RW	GSD

Set up the packet filter. The string must match appropriate syntax for packet filter setup. Note also that:

• The configuration is case-sentitive (e.g. "FEC,Cols:20" is not valid).
• Setting this option will fail if you use an unknown filter type.

An empty value for this option means that for this connection the filter isn't required, but it will accept any filter settings if provided by the peer. If this option is changed by both parties simultaneously, the result will be a configuration integrating parameters from both parties, that is:

• parameters provided by both parties are accepted, if they are identical
• parameters that are set only on one side will have the value defined by that side
• parameters not set in either side will be set as default

The connection will be rejected with SRT_REJ_FILTER code in the following cases:

• both sides define a different packet filter type
• for the same key two different values were provided by both sides
• mandatory parameters weren't provided by either side

In case of the built-in fec filter, the mandatory parameter is cols, all others have their default values. For example, the configuration specified as fec,cols:10 is fec,cols:10,rows:1,arq:onreq,layout:even. See how to configure the FEC filter in SRT Packet Filtering & FEC.

Below in the table are examples for the built-in fec filter. Note that the negotiated config need not have parameters in the given order.

Cases when negotiation succeeds:

Peer A	Peer B	Negotiated Config
(no filter)	(no filter)
fec,cols:10	fec	fec,cols:10,rows:1,arq:onreq,layout:even
fec,cols:10	fec,cols:10,rows:20	fec,cols:10,rows:20,arq:onreq,layout:even
fec,layout:staircase	fec,cols:10	fec,cols:10,rows:1,arq:onreq,layout:staircase

In these cases the configuration is rejected with SRT_REJ_FILTER code:

Peer A	Peer B	Error reason
fec	(no filter)	missing cols parameter
fec,rows:20,arq:never	fec,layout:even	missing cols parameter
fec,cols:20	fec,cols:10	cols parameter value conflict
fec,cols:20,rows:20	fec,cols:20,rows:10	rows parameter value conflict

In general it is recommended that one party defines the full configuration, while the other keeps this value empty.

Reading this option after the connection is established will return the full configuration that has been agreed upon by both parties (including default values).

For details, see SRT Packet Filtering & FEC.

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SRTO_PASSPHRASE

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_PASSPHRASE	0.0.0	pre	string		""	[10..80]	W	GSD

Sets the passphrase for encryption. This enables encryption on this party (or disables it, if an empty passphrase is passed). The password must be minimum 10 and maximum 79 characters long.

The passphrase is the shared secret between the sender and the receiver. It is used to generate the Key Encrypting Key using PBKDF2 (Password-Based Key Derivation Function 2).

When a socket with configured passphrase is being connected, the peer must have the same password set, or the connection is rejected. This behavior can be changed by SRTO_ENFORCEDENCRYPTION.

Note that since the introduction of bidirectional support, there's only one initial encryption key to encrypt the stream (new keys after refreshing will be updated independently), and there's no distinction between "service party that defines the password" and "client party that is required to set matching password" - both parties are equivalent, and in order to have a working encrypted connection, they have to simply set the same passphrase.

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SRTO_PAYLOADSIZE

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_PAYLOADSIZE	1.3.0	pre	int32_t	bytes	*	0.. *	W	GSD

Sets the maximum declared size of a single call to sending function in Live mode. When set to 0, there's no limit for a single sending call.

For Live mode: Default value is 1316, but can be increased up to 1456. Note that with the SRTO_PACKETFILTER option additional header space is usually required, which decreases the maximum possible value for SRTO_PAYLOADSIZE.

For File mode: Default value is 0 and it's recommended not to be changed.

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SRTO_PBKEYLEN

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_PBKEYLEN	0.0.0	pre	int32_t	bytes	0	*	RW	GSD

Encryption key length.

Possible values:

• 0 =PBKEYLEN (default value)
• 16 = AES-128 (effective value)
• 24 = AES-192
• 32 = AES-256

The use is slightly different in 1.2.0 (HSv4), and since 1.3.0 (HSv5):

• HSv4: This is set on the sender and enables encryption, if not 0. The receiver shall not set it and will agree on the length as defined by the sender. Being a sender is defined by the SRTO_SENDER socket option set to true; otherwise the party is a receiver.

• HSv5: The "default value" for PBKEYLEN is 0, which means that the PBKEYLEN won't be advertised. The "effective value" for PBKEYLEN is 16, but this applies only when neither party has set the value explicitly (i.e. when both are initially at the default value of 0). If any party has set an explicit value (16, 24, 32) it will be advertised in the handshake. If the other party remains at the default 0, it will accept the peer's value. The situation where both parties set a value should be treated carefully. Actually there are three intended methods of defining it, and all other uses are considered undefined behavior:

  • Unidirectional: the sender shall set PBKEYLEN and the receiver shall not alter the default value 0. The effective PBKEYLEN will be the one set on the sender. The receiver need not know the sender's PBKEYLEN, just the passphrase, PBKEYLEN will be correctly passed.

  • Bidirectional in Caller-Listener arrangement: it is recommended to use a rule whereby you will be setting the PBKEYLEN exclusively either on the Listener (the service defines the encryption rules strictly) or on the Caller (the service allows all clients to freely decide about encryption, or the server uses more elaborate rules for encryption).

  • Bidirectional in Rendezvous mode: In Rendezvous mode it is assumed that the settings (including PBKEYLEN) are known at both ends. as it is with Listener mode, so both parties should set the same passphrase and the same key length, or both should leave the SRTO_PBKEYLEN option unchanged (which results in default 16).

  • Unwanted behavior cases: if both parties set PBKEYLEN and the value on both sides is different, this is considered a conflict. It is resolved by the Initiator party, which takes its own value, if the SRTO_SENDER option is set to true. Otherwise, it takes the value from the Responder party. The assignment of Initiator-Responder roles matches the Caller-Listener layout. In the case of Rendezvous this assignment depends on the result of the cookie comparison. It is highly recommended to never allow this to happen, as this may result in having one party's setting of length = 32 be overridden by the other party's setting of length = 16.

Initiator		Responder		Result
SRTO_PBKEYLEN	SRTO_SENDER	SRTO_PBKEYLEN	SRTO_SENDER
0	any	0	any	AES-128
0	any	AES-128	any	AES-128
0	any	AES-192	any	AES-192
0	any	AES-256	any	AES-256
AES-128	any	0	any	AES-128
AES-128	any	AES-128	any	AES-128
AES-128	0	AES-192	any	AES-192
AES-128	0	AES-256	any	AES-256
AES-128	1	AES-192	any	AES-128
AES-128	1	AES-256	any	AES-128
AES-192	any	0	any	AES-192
AES-192	0	AES-128	any	AES-128
AES-192	any	AES-192	any	AES-192
AES-192	0	AES-256	any	AES-256
AES-192	1	AES-128	any	AES-192
AES-192	1	AES-256	any	AES-192
AES-256	any	0	any	AES-256
AES-256	0	AES-128	any	AES-128
AES-256	0	AES-192	any	AES-192
AES-256	any	AES-256	any	AES-256
AES-256	1	AES-128	any	AES-256
AES-256	1	AES-192	any	AES-256

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SRTO_PEERIDLETIMEO

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_PEERIDLETIMEO	1.3.3	pre	int32_t	ms	5000	0..	RW	GSD+

The maximum time in [ms] to wait until another packet is received from a peer since the last such packet reception. If this time is passed, the connection is considered broken on timeout.

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SRTO_PEERLATENCY

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_PEERLATENCY	1.3.0	pre	int32_t	ms	0	0..	RW	GSD

The latency value (as described in SRTO_RCVLATENCY) provided by the sender side as a minimum value for the receiver.

This value is only significant when SRTO_TSBPDMODE is enabled.

Reading the value of the option on an unconnected socket reports the configured value. Reading the value on a connected socket reports the effective receiver buffering latency of the peer.

The SRTO_PEERLATENCY option in versions prior to 1.3.0 is only available as SRTO_LATENCY.

See also SRTO_LATENCY.

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SRTO_PEERVERSION

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_PEERVERSION	1.1.0		int32_t	*			R	GS

SRT version used by the peer. The value 0 is returned if not connected, SRT handshake not yet performed (HSv4 only), or if peer is not SRT. See SRTO_VERSION for the version format.

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SRTO_RCVBUF

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_RCVBUF		pre-bind	int32_t	bytes	8192 bufs	*	RW	GSD+

Receive Buffer Size, in bytes. Note, however, that the internal setting of this value is in the number of buffers, each one of size equal to SRT payload size, which is the value of SRTO_MSS decreased by UDP and SRT header sizes (28 and 16). The value set here will be effectively aligned to the multiple of payload size.

• Minimum value: 32 buffers (46592 with default value of SRTO_MSS).

• Maximum value: SRTO_FC number of buffers (receiver buffer must not be greater than the Flight Flag size).

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SRTO_RCVDATA

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_RCVDATA			int32_t	pkts			R	S

Size of the available data in the receive buffer.

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SRTO_RCVKMSTATE

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_RCVKMSTATE	1.2.0		int32_t	enum			R	S

KM state on the agent side when it's a receiver.

Values defined in enum SRT_KM_STATE.

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SRTO_RCVLATENCY

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_RCVLATENCY	1.3.0	pre	int32_t	ms	*	0..	RW	GSD

The latency value in the receiving direction of the socket. This value is only significant when SRTO_TSBPDMODE is enabled.

Default value: 120 ms in Live mode, 0 in File mode (see SRTO_TRANSTYPE).

The latency value defines the minimum receiver buffering delay before delivering an SRT data packet from a receiving SRT socket to a receiving application.

The actual value of the receiver buffering delay Ln (the negotiated latency) used on a connection is determined by the negotiation in the connection establishment (handshake exchange) phase as the maximum of the SRTO_RCVLATENCY value and the value of SRTO_PEERLATENCY set by the peer.

The general idea for the latency mechanism is to keep the time distance between two consecutive received packets the same as the time when these same packets were scheduled for sending by the sender application (or per the time explicitly declared when sending - see srt_sendmsg2 for details). This keeps any packets that have arrived earlier than their delivery time in the receiver buffer until their delivery time comes. This should compensate for any jitter in the network and provides an extra delay needed for a packet retransmission.

For detailed information on how the latency setting influences the actual packet delivery time and how this time is defined, refer to the latency documentation.

Reading the SRTO_RCVLATENCY value on a socket after the connection is established provides the actual (negotiated) latency value Ln.

The receiver's buffer must be large enough to store the L segment of the stream, i.e. L × Bitrate bytes. Refer to SRTO_RCVBUF.

The sender's buffer must be large enough to store a packet up until it is either delivered (and acknowledged) or dropped by the sender due to it becoming too late to be delivered. In other words, D × Bitrate bytes, where D is the sender's drop delay value configured with SRTO_SNDDROPDELAY.

Buffering of data packets on the receiving side makes it possible to recover from packet losses using the ARQ (Automatic Repeat Request) technique, and to deal with varying RTT times (network jitter) in the network, providing a (close to) constant end-to-end latency of the transmission.

See also SRTO_LATENCY.

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SRTO_RCVSYN

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_RCVSYN		post	bool		true		RW	GSI

When true, sets blocking mode on reading function when it's not ready to perform the operation. When false ("non-blocking mode"), the reading function will in this case report error SRT_EASYNCRCV and return immediately. Details depend on the tested entity:

On a connected socket or group this applies to a receiving function (srt_recv and others) and a situation when there are no data available for reading. The readiness state for this operation can be tested by checking the SRT_EPOLL_IN flag on the aforementioned socket or group.

On a freshly created socket or group that is about to be connected to a peer listener this applies to any srt_connect call (and derived), which in "non-blocking mode" always returns immediately. The connected state for that socket or group can be tested by checking the SRT_EPOLL_OUT flag. Note that a socket that failed to connect doesn't change the SRTS_CONNECTING state and can be found out only by testing the SRT_EPOLL_ERR flag.

On a listener socket this applies to srt_accept call. The readiness state for this operation can be tested by checking the SRT_EPOLL_IN flag on this listener socket. This flag is also derived from the listener socket by the accepted socket or group, although the meaning of this flag is effectively different.

Note that when this flag is set only on a group, it applies to a specific receiving operation being done on that group (i.e. it is not derived from the socket of which the group is a member).

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SRTO_RCVTIMEO

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_RCVTIMEO		post	int32_t	ms	-1	-1, 0..	RW	GSI

Limits the time up to which the receiving operation will block (see SRTO_RCVSYN for details), such that when this time is exceeded, it will behave as if in "non-blocking mode". The -1 value means no time limit.

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SRTO_RENDEZVOUS

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_RENDEZVOUS		pre	bool		false		RW	S

Use Rendezvous connection mode (both sides must set this and both must use the procedure of srt_bind and then srt_connect (or srt_rendezvous) to one another.

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SRTO_RETRANSMITALGO

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_RETRANSMITALGO	1.4.2	pre	int32_t		1	[0, 1]	RW	GSD

An SRT sender option to choose between two retransmission algorithms:

• 0 - aggressive retransmission algorithm (default until SRT v1.4.4), and
• 1 - efficient retransmission algorithm (introduced in SRT v1.4.2; default since SRT v1.4.4).

The aggressive retransmission algorithm causes the SRT sender to schedule a packet for retransmission each time it receives a negative acknowledgement (NAK). On a network characterized by low packet loss levels and link capacity high enough to accommodate extra retransmission overhead, this algorithm increases the chances of recovering from packet loss with a minimum delay, and may better suit end-to-end latency constraints.

The new efficient algorithm optimizes the bandwidth usage by producing fewer retransmissions per lost packet. It takes SRT statistics into account to determine if a retransmitted packet is still in flight and could reach the receiver in time, so that some of the NAK reports are ignored by the sender. This algorithm better fits general use cases, as well as cases where channel bandwidth is limited.

To learn more about the algorithms, read "Improving SRT Retransmissions — Experiments with Simulated Live Streaming (Part 1)" article.

NOTE: This option is effective only on the sending side. It influences the decision as to whether a particular reported lost packet should be retransmitted at a certain time or not.

NOTE: The efficient retransmission algorithm can only be used when a receiver sends Periodic NAK reports. See SRTO_NAKREPORT.

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SRTO_REUSEADDR

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_REUSEADDR		pre-bind	bool		true		RW	GSD

When true, allows the SRT socket to use the binding address used already by another SRT socket in the same application. Note that SRT socket uses an intermediate object called Multiplexer to access the underlying UDP sockets, so multiple SRT sockets may share one UDP socket, and the packets received by this UDP socket will be correctly dispatched to the SRT socket to which they are currently destined. This has some similarities to the SO_REUSEADDR system socket option, although it's only used inside SRT.

TODO: This option weirdly only allows the socket used in bind() to use the local address that another socket is already using, but not to disallow another socket in the same application to use the binding address that the current socket is already using. What it actually changes is that when given an address in bind() is already used by another socket, this option will make the binding fail instead of adding the socket to the shared group of that socket that already has bound this address - but it will not disallow another socket to reuse its address.

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SRTO_SENDER

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_SENDER	1.0.4	pre	bool		false		W	S

Set sender side. The side that sets this flag is expected to be a sender. This flag is only required when communicating with a receiver that uses SRT version less than 1.3.0 (and hence HSv4 handshake), in which case if not set properly, the TSBPD mode (see SRTO_TSBPDMODE) or encryption will not work. Setting SRTO_MINVERSION to 1.3.0 is therefore recommended.

This flag in versions above 1.3.0 also influences the conflict resolution for SRTO_PBKEYLEN in the case where this flag is forcefully set on both connection parties simultaneously.

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SRTO_SNDBUF

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_SNDBUF		pre-bind	int32_t	bytes	8192 bufs	*	RW	GSD+

Sender Buffer Size. See SRTO_RCVBUF for more information.

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SRTO_SNDDATA

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_SNDDATA			int32_t	pkts			R	S

Size of the unacknowledged data in send buffer.

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SRTO_SNDDROPDELAY

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_SNDDROPDELAY	1.3.2	post	int32_t	ms	*	-1..	W	GSD+

Sets an extra delay before TLPKTDROP is triggered on the data sender. This delay is added to the default drop delay time interval value. Keep in mind that the longer the delay, the more probable it becomes that packets would be retransmitted uselessly because they will be dropped by the receiver anyway.

TLPKTDROP discards packets reported as lost if it is already too late to send them (the receiver would discard them even if received). The delay before the TLPKTDROP mechanism is triggered consists of the SRT latency (SRTO_PEERLATENCY), plus SRTO_SNDDROPDELAY, plus 2 * interval between sending ACKs (where the default interval between sending ACKs is 10 milliseconds). The minimum delay is 1000 + 2 * interval between sending ACKs milliseconds.

Special value -1: Do not drop packets on the sender at all (retransmit them always when requested).

Default: 0 in Live mode, -1 in File mode.

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SRTO_SNDKMSTATE

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_SNDKMSTATE	1.2.0		int32_t	enum			R	S

Peer KM state on receiver side for SRTO_KMSTATE

Values defined in enum SRT_KM_STATE.

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SRTO_SNDSYN

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_SNDSYN		post	bool		true		RW	GSI

When true, sets blocking mode on writing function when it's not ready to perform the operation. When false ("non-blocking mode"), the writing function will in this case report error SRT_EASYNCSND and return immediately.

On a connected socket or group this applies to a sending function (srt_send and others) and a situation when there's no free space in the sender buffer, caused by inability to send all the scheduled data over the network. Readiness for this operation can be tested by checking the SRT_EPOLL_OUT flag.

On a freshly created socket or group it will have no effect until the socket enters a connected state.

On a listener socket it will be derived by the accepted socket or group, but will have no effect on the listener socket itself.

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SRTO_SNDTIMEO

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_SNDTIMEO		post	int32_t	ms	-1	-1..	RW	GSI

limit the time up to which the sending operation will block (see SRTO_SNDSYN for details), so when this time is exceeded, it will behave as if in "non-blocking mode". The -1 value means no time limit.

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SRTO_STATE

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_STATE			int32_t	enum			R	S

Returns the current socket state, same as srt_getsockstate.

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SRTO_STREAMID

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_STREAMID	1.3.0	pre	string		""	[512]	RW	GSD

• A string that can be set on the socket prior to connecting. The listener side will be able to retrieve this stream ID from the socket that is returned from srt_accept (for a connected socket with that stream ID). You usually use SET on the socket used for srt_connect, and GET on the socket retrieved from srt_accept. This string can be used completely free-form. However, it's highly recommended to follow the SRT Access Control (Stream ID) Guidlines.

• As this uses internally the std::string type, there are additional functions for it in the legacy/C++ API (udt.h): srt::setstreamid and srt::getstreamid.

• This option is not useful for a Rendezvous connection, since one side would override the value from the other side resulting in an arbitrary winner. Also in this connection both peers are known to one another and both have equivalent roles in the connection.

• IMPORTANT: This option is not derived by the accepted socket from the listener socket, and setting it on a listener socket (see srt_listen function) doesn't influence anything.

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SRTO_TLPKTDROP

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_TLPKTDROP	1.0.6	pre	bool		*		RW	GSD

Too-late Packet Drop. When enabled on receiver, it skips missing packets that have not been delivered in time and delivers the subsequent packets to the application when their time-to-play has come. It also sends a fake ACK to the sender. When enabled on sender and enabled on the receiving peer, sender drops the older packets that have no chance to be delivered in time. It is automatically enabled in sender if receiver supports it.

Default: true in Live mode, false in File mode (see SRTO_TRANSTYPE)

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SRTO_TRANSTYPE

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_TRANSTYPE	1.3.0	pre	int32_t	enum	SRTT_LIVE	*	W	S

Sets the transmission type for the socket, in particular, setting this option sets multiple other parameters to their default values as required for a particular transmission type. This sets the following options to their defaults in particular mode:

• SRTO_CONGESTION
• SRTO_MESSAGEAPI
• SRTO_NAKREPORT
• SRTO_RCVLATENCY, also set as SRTO_LATENCY
• SRTO_TLPKTDROP
• SRTO_TSBPDMODE

Values defined by enum SRT_TRANSTYPE.

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SRTO_TSBPDMODE

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_TSBPDMODE	0.0.0	pre	bool		*		W	S

When true, use Timestamp-based Packet Delivery mode. In this mode the packet's time is assigned at the sending time (or allowed to be predefined), transmitted in the packet's header, and then restored on the receiver side so that the time intervals between consecutive packets are preserved when delivering to the application.

Default: true in Live mode, false in File mode (see SRTO_TRANSTYPE).

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SRTO_UDP_RCVBUF

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_UDP_RCVBUF		pre-bind	int32_t	bytes	8192 bufs	*	RW	GSD+

UDP Socket Receive Buffer Size. Configured in bytes, maintained in packets based on MSS value. Receive buffer must not be greater than FC size.

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SRTO_UDP_SNDBUF

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_UDP_SNDBUF		pre-bind	int32_t	bytes	65536	*	RW	GSD+

UDP Socket Send Buffer Size. Configured in bytes, maintained in packets based on SRTO_MSS value.

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SRTO_VERSION

OptName	Since	Restrict	Type	Units	Default	Range	Dir	Entity
SRTO_VERSION	1.1.0		int32_t				R	S

Local SRT version. This is the highest local version supported if not connected, or the highest version supported by the peer if connected.

The version format in hex is 0x00XXYYZZ for x.y.z in human readable form. For example, version 1.4.2 is encoded as 0x010402.

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