# SRT API

The SRT C API (defined in srt.h file) is largely based in design on the legacy UDT API, with some important changes. The 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 and srt-file-transmit. All SRT related material is contained in transmitmedia.* files in the apps 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](#setup-and-teardown)

- [Creating and Destroying a Socket](#creating-and-destroying-a-socket)

- [Binding and Connecting](#binding-and-connecting)

- [Sending and Receiving](#sending-and-receiving)

- [Blocking and Non-blocking Modes](#blocking-and-non-blocking-modes) - [EPoll (Non-blocking Mode Events)](#epoll-non-blocking-mode-events)

- [Transmission Types](#transmission-types) - [Transmission Method: Live](#transmission-method-live) - [Transmission Method: Buffer](#transmission-method-buffer) - [Transmission Method: Message](#transmission-method-message)

## Setup and Teardown

Before any part of the SRT C API can be used, the user should call the 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 first have to create an SRT socket. 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

``c++ SRTSOCKET srt_create_socket(); int srt_close(SRTSOCKET s); ``

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

### Usage

``c++ sock = srt_create_socket(); ``

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

``c++ 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

1. 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.

2. The term “port” used in SRT is occasionally identical to the term “UDP port”. However SRT offers more flexibility than UDP (or TCP, 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

``c++ int srt_bind(SRTSOCKET u, const struct sockaddr* name, int namelen); int srt_bind_acquire(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 *_acquire version simply takes over the given UDP socket and copies the bound address setting from it.

``c++ int srt_listen(SRTSOCKET u, int backlog); ``

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

``c++ 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.

``c++ 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.

```c++ 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.

IMPORTANT: The connection may fail, but the socket that was used for connecting is not automatically closed and it’s also not in broken state (broken state can be only if a socket was first successfully connected and then broken). When using blocking mode, the connection failure will result in reporting an error from this function call. In non-blocking mode the connection failure is designated by the SRT_EPOLL_ERR flag set for this socket in the epoll container. After that failure you can read an extra information from the socket using srt_getrejectreason function, and then you should close the socket.

### Listener (Server) Example

```c++ 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 ) {

> int sa_len = sizeof sa; newsocket = srt_accept(sock, (sockaddr*)&sa, &sa_len); HandleNewClient(newsocket, sa);

# }

### Caller (Client) Example

```c++ sockaddr_in sa = { … }; // set target IP and port

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

### Rendezvous Example

```c++ 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

```c++ 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 which 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: unused - 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 SRT clock time. If the connection is not between SRT peers or if Timestamp-Based Packet Delivery mode (TSBPDMODE) is not enabled (see [SRT API Socket Options](API-socket-options.md)), 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

```c++ 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:

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

nb = srt_send(u, buf, nb);

SRT_MSGCTRL mc = srt_msgctrl_default; nb = srt_sendmsg2(u, buf, nb, &mc); ```

Receiving a payload:

```c++ nb = srt_recvmsg(u, buf, nb); nb = srt_recv(u, buf, nb);

SRT_MSGCTRL mc = srt_msgctrl_default; nb = srt_recvmsg2(u, buf, nb, &mc); ```

### Transmission Types Available in SRT

Mode settings determine how the sender and receiver functions work. The main SRT [socket options](API-socket-options.md) 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

See [Transmission types](#transmission-types) below.

## Blocking and Non-blocking Modes

SRT functions can also work in blocking and non-blocking modes, 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 an 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

```c++ 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); int srt_epoll_clear_usocks(int eid); ```

#### 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.

Event flags are of various categories: IN, OUT and ERR are events, which are level-triggered by default and become edge-triggered if combined with SRT_EPOLL_ET flag. The latter is only an edge-triggered flag, not an event. There’s also an SRT_EPOLL_UPDATE flag, which is an edge-triggered only event, and it reports an event on the listener socket that handles socket group new connections for an already connected group - this is for internal use only, and it’s used in the internal code for socket groups.

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 at least one event is detected 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 interruptible, 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 in the direction for which 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, making it appear as if it were ready for an operation, then when that operation occurs it will simply result in an error. You can use this as an alternative error check method.

This function also reports an 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 sockets and reports all events collected at that moment in an array with the following structure:

```c++ 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 extra srt_epoll_clear_usocks function removes all subscriptions from the epoll container.

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

## Transmission Types

NOTE: There might be a difference in terminology used in [Internet Draft](https://datatracker.ietf.org/doc/html/draft-sharabayko-srt-01) and current documentation. Please consult [Data Transmission Modes](https://tools.ietf.org/html/draft-sharabayko-srt-01#section-4.2) and [Best Practices and Configuration Tips for Data Transmission via SRT](https://datatracker.ietf.org/doc/html/draft-sharabayko-srt-01#section-7) sections of the Internet Draft additionally. The current section is going to be reworked accordingly.

SRT was originally intended to be used for Live Streaming and therefore its main and default transmission type is “live”. However, SRT supports 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](#transmission-method-live)

- [Buffer](#transmission-method-buffer)

- [Message](#transmission-method-message)

### Terminology

The following terms are used in the description of transmission types:

HANGUP / RESUME: These terms have different meanings depending on the blocking state. They describe how a particular function behaves when performing an operation requires a specific readiness condition to be satisfied.

In blocking mode HANGUP means that the function blocks until a condition is satisfied. RESUME means that the condition is satisfied and the function performs the required operation.

In non-blocking mode the only difference is that HANGUP, instead of blocking, makes the function exit immediately with an appropriate error code (such as SRT_EASYNC*, SRT_ETIMEOUT or SRT_ECONGEST) explaining why the function is not ready to perform the operation. Refer to the error descriptions in [API-functions.md](API-functions.md) for details.

The following types of operations are involved:

1. Reading data: srt_recv, srt_recvmsg, srt_recvmsg2, srt_recvfile.

> The function HANGS UP if there are no available data to read, and RESUMES when readable data become available (SRT_EPOLL_IN flag set in epoll). Use SRTO_RCVSYN to control blocking mode here.

2. Writing data: srt_send, srt_sendmsg, srt_sendmsg2, srt_sendfile.

> The function HANGS UP if the sender buffer becomes full and unable to store any additional data, and RESUMES if the data scheduled for sending have been removed from the sender buffer (after being sent and acknowledged) and there is enough free space in the sender buffer to store data (SRT_EPOLL_OUT flag set in epoll). Use SRTO_SNDSYN to control blocking mode here.

3. Accepting an incoming connection: srt_accept

> The function HANGS UP if there are no new connections reporting in, and RESUMES when a new connection has been processed and a new socket or group has been created to handle it. Note that this function requires the listener socket to get the connection (the flag SRTO_RCVSYN set on the listener socket controls the blocking mode for this operation). Note also that the blocking mode for a similar srt_accept_bond function is controlled exclusively by its timeout parameter because it can work with multiple listener sockets, potentially with different settings.

4. Connecting: srt_connect and its derivatives

> The function HANGS UP in the beginning, and RESUMES when the socket used for connecting is either ready to perform transmission operations or has failed to connect. It behaves a little differently in non-blocking mode – the function should be called only once, and it simply returns a success result as a “HANGUP”. Calling it again with the same socket would be an error. Calling it with a group would start a completely new connection. It is only possible to determine whether an operation has finished (“has RESUMED”) from epoll flags. The socket, when successfully connected, would have SRT_EPOLL_OUT set, that is, becomes ready to send data, and SRT_EPOLL_ERR when it failed to connect.

BLIND / FAST / LATE REXMIT: BLIND REXMIT is a situation where packets that were sent are still not acknowledged, either in the 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 explained below.

### Transmission Method: Live

Setting SRTO_TRANSTYPE to SRTT_LIVE sets the following [socket options](API-socket-options.md):

- [SRTO_TSBPDMODE](API-socket-options.md#SRTO_TSBPDMODE) = true

- [SRTO_RCVLATENCY](API-socket-options.md#SRTO_RCVLATENCY) = 120

- [SRTO_PEERLATENCY](API-socket-options.md#SRTO_PEERLATENCY) = 0

- [SRTO_TLPKTDROP](API-socket-options.md#SRTO_TLPKTDROP) = true

- [SRTO_LINGER](API-socket-options.md#SRTO_LINGER) = 0

- [SRTO_MESSAGEAPI](API-socket-options.md#SRTO_MESSAGEAPI) = true

- [SRTO_NAKREPORT](API-socket-options.md#SRTO_NAKREPORT) = true

- [SRTO_RETRANSMITALGO](API-socket-options.md#SRTO_RETRANSMITALGO) = 1

- [SRTO_PAYLOADSIZE](API-socket-options.md#SRTO_PAYLOADSIZE) = 1316

- [SRTO_CONGESTION](API-socket-options.md#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 (i.e. by adding extra time if the interval between two consecutive packets would otherwise be too short for the defined speed limit). Note that it is not intended 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](API-socket-options.md#SRTO_TSBPDMODE)).

You may wish to tweak some of the parameters below:

- SRTO_TSBPDMODE: You can turn off controlled latency if your application uses

its own method of latency control.

- SRTO_RCVLATENCY: You can increase the latency time, if this is

too short. Setting a shorter latency than the 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), this 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 loss reports, when the lost

packet was not recovered quickly enough, which raises suspicions that the loss report itself was lost. Without it, the loss report 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_RETRANSMITALGO: Given the receiver sends periodic NAK reports,

the sender can reduce the retransmission overhead by not retransmitting a loss more often than once per RTT (value 1).

- 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 (i.e. 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 [socket options](API-socket-options.md):

- [SRTO_TSBPDMODE](API-socket-options.md#SRTO_TSBPDMODE) = false

- [SRTO_RCVLATENCY](API-socket-options.md#SRTO_RCVLATENCY) = 0

- [SRTO_PEERLATENCY](API-socket-options.md#SRTO_PEERLATENCY) = 0

- [SRTO_TLPKTDROP](API-socket-options.md#SRTO_TLPKTDROP) = false

- [SRTO_LINGER](API-socket-options.md#SRTO_LINGER) = 180 s

- [SRTO_MESSAGEAPI](API-socket-options.md#SRTO_MESSAGEAPI) = false

- [SRTO_NAKREPORT](API-socket-options.md#SRTO_NAKREPORT) = false

- [SRTO_RETRANSMITALGO](API-socket-options.md#SRTO_RETRANSMITALGO) = 0

- [SRTO_PAYLOADSIZE](API-socket-options.md#SRTO_PAYLOADSIZE) = 0

- [SRTO_CONGESTION](API-socket-options.md#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 filled, 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 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 the 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).

Most of the parameters described above 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 setting 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. 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 on which SRT was based (UDT) somewhat misleadingly used the terms STREAM and DGRAM, and used the system symbols SOCK_STREAM and SOCK_DGRAM in the socket creation function. A “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 multiple UDP packets and has clearly defined boundaries. It’s 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.

In message mode, every sending function sends exactly as much data as it is passed in a single sending function call. The receiver also receives 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. Note that this TTL only applies to packets that have been lost and should be retransmitted.

- 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 the 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.

For more information, see [SRT API Socket Options](API-socket-options.md).

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