The SRT C API (defined in srt.h file) is largely based in design on the legacy UDT API, with some important changes. The API contained in udt.h file contains the legacy UDT API plus some minor optional functions that require the C++ standard library to be used. There are a few optional C++ API functions stored there, as there is no real C++ API for SRT. These functions may be useful in certain situations.

There are some example applications so that you can see how the API is being used, including srt-live-transmit, srt-file-transmit and srt-multiplex. All SRT related material is contained in transmitmedia.* files in the common directory which is used by all applications. See SrtSource::Read and SrtTarget::Write as examples of how data are read and written in SRT.

Setup and teardown

Before any part of the SRT C API can be used, the user should call srt_startup() function. Likewise, before the application exits, the srt_cleanup() function should be called. Note that one of the things the startup function does is to create a new thread, so choose the point of execution for these functions carefully.

Creating and destroying a socket

To do anything with SRT, you have to create an SRT socket first. The term "socket" in this case is used because of its logical similarity to system-wide sockets. An SRT socket is not directly related to system sockets, but like a system socket it is used to define a point of communication.

Synopsis

SRTSOCKET srt_socket(int af, int, int);
int srt_close(SRTSOCKET s);

The srt_socket function is based on the legacy UDT API except the first parameter. The other two are ignored.

Note that SRTSOCKET is just an alias for int; this is a legacy naming convention from UDT, which is here only for clarity.

Usage

sock = srt_socket(AF_INET, SOCK_DGRAM, 0);

This creates a socket, which can next be configured and then used for communication.

srt_close(sock);

This closes the socket and frees all its resources. Note that the true life of the socket does not end exactly after this function exits - some details are being finished in a separate "SRT GC" thread. Still, at least all shared system resources (such as listener port) should be released after this function exits.

Important Remarks

Please note that the use of SRT with AF_INET6 has not been fully tested; use at your own risk.
SRT uses the system UDP protocol as an underlying communication layer, and so it uses also UDP sockets. The underlying communication layer is used only instrumentally, and SRT manages UDP sockets as its own system resource as it pleases - so in some cases it may be reasonable for multiple SRT sockets to share one UDP socket, or for one SRT socket to use multiple UDP sockets.
The term "port" used in SRT is occasionally identical to the term "UDP port". However SRT offers more flexibility than UDP (or TCP, if we think about the more logical similarity) because it manages ports as its own resources. For example, one port may be shared between various services.

Binding and connecting

Connections are established using the same philosophy as TCP, using functions with names and signatures similar to the BSD Socket API. What is new here is the rendezvous mode.

Synopsis

int srt_bind(SRTSOCKET u, const struct sockaddr* name, int namelen);
int srt_bind_peerof(SRTSOCKET u, UDPSOCKET udpsock);

This function sets up the "sockname" for the socket, that is, the local IP address of the network device (use INADDR_ANY for using any device) and port. Note that this can be done on both listening and connecting sockets; for the latter it will define the outgoing port. If you don't set up the outgoing port by calling this function (or use port number 0), a unique port number will be selected automatically.

The *_peerof version simply copies the bound address setting from an existing UDP socket.

int srt_listen(SRTSOCKET u, int backlog);

This sets the backlog (maximum allowed simultaneously pending connections) and puts the socket into listening state -- that is, incoming connections will be accepted in the call srt_accept.

SRTSOCKET srt_accept(SRTSOCKET u, struct sockaddr* addr, int* addrlen);

This function accepts the incoming connection (the peer should do srt_connect) and returns a socket that is exclusively bound to an opposite socket at the peer. The peer's address is returned in the addr argument.

int srt_connect(SRTSOCKET u, const struct sockaddr* name, int namelen);
int srt_connect_debug(SRTSOCKET u, const struct sockaddr* name, int namelen, int forced_isn);

This function initiates the connection of a given socket with its peer's counterpart (the peer gets the new socket for this connection from srt_accept). The address for connection is passed in 'name'. The connect_debug version allows for enforcing the ISN (initial sequence number); this is used only for debugging or unusual experiments.

int srt_rendezvous(SRTSOCKET u, const struct sockaddr* local_name, int local_namelen,
    const struct sockaddr* remote_name, int remote_namelen);

A convenience function that combines the calls to bind, setting the SRTO_RENDEZVOUS flag, and connecting to the rendezvous counterpart. For simplest usage, the local_name should be set to INADDR_ANY (or a specified adapter's IP) and port. Note that both local_name and remote_name must use the same port. The peer to which this is going to connect should call the same function, with appropriate local and remote addresses. A rendezvous connection means that both parties connect to one another simultaneously.

SRT Usage - listener (server)

    sockaddr_in sa = { ... }; // set local listening port and possibly interface's IP
    int st = srt_bind(sock, (sockaddr*)&sa, sizeof sa);
    srt_listen(sock, 5);
    while ( !finish ) {
       socklen_t sa_len = sizeof sa;
       newsocket = srt_accept(sock, (sockaddr*)&sa, &sa_len);
       HandleNewClient(newsocket, sa);
    }

SRT Usage - caller (client)

    sockaddr_in sa = { ... }; // set target IP and port

    int st = srt_connect(sock, (sockaddr*)&sa, sizeof sa);
    HandleConnection(sock);

SRT Usage - rendezvous

    sockaddr_in lsa = { ... }; // set local listening IP/port
    sockaddr_in rsa = { ... }; // set remote IP/port

    srt_setsockopt(m_sock, 0, SRTO_RENDEZVOUS, &yes, sizeof yes);
    int stb = srt_bind(sock, (sockaddr*)&lsa, sizeof lsa);
    int stc = srt_connect(sock, (sockaddr*)&rsa, sizeof rsa);
    HandleConnection(sock);

or simpler

    sockaddr_in lsa = { ... }; // set local listening IP/port
    sockaddr_in rsa = { ... }; // set remote IP/port

    int stc = srt_rendezvous(sock, (sockaddr*)&lsa, sizeof lsa,
                                   (sockaddr*)&rsa, sizeof rsa);
    HandleConnection(sock);

Sending and Receiving

The SRT API for sending and receiving is split into three categories: simple, rich, and for files only.

The simple API includes: srt_send and srt_recv functions. They need only the socket and the buffer to send from or receive to, just like system read and write functions.

The rich API includes the srt_sendmsg and srt_recvmsg functions. Actually srt_recvmsg is provided for convenience and backward compatibility, as it is identical to srt_recv. The srt_sendmsg receives more parameters, specifically for messages. The srt_sendmsg2 and srt_recvmsg2 functions receive the socket, buffer, and the SRT_MSGCTRL object, which is an input-output object specifying extra data for the operation.

Functions with the msg2 suffix use the SRT_MSGCTRL object, and have the following interpretation (except flags and boundary that are reserved for future use and should be 0):

srt_sendmsg2:
- msgttl: [IN] maximum time (in ms) to wait for successful delivery (-1: indefinitely)
- inorder: [IN] if false, the later sent message is allowed to be delivered earlier
- srctime: [IN] timestamp to be used for sending (0 if current time)
- pktseq: unused
- msgno: [OUT]: message number assigned to the currently sent message
srt_recvmsg2
- msgttl, inorder: unused
- srctime: [OUT] timestamp set for this dataset when sending
- pktseq: [OUT] packet sequence number (first packet from the message, if it spans multiple UDP packets)
- msgno: [OUT] message number assigned to the currently received message

Please note that the msgttl and inorder arguments and fields in SRT_MSGCTRL are meaningful only when you use the message API in file mode (this will be explained later). In live mode, which is the SRT default, packets are always delivered when the time comes (always in order), where you don't want a packet to be dropped before sending (so -1 should be passed here).

The srctime parameter is an SRT addition for applications (i.e. gateways) forwarding SRT streams. It permits pulling and pushing of the sender's original time stamp, converted to local time and drift adjusted. The srctime parameter is the number of usec (since epoch) in local time. If the connection is not between SRT peers or if Timestamp-Based Packet Delivery mode (TSBPDMODE) is not enabled (see Options), the extracted srctime will be 0. Passing srctime = 0 in sendmsg is like using the API without srctime and the local send time will be used (if TSBPDMODE is enabled and receiver supports it).

Synopsis

int srt_send(SRTSOCKET s, const char* buf, int len);
int srt_sendmsg(SRTSOCKET s, const char* buf, int len, int msgttl, bool inorder, uint64_t srctime);
int srt_sendmsg2(SRTSOCKET s, const char* buf, int len, SRT_MSGCTRL* msgctrl);

int srt_recv(SRTSOCKET s, char* buf, int len);
int srt_recvmsg(SRTSOCKET s, char* buf, int len);
int srt_recvmsg2(SRTSOCKET s, char* buf, int len, SRT_MSGCTRL* msgctrl);

Usage

Sending a payload:

nb = srt_sendmsg(u, buf, nb, -1, true);

nb = srt_send(u, buf, nb);

SRT_MSGCTL mc = srt_msgctl_default;
nb = srt_sendmsg2(u, buf, nb, &mc);

Receiving a payload:

nb = srt_recvmsg(u, buf, nb);
nb = srt_recv(u, buf, nb);

SRT_MSGCTL mc = srt_msgctl_default;
nb = srt_recvmsg2(u, buf, nb, &mc);

Transmission Modes

Mode settings determine how the sender and receiver functions work. The main socket options (see below for full description) that control it are:

SRTO_TRANSTYPE. Sets several parameters in accordance with the selected mode:
- SRTT_LIVE (default) the Live mode (for live stream transmissions)
- SRTT_FILE the File mode (for "no time controlled" fastest data transmission)
SRTO_MESSAGEAPI
- true: (default in Live mode): use Message API
- false: (default in File mode): use Buffer API

We have then three cases (note that Live mode implies Message API):

Live mode (default)

In this mode, the application is expected to send single pieces of data that are already under sending speed control. Default size is 1316, which is 7 * 188 (MPEG TS unit size). With default settings in this mode, the receiver will be delivered payloads with the same time distances between them as when they were sent, with a small delay (default 120 ms).
File mode, Buffer API (default when set SRTT_FILE mode)

In this mode the application may deliver data with any speed and of any size. The facility will try to send them as long as there is buffer space for it. A single call for sending may send only fragments of the buffer, and the receiver will receive as much as is available and fits in the buffer.
File mode, Message API (when SRTO_TRANSTYPE is SRTT_FILE and SRTO_MESSAGEAPI is true)

In this mode the application delivers single pieces of data that have declared boundaries. The sending is accepted only when the whole message can be scheduled for sending, and the receiver will be given either the whole message, or nothing at all, including when the buffer is too small for the whole message.

The File mode and its Buffer and Message APIs are derived from UDT, just implemented in a slightly different way. This will be explained below in HISTORICAL INFO under "Transmission Method: Message".

Blocking and Non-blocking Mode

SRT functions can also work in blocking and non-blocking mode, for which there are two separate options for sending and receiving: SRTO_SNDSYN and SRTO_RCVSYN. When blocking mode is used, a function will not exit until the availability condition is satisfied; in non-blocking mode the function always exits immediately, and in case of lack of resource availability, it returns an error with appropriate code. The use of non-blocking mode usually requires using some polling mechanism, which in SRT is EPoll.

Note also that the blocking and non-blocking modes apply not only for sending and receiving. For example, SNDSYN defines blocking for srt_connect and RCVSYN defines blocking for srt_accept. The SNDSYN also makes srt_close exit only after the sending buffer is completely empty.

EPoll (Non-blocking Mode Events)

EPoll is a mechanism to track the events happening on the sockets, both "system sockets" (see SYSSOCKET type) and SRT Sockets. Note that SYSSOCKET is also an alias for int, used only for clarity.

Synopsis

int srt_epoll_update_usock(int eid, SRTSOCKET u, const int* events = NULL);
int srt_epoll_update_ssock(int eid, SYSSOCKET s, const int* events = NULL);
int srt_epoll_wait(int eid, SRTSOCKET* readfds, int* rnum, SRTSOCKET* writefds, int* wnum,
                    int64_t msTimeOut,
                    SYSSOCKET* lrfds, int* lrnum, SYSSOCKET* lwfds, int* lwnum);
int srt_epoll_uwait(int eid, SRT_EPOLL_EVENT* fdsSet, int fdsSize, int64_t msTimeOut);

SRT Usage

SRT socket being a user level concept, the system epoll (or other select) cannot be used to handle SRT non-blocking mode events. Instead, SRT provides a user-level epoll that supports both SRT and system sockets.

The srt_epoll_update_{u|s}sock() API functions described here are SRT additions to the UDT-derived srt_epoll_add_{u|s}sock() and epoll_remove_{u|s}sock() functions to atomically change the events of interest. For example, to remove SRT_EPOLL_OUT but keep SRT_EPOLL_IN for a given socket with the existing API, the socket must be removed from epoll and re-added. This cannot be done atomically, the thread protection (against the epoll thread) being applied within each function but unprotected between the two calls. It is then possible to lose an SRT_EPOLL_IN event if it fires while the socket is not in the epoll list.

Once the subscriptions are made, you can call an SRT polling function (srt_epoll_wait or srt_epoll_uwait) that will block until an event is raised on any of the subscribed sockets. This function will exit as soon as st least one event is deteted or a timeout occurs. The timeout is specified in [ms], with two special values:

0: check and report immediately (don't wait)
-1: wait indefinitely (not interruptable, even by a system signal)

There are some differences in the synopsis between these two:

srt_epoll_wait: Both system and SRT sockets can be subscribed. This function reports events on both socket types according to subscriptions, in these arrays:
- readfds and lrfds: subscribed for IN and ERR
- writefds and lwfds: subscribed for OUT and ERR

where:

- `readfds` and `writefds` report SRT sockets ("user" socket)
- `lrfds` and `lwfds` report system sockets

Note: this function provides no straightforward possibility to report sockets with an error. If you want to distinguish a report of readiness for operation from an error report, the only way is to subscribe the socket in only one direction (either SRT_EPOLL_IN or SRT_EPOLL_OUT, but not both) and SRT_EPOLL_ERR, and then check the socket's presence in the array for which's direction the socket wasn't subscribed (for example, when an SRT socket is subscribed for SRT_EPOLL_OUT | SRT_EPOLL_ERR, its presence in readfds means that an error is reported for it). This need not be a big problem because when an error is reported on a socket, an appearance as if it were ready for an operation, followed by doing this operation, will simply result in an error from that operation, so you can use it also as an alternative error check method.

This function also reports error of type SRT_ETIMEOUT when no socket is ready as the timeout elapses (including 0). This behavior is different in srt_epoll_uwait.

Note that in this function there's a loop that checks for socket readiness every 10ms. Thus, the minimum poll timeout the function can reliably support, when system sockets are involved, is also 10ms. The return time from a poll function can only be quicker when there is an event raised on one of the active SRT sockets.

srt_epoll_uwait: In this function only the SRT sockets can be subscribed (it reports error if you pass an epoll id that is subscribed to system sockets). This function waits for the first event on subscribed SRT socket and reports all events collected at this moment in an array of this structure:

typedef struct SRT_EPOLL_EVENT_
{
    SRTSOCKET fd;
    int       events;
} SRT_EPOLL_EVENT;

Every item reports a single socket with all events as flags.

When the timeout is not -1, and no sockets are ready until the timeout time passes, this function returns 0. This behavior is different in srt_epoll_wait.

The SRT EPoll system does not supports all features of Linux epoll. For example, it only supports level-triggered events for system sockets.

Options

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

Synopsis

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).

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

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 Binding column should define for these options one of the following statements concerning setting a value:

pre: For connecting a socket it must be set prior to calling srt_connect() and never changed thereafter. For a listener socket it should be set to a binding socket and it will be derived by every socket returned by srt_accept().
post: This flag can be changed any time, including after the socket is connected. On binding a socket setting this flag is effective only on this socket itself. Note though that there are some post-bound options that have important meaning when set prior to connecting.

This option list is sorted alphabetically. Note that some options can be either only a retrieved (GET) or specified (SET) value.

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_CONNTIMEO`	1.1.2	pre	`int`	msec	3000	tbd

Connect timeout. SRT cannot connect for RTT > 1500 msec (2 handshake exchanges) with the default connect timeout of 3 seconds. This option applies to the caller and rendezvous connection modes. The connect timeout is 10 times the value set for the rendezvous mode (which can be used as a workaround for this connection problem with earlier versions)

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_EVENT`		n/a	`int32_t`		n/a	n/a

[GET] - 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).

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_FC`		pre	`int`	pkts	25600	32..

Flight Flag Size (maximum number of bytes that can be sent without being acknowledged)

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_INPUTBW`	1.0.5	post	`int64_t`	bytes/s	0	0..

Sender nominal input rate. Used along with OHEADBW, when MAXBW is set to relative (0), to calculate maximum sending rate when recovery packets are sent along with main media stream (INPUTBW * (100 + OHEADBW) / 100). If INPUTBW is not set while MAXBW is set to relative (0), the actual input rate is evaluated inside the library.

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_IPTOS`	1.0.5	pre	`int32_t`		(platform default)	0..255

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.
Sender: user configurable, default: 0xB8

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_ISN`	1.3.0	post	`int32_t`	sequence	n/a	n/a

[GET] - The value of the ISN (Initial Sequence Number), which is the first sequence number put on a firstmost sent UDP packets carrying 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, not predicted in any regular development.

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_IPTTL`	1.0.5	pre	`int32_t`	hops	(platform default)	1..255

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.
Sender: user configurable, default: 64

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_IPV6ONLY`	1.4.0	pre	`int`	n/a	(platform default)	-1..1

[GET or SET] - Set system socket flag IPV6ONLY. When set to 0 a listening socket binding an IPv6 address accepts also IPv4 clients (their addresses will be formatted as IPv4-mapped IPv6 addresses). By default (-1) this option is not set and the platform default value is used.

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_KMREFRESHRATE`	1.3.2	pre	`int32_t`	pkts	0x1000000	0..unlimited

[GET or SET] - 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.

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_KMPREANNOUNCE`	1.3.2	pre	`int32_t`	pkts	0x1000	see below

[GET or SET] - 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 (i.e. the number of packets that have already left the sender but have not yet arrived at the receiver).

[GET or SET] - 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 (i.e. the number of packets that have already left the sender but have not yet arrived at the receiver).

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_KMSTATE`	1.0.2	n/a	`int32_t`		n/a	n/a

[GET] - 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 shall not be used if the application meant to use the versions at least 1.3.0 and does not use the SRTO_SENDER flag.

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_LATENCY`	1.0.2	pre	`int32_t`	msec	0	positive only

This flag sets both SRTO_RCVLATENCY and SRTO_PEERLATENCY to the same value. Note that prior to version 1.3.0 this is the only flag to set the latency, however this is effectively equivalent to setting SRTO_PEERLATENCY, when the side is sender (see SRTO_SENDER) and SRTO_RCVLATENCY when the side is receiver, and the bidirectional stream sending in version 1.2.0is not supported.

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_LINGER`		pre	linger	secs	on (180)

Linger time on close (see SO_LINGER).
SRT recommended value: off (0).

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_LOSSMAXTTL`	1.2.0	pre	`int`	packets	0	reasonable

[SET] - The value up to which the Reorder Tolerance may grow. When Reorder Tolerance is > 0, then packet loss report is delayed until that number of packets come in. Reorder Tolerance increases every time a "belated" packet has come, but it wasn't due to retransmission (that is, when UDP packets tend to come out of order), with the difference between the latest sequence and this packet's sequence, and not more than the value of this option. By default it's 0, which means that this mechanism is turned off, and the loss report is always sent immediately upon experiencing a "gap" in sequences.

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_MAXBW`	1.0.5	pre	`int64_t`	bytes/sec	-1	-1

[GET or SET] - Maximum send bandwidth.
-1: infinite (CSRTCC limit is 30mbps)
= 0: relative to input rate (SRT 1.0.5 addition, see SRTO_INPUTBW)
>0: absolute limit
SRT recommended value: 0 (relative)

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_MESSAGEAPI`	1.3.0	pre	bool	boolean	true

[SET] - When set, this socket uses the Message API[*], otherwise it uses Buffer API. Note that in live mode (see SRTO_TRANSTYPE option) there's only message API available. In File mode you can chose to use one of two modes:
- Stream API (default, when this option is false). In this mode you may send as many data as you wish with one sending instruction, or even use dedicated functions that read directly from 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 the 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 across 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 buffer as necessary to receive the message, otherwise the message will not be given up. When the message is not complete (not all packets received or there was a packet loss) it will not be given up. The messages that are sent later, but were earlier reassembled by the receiver, will be given up to the received once ready, if the inorder flag (see srt_sendmsg) was set to false.
As a comparison to the standard system protocols, the Stream API makes the transmission similar to TCP, whereas the Message API functions like the SCTP protocol.

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_MINVERSION`	1.3.0	pre	`int32_t`	version	0	up to current

[SET] - 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.

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_MSS`		pre	`int`	bytes	1500	76..

Maximum Segment Size. Used for buffer allocation and rate calculation using packet counter assuming fully filled packets. The smallest MSS between the peers is used. This is 1500 by default in the overall internet. This is the maximum size of the UDP packet and can be only decreased, unless you have some unusual dedicated network settings. Not to be mistaken 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

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_NAKREPORT`	1.1.0	pre	`bool`	true	true	false

[GET or SET] - When set to true, Receiver will send UMSG_LOSSREPORT messages periodically until the lost packet is retransmitted or intentionally dropped

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_OHEADBW`	1.0.5	post	`int`	%	25	5..100

Recovery bandwidth overhead above input rate (see SRTO_INPUTBW).
Sender: user configurable, default: 25%.
To do: set-only. get should be supported.

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_PACKETFILTER`	1.4.0	pre	string			[...512]

[SET] - Set up the packet filter. The string must match appropriate syntax for packet filter setup.

For details, see Packet Filtering & FEC.

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_PASSPHRASE`	0.0.0	pre	string		[0]	[10..79]

[SET] - Sets the passphrase for encryption. This turns encryption on on this side (or turns it off, if empty passphrase is passed).
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). It is used on the receiver only if the received data is encrypted. The configured passphrase cannot be get back (write-only). Sender and receiver: user configurable.

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_PAYLOADSIZE`	1.3.0	pre	int	bytes	1316 (Live)	up to MTUsize-28-16, usually 1456

[SET] - Sets the maximum declared size of a single call to sending function in Live mode. Use 0 if this value isn't used (which is default in file mode). This value shall not be exceeded for a single data sending instruction in Live mode

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_PBKEYLEN`	0.0.0	pre	`int32_t`	bytes	0	0 16(128/8) 24(192/8) 32(256/8)

[GET or SET] - Sender encryption key length. The use is slightly different in 1.2.0 (HSv4) and 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.
- HSv5: On the sending party it will default to 16 if not changed the default 0 and the passphrase was set. The party that has set this value to non-zero value will advertise it at the beginning of the handshake. Actually there are two methods of defining it predicted to be used and all other uses are considered an 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: use a rule in your use case that you will be setting the PBKEYLEN exclusively either on the Listener or on the Caller. Simply the value set on the Listener will win, if set.
  - Bidirectional in Rendezvous arrangement: you have to know both parties passphrases as well as PBKEYLEN and you shall set PBKEYLEN to the same value on both parties (or leave the default value on both parties, which will result in 16)
  - Unwanted behavior cases: if both parties set PBKEYLEN and the value on both sides is different, the effective PBKEYLEN will be the one that is set on the Responder party, which may also override the PBKEYLEN 32 set by the sender to value 16 if such value was used by the receiver. The Responder party is Listener in Caller-Listener arrangement, and in Rendezvous it's the matter of luck which one.
Possible values:
- 0 (PBKEYLEN not set)
- 16 (effective default) = AES-128
- 24 = AES-192
- 32 = AES-256
Sender: user configurable.

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_PEERIDLETIMEO`	1.3.3	pre	`int32_t`	msec	5000	positive only

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

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_PEERLATENCY`	1.3.0	pre	`int32_t`	msec	0	positive only

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

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_PEERVERSION`	1.1.0	n/a	`int32_t`	n/a	n/a	n/a

[GET] - Peer SRT version. 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.

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_RCVBUF`		pre	`int`	bytes	8192 × (1500-28)	32 × (1500-28) ..FC × (1500-28)

Receive Buffer Size.
Receive buffer must not be greater than FC size.
Warning: configured in bytes, converted in packets when set based on MSS value. For desired result, configure MSS first.

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_RCVDATA`		n/a	`int32_t`	pkts	n/a

[GET] - Size of the available data in the receive buffer.

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_RCVKMSTATE`	1.2.0	post	enum	n/a	n/a

[GET] - KM state on the agent side when it's a receiver, as per SRTO_KMSTATE
Values defined in enum SRT_KM_STATE:
- SRT_KM_S_UNSECURED: no decryption (even if sent data are encrypted)
- SRT_KM_S_SECURING: securing: (HSv4 only) encryption is desired, but KMX handshake not yet done, still waiting (until done, behaves like UNSECURED)
- SRT_KM_S_SECURED: KM exchange was successful and it will be decrypting encrypted data
- SRT_KM_S_NOSECRET: (HSv5 only) This site has set password, but data will be received as plain
- SRT_KM_S_BADSECRET: The password is wrong, encrypted payloads won't be decrypted.

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_RCVLATENCY`	1.3.0	pre	`int32_t`	msec	120	positive only

NB: The default live mode settings set SRTO_RCVLATENCY to 120 ms! The buffer mode settings set SRTO_RCVLATENCY to 0.
The time that should elapse since the moment when the packet was sent and the moment when it's delivered to the receiver application in the receiving function. This time should be a buffer time large enough to cover the time spent for sending, unexpectedly extended RTT time, and the time needed to retransmit the lost UDP packet. The effective latency value will be the maximum of this options' value and the value of SRTO_PEERLATENCY set by the peer side. This option in pre-1.3.0 version is available only as SRTO_LATENCY.

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_RCVSYN`		pre	`bool`	true	true	false

[GET or SET] - Synchronous (blocking) receive mode

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_RCVTIMEO`		post	`int`	msecs	-1	-1..

[GET or SET] - Blocking mode receiving timeout (-1: infinite)

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_RENDEZVOUS`		pre	`bool`	false	true	false

[GET or SET] - Use Rendezvous connection mode (both sides must set this and both must use bind/connect to one another.

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_REUSEADDR`		pre			true	true, false

When true, allows the SRT socket use the binding address used already by another SRT socket in the same application. Note that SRT socket use an intermediate object to access the underlying UDP sockets called Multiplexer, so multiple SRT socket may share one UDP socket and the packets received to this UDP socket will be correctly dispatched to the SRT socket to which they are currently destined. This has some similarities to 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 address in bind() is already used by another socket, this option will make the binding fail instead of making the socket added to the shared group of that socket that already has bound this address - but it will not disallow another socket reuse its address.

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_SENDER`	1.0.4	pre	`int32_t` bool?		false

Set sender side. The side that sets this flag is expected to be a sender. It's required when any of two connection sides supports at most HSv4 handshake, and therefore the sender side is the side that initiates the SRT extended handshake (which won't be done at all, if none of the sides sets this flag). This flag is superfluous, if both parties are at least version 1.3.0 (this shall be enforced by setting this value to SRTO_MINVERSION if you expect that it be true) and therefore support HSv5 handshake, where the SRT extended handshake is done with the overall handshake process. This flag is however obligatory if at least one party may be SRT below version 1.3.0 and does not support HSv5.

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_CONGESTION`	1.3.0	pre	`const char*`	predefined	"live"	"live" or "file"

[SET] - 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.
TODO: might be reasonable to allow an "adaptive" congestion controller, which will make the side that sets it accept whatever controller type is set by the peer, including different per connection

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_SNDBUF`		pre	`int`	bytes	8192 × (1500-28)

Send Buffer Size. Warning: configured in bytes, converted in packets, when set, based on MSS value. For desired result, configure MSS first.

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_SNDDATA`		n/a	`int32_t`	pkts	n/a	n/a

[GET] - Size of the unacknowledged data in send buffer.

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_SNDDROPDELAY`	1.3.2	pre	`int`	ms	0

[SET] - Sets an extra delay before TLPKTDROP is triggered on the data sender. TLPKTDROP discards packets reported as lost if it is already too late to send them (the receiver would discard them even if received). The total delay before TLPKTDROP is triggered consists of the LATENCY (SRTO_PEERLATENCY), plus SRTO_SNDDROPDELAY, plus 2 * the ACK interval (default = 20ms). SRTO_SNDDROPDELAY extends the tolerance for retransmitting packets at the expense of more likely retransmitting them uselessly. To be effective, it must have a value greater than 1000 - SRTO_PEERLATENCY.

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_SNDKMSTATE`	1.2.0	post	enum	n/a	n/a

[GET] - Peer KM state on receiver side for SRTO_KMSTATE
Values defined in enum SRT_KM_STATE:
- SRT_KM_S_UNSECURED: data will not be encrypted
- SRT_KM_S_SECURING: (HSv4 only): encryption is desired, but KM exchange isn't finished. Payloads will be encrypted, but the receiver won't be able to decrypt them yet.
- SRT_KM_S_SECURED: payloads will be encrypted and the receiver will decrypt them
- SRT_KM_S_NOSECRET: Encryption is desired on this side and payloads will be encrypted, but the receiver didn't set the password and therefore won't be able to decrypt them
- SRT_KM_S_BADSECRET: Encryption is configured on both sides, but the password is wrong (in HSv5 terms: both sides have set different passwords). The payloads will be encrypted and the receiver won't be able to decrypt them.

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_SNDSYN`		post	`bool`	true	true	false

[GET or SET] - Synchronous (blocking) send mode

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_SNDTIMEO`		post	`int`	msecs	-1	-1..

[GET or SET] - Blocking mode sending timeout (-1: infinite)

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_STATE`		n/a	`int32_t`		n/a	n/a

[GET] - UDT connection state. (See enum SRT_SOCKSTATUS)

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_STREAMID`	1.3.0	pre	`const char*`		empty	any string

[GET or SET] - A string limited to 512 characters that can be set on the socket prior to connecting. This stream ID will be able to be retrieved by the listener side from the socket that is returned from srt_accept and was connected by a socket with that set stream ID (so 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 guidlines.

As this uses internally the std::string type, there are additional functions for it in the legacy/C++ API (udt.h): UDT::setstreamid and UDT::getstreamid. This option doesn't make sense in Rendezvous connection; the result might be that simply one side will override the value from the other side and it's the matter of luck which one would win

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_ENFORCEDENCRYPTION`	1.3.2	pre	`int (bool)`		true	false

[SET] - This option enforces that both connection parties have the same passphrase set (including empty, that is, with no encryption), or 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.

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_TLPKTDROP`	1.0.6	pre	`int32_t` bool?	true	true	false

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.

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_TRANSTYPE`	1.3.0	pre	enum		`SRTT_LIVE`	alt: `SRTT_FILE`

[SET] - 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.
- SRTT_LIVE: Set options as for live transmission. In this mode, you should send by one sending instruction only so many data that fit in one UDP packet, and limited to the value defined first in SRTO_PAYLOADSIZE option (1316 is default in this mode). There is no speed control in this mode, only the bandwidth control, if configured, in order to not exceed the bandwidth with the overhead transmission (retransmitted and control packets).
- SRTT_FILE: Set options as for non-live transmission. See SRTO_MESSAGEAPI for further explanations

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_TSBPDMODE`	0.0.0	pre	`int32_t` (bool?)	false	true	false

Timestamp-based Packet Delivery mode. This flag is set to true by default and as a default flag set in live mode.

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_UDP_RCVBUF`		pre	`int`	bytes	8192 × 1500	MSS..

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

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_UDP_SNDBUF`		pre	`int`	bytes	65536	MSS..

UDP Socket Send Buffer Size. Configured in bytes, maintained in packets based on SRTO_MSS value. SRT recommended value: 1024*1024

OptName	Since	Binding	Type	Units	Default	Range
`SRTO_VERSION`	1.1.0	n/a	`int32_t`		n/a	n/a

[GET] - 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 0xXXYYZZ for x.y.z in human readable form, where x = ("%d", (version>>16) & 0xff), etc.
SET could eventually be supported for testing

Transmission types

SRT has been mainly created for Live Streaming and therefore its main and default transmission type is "live". SRT supports, however, the modes that the original UDT library supported, that is, file and message transmission.

There are two general modes: Live and File transmission. Inside File transmission mode, there are also two possibilities: Buffer API and Message API. The Live mode uses Message API. However it doesn't exactly match the description of the Message API because it uses a maximum single sending buffer up to the size that fits in one UDP packet.

There are two options to set a particular type:

SRTO_TRANSTYPE: uses the enum value with SRTT_LIVE for live mode and SRTT_FILE for file mode. This option actually changes several parameters to their default values for that mode. After this is done, additional parameters, including those that are set here, can be further changed.
SRTO_MESSAGEAPI: This sets the Message API (true) or Buffer API (false)

This makes possible a total of three data transmission methods:

Live
Buffer
Message

NOTE THE TERMS used below:

HANGUP and RESUME: "Function HANGS UP" means that it returns an error from the MJ_AGAIN category (see SRT_EASYNC*, SRT_ETIMEOUT and SRT_ECONGEST symbols from SRT_ERRNO enumeration type), if it's in non-blocking mode. In blocking mode it will block until the condition that caused the HANGUP no longer applies, which is defined as that the function RESUMES. In nonblocking mode, the function RESUMES when the call to it has done something and returned the non-error status. The blocking mode in SRT is separate for sending and receiving and set by SRTO_SNDSYN and SRTO_RCVSYN options respectively
BLIND REXMIT: A situation where packets that were sent are still not acknowledged, either in expected time frame, or when another ACK has come for the same number, but no packets have been reported as lost, or at least not for all still unacknowledged packets. The congestion control class is responsible for the algorithm for taking care of this situation, which is either FASTREXMIT or LATEREXMIT. This will be expained below.

Transmission method: Live

Setting SRTO_TRANSTYPE to SRTT_LIVE sets the following parameters:

SRTO_TSBPDMODE = true
SRTO_RCVLATENCY = 120
SRTO_PEERLATENCY = 0
SRTO_TLPKTDROP = true
SRTO_MESSAGEAPI = true
SRTO_NAKREPORT = true
SRTO_PAYLOADSIZE = 1316
SRTO_CONGESTION = "live"

In this mode, every call to a sending function is allowed to send only so much data, as declared by SRTO_PAYLOADSIZE, whose value is still limited to a maximum of 1456 bytes. The application that does the sending is by itself responsible for calling the sending function in appropriate time intervals between subsequent calls. By default, this implies that the receiver uses 120 ms of latency, which is the declared time interval between the moment when the packet is scheduled for sending at the sender side, and when it is received by the receiver application (that is, the data are kept in the buffer and declared as not received, until the time comes for the packet to "play").

This mode uses the LiveCC congestion control class, which puts only a slight limitation on the bandwidth, if needed, just to add extra time, if the distance between two consecutive packets would be too short for the defined speed limit. Note that it is not predicted to work with "virtually infinite" ingest speeds (such as, for example, reading directly from a file). Therefore the application is not allowed to stream data with maximum speed -- it must take care that the speed of data being sent is in rhythm with timestamps in the live stream. Otherwise the behavior is undefined and might be surprisingly disappointing.

The reading function will always return only a payload that was sent, and it will HANGUP until the time to play has come for this packet (if TSBPD mode is on) or when it is available without gaps of lost packets (if TSBPD mode is off - see SRTO_TSBPDMODE).

You may wish to tweak some of the parameters below:

SRTO_TSBPDMODE: you can turn off controlled latency, if your application uses some alternative and its own method of latency control
SRTO_RCVLATENCY: you can increase the latency time, if this is too short (setting shorter latency than default is strongly discouraged, although in some very specific and dedicated networks this may still be reasonable). Note that SRTO_PEERLATENCY is an option for the sending party, which is the minimum possible value for a receiver.
SRTO_TLPKTDROP: When true (default), it will drop the packets that haven't been retransmitted on time, that is, before the next packet that is already received becomes ready to play. You can turn this off to always ensure a clean delivery. However, a lost packet can simply pause a delivery for some longer, potentially undefined time, and cause even worse tearing for the player. Setting higher latency will help much more in the case when TLPKTDROP causes packet drops too often.
SRTO_NAKREPORT: Turns on repeated sending of lossreport, when the lost packet was not recovered quickly enough, which raises suspicions that the lossreport itself was lost. Without it, the lossreport will be always reported just once and never repeated again, and then the lost payload packet will be probably dropped by the TLPKTDROP mechanism.
SRTO_PAYLOADSIZE: Default value is for MPEG TS; if you are going to use SRT to send any different kind of payload, such as, for example, wrapping a live stream in very small frames, then you can use a bigger maximum frame size, though not greater than 1456 bytes.

Parameters from the modified for transmission type list, not mentioned in the list above, are crucial for Live mode and shall not be changed.

The BLIND REXMIT situation is resolved using the FASTREXMIT algorithm by LiveCC: sending non-acknowledged packets blindly on the premise that the receiver lingers too long before acknowledging them. This mechanism isn't used (that is, the BLIND REXMIT situation isn't handled at all) when SRTO_NAKREPORT is set by the peer -- the NAKREPORT method is considered so effective that FASTREXMIT isn't necessary.

Transmission method: Buffer

Setting SRTO_TRANSTYPE to SRTT_FILE sets the following parameters:

SRTO_TSBPDMODE = false
SRTO_RCVLATENCY = 0
SRTO_PEERLATENCY = 0
SRTO_TLPKTDROP = false
SRTO_MESSAGEAPI = false
SRTO_NAKREPORT = false
SRTO_PAYLOADSIZE = 0
SRTO_CONGESTION = "file"

In this mode, calling a sending function is allowed to potentially send virtually any size of data. The sending function will HANGUP only if the sending buffer is completely replete, and RESUME if the sending buffers are available for at least one smallest portion of data passed for sending. The sending function need not send everything in this call, and the caller must be aware that the sending function might return sent data of smaller size than was actually requested.

From the receiving function there will be retrieved as many data as the minimum of the passed buffer size and available data; data still available and not retrieved by this call will be available for retrieval in the next call.

There is also a dedicated pair of functions that can only be used in this mode: srt_sendfile and srt_recvfile. These functions can be used to transmit the whole file, or a fragment of it, based on the offset and size.

This mode uses the FileCC congestion control class, which is a direct copy of the UDT's CUDTCC congestion control class, adjusted to the needs of SRT's congestion control framework. This class generally sends the data with maximum speed in the beginning, until the flight window is full, and then keeps the speed at the edge of the flight window, only slowing down in the case where packet loss was detected. The bandwidth usage can be directly limited by SRTO_MAXBW option.

The BLIND REXMIT situation is resolved in FileCC using the LATEREXMIT algorithm: when the repeated ACK was received for the same packet, or when the loss list is empty and the flight window is full, all packets since the last ACK are sent again (that's more or less the TCP behavior, but in contrast to TCP, this is done as a very low probability fallback).

As you can see in the parameters described above, most have false or 0 values as they usually designate features used in Live mode. None are used with File mode. The only option that makes sense to modify after the SRTT_FILE type was set is SRTO_MESSAGEAPI, which is described below.

Transmission method: Message

Setting SRTO_TRANSTYPE to SRTT_FILE and then SRTO_MESSAGEAPI to true implies usage of the Message transmission method. Parameters are set as described above for the Buffer method, with the exception of SRTO_MESSAGEAPI, and the "file" congestion controller is also used in this mode. It differs from the Buffer method, however, in terms of the rules concerning sending and receiving.

HISTORICAL INFO: The library that SRT was based on, UDT, somewhat misleadingly used the terms STREAM and DGRAM, and used the system symbols SOCK_STREAM and SOCK_DGRAM in the socket creation function. The "datagram" in the UDT terminology has nothing to do with the "datagram" term in networking terminology, where its size is limited to as much it can fit in one MTU. In UDT it is actually a message, which may span through multiple UDP packets and has clearly defined boundaries. It's something rather similar to the SCTP protocol. Also, in UDP the API functions were strictly bound to DGRAM or STREAM mode: UDT::send/UDT::recv were only for STREAM and UDT::sendmsg/UDT::recvmsg only for DGRAM. In SRT this is changed: all functions can be used in all modes, except srt_sendfile/srt_recvfile, and how the functions actually work is controlled by the SRTO_MESSAGEAPI flag.

The message mode means that every sending function sends exactly as much data as it is passed in a single sending function call, and the receiver receives also not less than exactly the number of bytes that was sent (although every message may have a different size). Every message may also have extra parameters:

TTL defines how much time (in ms) the message should wait in the sending buffer for the opportunity to be picked up by the sender thread and sent over the network; otherwise it is dropped.
INORDER, when true, means the messages must be read by the receiver in exactly the same order in which they were sent. In the situation where a message suffers a packet loss, this prevents any subsequent messages from achieving completion status prior to recovery of the preceding message.

The sending function will HANGUP when the free space in the sending buffer does not exactly fit the whole message, and it will only RESUME if the free space in the sending buffer grows up to this size. The call to the sending function also returns with an error, when the size of the message exceeds the total size of the buffer (this can be modified by SRTO_SNDBUF option). In other words, it is not designed to send just a part of the message -- either the whole message is sent, or nothing at all.

The receiving function will HANGUP until the whole message is available for reading; if the message spans multiple UDP packets, then the function RESUMES only when every single packet from the message has been received, including recovered packets, if any. When the INORDER flag is set to false and parts of multiple messages are currently available, the first message that is complete (possibly recovered) is returned. Otherwise the function does a HANGUP until the next message is complete. The call to the receiving function is rejected if the buffer size is too small for a single message to fit in it.

Note that you can use any of the sending and receiving functions for sending and receiving messages, except sendfile/recvfile, which are dedicated exclusively for Buffer API.

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API.md

API.md

Setup and teardown

Creating and destroying a socket

Synopsis

Usage

Important Remarks

Binding and connecting

Synopsis

SRT Usage - listener (server)

SRT Usage - caller (client)

SRT Usage - rendezvous

Sending and Receiving

Synopsis

Usage

Transmission Modes

Blocking and Non-blocking Mode

EPoll (Non-blocking Mode Events)

Synopsis

SRT Usage

Options

Synopsis

Transmission types

Transmission method: Live

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Transmission method: Message

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Setup and teardown

Creating and destroying a socket

Synopsis

Usage

Important Remarks

Binding and connecting

Synopsis

SRT Usage - listener (server)

SRT Usage - caller (client)

SRT Usage - rendezvous

Sending and Receiving

Synopsis

Usage

Transmission Modes

Blocking and Non-blocking Mode

EPoll (Non-blocking Mode Events)

Synopsis

SRT Usage

Options

Synopsis

Transmission types

Transmission method: Live

Transmission method: Buffer

Transmission method: Message