diff --git a/docs/API.md b/docs/API.md index 48e522ad6..182ecc41a 100644 --- a/docs/API.md +++ b/docs/API.md @@ -1,33 +1,59 @@ +# 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) + - [Synopsis](#synopsis) + - [Usage](#usage) + - [Important Remarks](#important-remarks) + - [Binding and connecting](#binding-and-connecting) + - [Synopsis](#synopsis) + - [SRT Usage - listener (server)](#srt-usage---listener-server) + - [SRT Usage - rendezvous](#srt-usage---rendezvous) + - [Sending and Receiving](#sending-and-receiving) + - [Synopsis](#synopsis) + - [Usage](#usage) + - [Transmission types available in SRT](#transmission-types-available-in-srt) + - [Blocking and Non-blocking Mode](#blocking-and-non-blocking-mode) + - [EPoll (Non-blocking Mode Events)](#epoll-non-blocking-mode-events)) + - [Synopsis](#synopsis) + - [SRT Usage](#srt-usage) + - [Transmission types](#transmission-types) + - [Terminology](#terminology) + - [Transmission method: Live](#transmission-method-live) + - [Transmission method: Buffer](#transmission-method-buffer) + - [Transmission method: Message](#transmission-method-message) + + +**NOTE**: The socket option descriptions originally contained in this document +have been moved to [APISocketOptions.md](https://github.com/Haivision/srt/blob/master/docs/APISocketOptions.md) -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()` +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. +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 +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 @@ -62,23 +88,23 @@ Important Remarks 1. Please note that the use of SRT with `AF_INET6` has not been fully tested; use at your own risk. -2. 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 +2. 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. 3. 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. +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. +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 -------- @@ -92,13 +118,13 @@ this can be done on both listening and connecting sockets; for the latter it wil 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. +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 +puts the socket into a listening state -- that is, incoming connections will be accepted in the call `srt_accept`. SRTSOCKET srt_accept(SRTSOCKET u, struct sockaddr* addr, int* addrlen); @@ -174,51 +200,53 @@ or simpler Sending and Receiving ===================== -The SRT API for sending and receiving is split into three categories: simple, -rich, and for files only. +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 **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 +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. +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 +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 + * `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). + * `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 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). +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 [APISocketOptions.md](https://github.com/Haivision/srt/blob/master/docs/APISocketOptions.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 @@ -257,8 +285,9 @@ Receiving a payload: Transmission types available in SRT ----------------------------------- -Mode settings determine how the sender and receiver functions work. -The main socket options (see below for full description) that control it are: +Mode settings determine how the sender and receiver functions work. The main +[socket options](APISocketOptions.md) +that control it are: * `SRTO_TRANSTYPE`. Sets several parameters in accordance with the selected mode: @@ -268,8 +297,7 @@ mode: * true: (default in Live mode): use Message API * false: (default in File mode): use Buffer API -See below [Transmission types](#transmission-types). - +See [Transmission types](#transmission-types) below. Blocking and Non-blocking Mode @@ -278,14 +306,14 @@ 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 +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 +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` +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. @@ -330,17 +358,17 @@ 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 connection for an already connected group - this is for internal use -only and it's used in the internal code for socket groups. +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 st least one event is deteted or a timeout occurs. The timeout is +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 interruptable, even by a system signal) + - 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: @@ -356,20 +384,20 @@ where: - `readfds` and `writefds` report SRT sockets ("user" socket) - `lrfds` and `lwfds` report system sockets -Note: this function provides no straightforward possibility to report +**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 +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, 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. +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 error of type `SRT_ETIMEOUT` when no socket is +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`. @@ -382,8 +410,8 @@ SRT sockets. 2. `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: +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: ``` typedef struct SRT_EPOLL_EVENT_ @@ -405,1210 +433,103 @@ 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. -Options -======= - -There's a general method of setting options on a socket in the SRT C API, similar -to the system setsockopt/getsockopt functions. - -Types used in socket options ----------------------------- - -Possible types of socket options are: - -* `int32_t` - this type can usually be treated as an `int` equivalent since -this type does not change size on 64-bit systems. For clarity, options use -this fixed size integer. In some cases the value is expressed using an -enumeration type (see below). - -* `int64_t` - Some options need the paramter specified as 64-bit integer - -* `bool` - Requires the use of a boolean type (`` for C, or built-in -for C++). When setting an option, passing the value through an `int` type is -also properly recognized. When getting an option, however, you should use the -`bool` type, although you can risk passing a variable of `int` type initialized -with 0 and then check if the resulting value is equal to 0 (just don't compare -the result with 1). - -* `string` - When setting an option, pass the character array pointer as value -and the string length as length. When getting, pass an array of sufficient size -(as specified in the size variable). Every option with this type that can be -read should specify the maximum length of that array. - - -Enumeration types used in options ---------------------------------- - - -### 1. `SRT_TRANSTYPE` - -Used by `SRTO_TRANSTYPE` option: - -* `SRTT_LIVE`: Live mode. -* `SRTT_FILE`: File mode. - -See below [Transmission types](#transmission-types) for details. - - -### 2. `SRT_KM_STATE` - -The defined encryption state as performed by the Key Material Exchange, used -by `SRTO_RCVKMSTATE`, `SRTO_SNDKMSTATE` and `SRTO_KMSTATE` options: - -- `SRT_KM_S_UNSECURED`: no encryption/descryption. If this state is only on -the receiver, received encrypted packets will be dropped. - -- `SRT_KM_S_SECURING`: pending security (HSv4 only). This is a temporary state -used only if the connection uses HSv4 and the Key Material Exchange is -not finished yet. On HSv5 this is not possible because the Key Material -Exchange for the initial key is done in the handshake. - -- `SRT_KM_S_SECURED`: KM exchange was successful and the data will be sent -encrypted and will be decrypted by the receiver. This state is only possible on -both sides in both directions simultaneously. - -- `SRT_KM_S_NOSECRET`: If this state is in the sending direction -(`SRTO_SNDKMSTATE`), then it means that the sending party has set a -passphrase, but the peer did not. In this case the sending party can receive -unencrypted packets from the peer, but packets it sends to the peer will be -encrypted and the peer will not be able to decrypt them. -This state is only possible in HSv5. - -- `SRT_KM_S_BADSECRET`: The password is wrong (set differently on each party); -encrypted payloads won't be decrypted in either direction. - -Note that with the default value of `SRTO_ENFORCEDENCRYPTION` option (true), -the state is equal on both sides in both directions, and it can be only -`SRT_KM_S_UNSECURED` or `SRT_KM_S_SECURED` (in other cases the connection -is rejected). Otherwise it may happen that either both sides have different -password and the state is `SRT_KM_S_BADSECRET` in both directions, or only -one party has set a password, in which case the KM state is as follows: - -| | `SRTO_RCVKMSTATE` | `SRTO_SNDKMSTATE` | -|--------------------------|----------------------|----------------------| -| Party with no password: | `SRT_KM_S_NOSECRET` | `SRT_KM_S_UNSECURED` | -| Party with password: | `SRT_KM_S_UNSECURED` | `SRT_KM_S_NOSECRET` | - - -Getting and setting options ---------------------------- - -Legacy version: - - int srt_getsockopt(SRTSOCKET socket, int level, SRT_SOCKOPT optName, void* optval, int& optlen); - int srt_setsockopt(SRTSOCKET socket, int level, SRT_SOCKOPT optName, const void* optval, int optlen); - -New version: - - int srt_getsockflag(SRTSOCKET socket, SRT_SOCKOPT optName, void* optval, int& optlen); - int srt_setsockflag(SRTSOCKET socket, SRT_SOCKOPT optName, const void* optval, int optlen); - -(In the legacy version, there's an additional unused `level` parameter. It was there -in the original UDT API just to mimic the system `setsockopt` function, but it's ignored). - -Some options require a value of type bool and some others of an integer type, -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 `int32_t` and `bool` -types, but this is not so in the case of getting an option value. - -**UDT project legacy note**: Almost all options from the UDT library are -derived (there are a few deleted, including some deprecated already in UDT). -Many new SRT options have been added. All options are available exclusively -with the `SRTO_` prefix. Old names are provided as alias names in the `udt.h` -legacy/C++ API file. Note the translation rules: -* `UDT_` prefix from UDT options was changed to the prefix `SRTO_` -* `UDP_` prefix from UDT options was changed to the prefix `SRTO_UDP_` -* `SRT_` prefix in older SRT versions was changed to `SRTO_` - -The table further below shows the characteristics of the options, according -to the following legend: - -1. **Since** - -Defines the SRT version when this option was first introduced. If this field -is empty, it's an option derived from UDT. "Version 0.0.0" is the oldest version -of SRT ever created and put into use. - -2. **Binding** - -Defines limitation on setting the option (the field is empty if the option -is not settable, see **Dir** column): - -* pre: For a connecting socket (both as caller and rendezvous) it must be set -prior to calling `srt_connect()` or `srt_bind()` and never changed thereafter. -For a listener socket it should be set to a listening 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 (as well as on an accepted socket). On a listening 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. - -3. **Type** - -The data type of the option; see above. - -4. **Units** - -Roughly specified unit, if the value defines things like length or time. -It can also define more precisely what kind of specialization can be used -when the type is integer: - -* enum: the possible values are defined in an enumeration type -* flags: the integer value is a collection of bit flags - -5. **Default** - -The exact default value, if it can be easily specified. For a more complicated -state of the default state of a particular option, it will be explained in the -description (when marked by asterisk). For non-settable options this field is -empty. - -6. **Range** - -If a value of an integer type has a limited range, or only specified value -allowed, it will be specified here, otherwise empty. A ranged value can be -specified as: - -* X-... : specifies only a minimum value -* X-Y,Z : values between X and Y are allowed, and additionally Z - -If the value is of `string` type, this field will contain its maximum size -in square brackets. - -If the range contains additionally an asterisk, it means that more elaborate -restrictions on the value apply, as explained in the description. - -7. **Dir** - -Option direction: W if can be set, R if can be retrieved, RW if both. - -6. **Entity** - -This describes whether the option can be set on the socket or the group. -The G and S options may appear together, in which case both possibilities apply. -The D and I options, mutually exclusive, appear always with G. -The + marker can only coexist with GS. - -Possible specifications are: - -* S: This option can be set on a single socket (exclusively, if not GS) - -* G: This option can be set on a group (exclusively, if not GS) - -* D: If set on a group, it will be derived by the member socket - -* I: If set on a group, it will be taken and managed exclusively by the group - -* +: This option is also allowed to be set individually on a group member -socket through a configuration object in `SRT_SOCKGROUPCONFIG` prepared by -`srt_create_config`. Note that this setting may override the setting derived -from the group. - -This option list is sorted alphabetically. - -| OptName | Since | Binding | Type | Units | Default | Range | Dir |Entity| -| --------------------- | ----- | ------- | -------- | ------ | -------- | ------ |-----|------| -| `SRTO_BINDTODEVICE` | 1.5.0 | pre | `string` | | | | RW | GSD+ | - -- Refers to the `SO_BINDTODEVICE` system socket option for `SOL_SOCKET` level. -This effectively limits the packets received by this socket to only those -that are targeted to that device. The device is specified by name passed as -string. The setting becomes effective after binding the socket (including -default-binding when connecting). - -- NOTE: This option is only available on Linux and available there by default. -On all other platforms setting this option will always fail. - -- NOTE: With the default system configuration, this option is only available -for a process that runs as root. Otherwise the function that applies the setting -(`srt_bind`, `srt_connect` etc.) will fail. - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| ------------------ | ----- | ------- | --------- | ------ | -------- | ------ | --- | ------ | -| `SRTO_CONNTIMEO` | 1.1.2 | pre | `int32_t` | msec | 3000 | 0.. | W | GSD+ | - -- Connect timeout. This option applies to the caller and rendezvous connection -modes. For the rendezvous mode (see `SRTO_RENDEZVOUS`) the effective connection timeout -will be 10 times the value set with `SRTO_CONNTIMEO`. - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| ----------------- | ----- | ------- | --------- | ------ | -------- | ------ | --- | ------ | -| `SRTO_DRIFTTRACER`| 1.5.0 | post | `bool` | | true | | RW | GSD | - -- Enables or disables time drift tracer (receiver). - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| ----------------- | ----- | ------- | --------- | ------ | -------- | ------ | --- | ------ | -| `SRTO_EVENT` | | | `int32_t` | flags | | | R | S | - -- Returns bit flags set according to the current active events on the socket. -- Possible values are those defined in `SRT_EPOLL_OPT` enum (a combination of -`SRT_EPOLL_IN`, `SRT_EPOLL_OUT` and `SRT_EPOLL_ERR`). - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| ----------------- | ----- | ------- | --------- | ------ | -------- | ------ | --- | ------ | -| `SRTO_FC` | | pre | `int32_t` | pkts | 25600 | 32.. | RW | GSD | - -- Flight Flag Size (maximum number of bytes that can be sent without -being acknowledged) - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| -------------------- | ----- | ------- | --------- | ------ | -------- | ------ | --- | ------ | -| `SRTO_GROUPCONNECT` | 1.5.0 | pre | `int32_t` | | 0 | 0...1 | W | S | - -- When this flag is set to 1 on a listener socket, it allows this socket to -accept group connections. When set to the default 0, group connections will be -rejected. Keep in mind that if the `SRTO_GROUPCONNECT` flag is set to 1 (i.e. -group connections are allowed) `srt_accept` may return a socket **or** a group -ID. A call to `srt_accept` on a listener socket that has group connections -allowed must take this into consideration. It's up to the caller of this -function to make this distinction and to take appropriate action depending on -the type of entity returned. - -- When this flag is set to 1 on an accepted socket that is passed to the -listener callback handler, it means that this socket is created for a group -connection and it will become a member of a group. Note that in this case -only the first connection within the group will result in reporting from -`srt_accept` (further connections are handled in the background), and this -function will return the group, not this socket ID. - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| -------------------- | ----- | ------- | ---------- | ------ | -------- | ------ | --- | ------ | -| `SRTO_GROUPTYPE` | | pre | `int32_t` | enum | | | R | S | - -- This option is read-only and it is intended to be called inside the listener -callback handler (see `srt_listen_callback`). Possible values are defined in -the `SRT_GROUP_TYPE` enumeration type. - -- This option returns the group type that is declared in the incoming connection. -If the incoming connection is not going to make a group-member connection, then -the value returned is `SRT_GTYPE_UNDEFINED`. If this option is read in any other -context than inside the listener callback handler, the value is undefined. - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| --------------------- | ----- | ------- | ---------- | ------ | -------- | ------ | --- | ------ | -| `SRTO_GROUPSTABTIMEO` | | pre | `int32_t` | ms | 40 | 10-... | W | GSD+ | - -- This setting is used for groups of type `SRT_GTYPE_BACKUP`. It defines the stability -timeout, which is the maximum interval between two consecutive packets retrieved from -the peer on the currently active link. These two packets can be of any type, -but this setting usually refers to control packets while the agent is a sender. -Idle links exchange only keepalive messages once per second, so they do not -count. Note that this option is meaningless on sockets that are not members of -the Backup-type group. - -- This value should be set with a thoroughly selected balance and correspond to -the maximum stretched response time between two consecutive ACK messages. By default -ACK messages are sent every 10ms (so this interval is not dependent on the network -latency), and so should be the interval between two consecutive received ACK -messages. Note, however, that the network jitter on the public internet causes -these intervals to be stretched, even to multiples of that interval. Both large -and small values of this option have consequences: - -- Large values of this option prevent overreaction on highly stretched response -times, but introduce a latency penalty - the latency must be greater -than this value (otherwise switching to another link won't preserve -smooth signal sending). Large values will also contribute to higher packet -bursts sent at the moment when an idle link is activated. - -- Smaller values of this option respect low latency requirements very -well, may cause overreaction on even slightly stretched response times. This is -unwanted, as a link switch should ideally happen only when the currently active -link is really broken, as every link switch costs extra overhead (it counts -for 100% for a time of one ACK interval). - -- Note that the value of this option is not allowed to exceed the value of -`SRTO_PEERIDLETIMEO`. Usually it is only meaningful if you change the latter -option, as the default value of it is way above any sensible value of -`SRTO_GROUPSTABTIMEO`. - - ---- -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| ---------------- | ----- | ------- | ---------- | ------ | -------- | ------ | --- | ------ | -| `SRTO_INPUTBW` | 1.0.5 | post | `int64_t` | B/s | 0 | 0.. | W | GSD | - -- This option is effective only if `SRTO_MAXBW` is set to 0 (relative). It -controls the maximum bandwidth together with `SRTO_OHEADBW` option according -to the formula: `MAXBW = INPUTBW * (100 + OHEADBW) / 100`. When this option -is set to 0 (automatic) then the real INPUTBW value will be estimated from -the rate of the input (cases when the application calls the `srt_send*` -function) during transmission. - -- *Recommended: set this option to the predicted bitrate of your live stream -and keep default 25% value for `SRTO_OHEADBW`*. - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| ---------------- | ----- | ------- | ---------- | ------ | -------- | ------ | --- | ------ | -| `SRTO_IPTOS` | 1.0.5 | pre | `int32_t` | | (system) | 0..255 | RW | GSD | - -- IPv4 Type of Service (see `IP_TOS` option for IP) or IPv6 Traffic Class (see `IPV6_TCLASS` -of IPv6) depending on socket address family. Applies to sender only. - -- When getting, the returned value is the user preset for non-connected sockets and the actual -value for connected sockets. - -- *Sender: user configurable, default: 0xB8* - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| ---------------- | ----- | ------- | ---------- | ------ | -------- | ------ | --- | ------ | -| `SRTO_ISN` | 1.3.0 | | `int32_t` | | | | R | S | - -- The value of the ISN (Initial Sequence Number), which is the first sequence - number put on 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 | Dir | Entity | -| ---------------- | ----- | ------- | ---------- | ------ | -------- | ------ | --- | ------ | -| `SRTO_IPTTL` | 1.0.5 | pre | `int32_t` | hops | (system) | 1..255 | RW | GSD | - -- IPv4 Time To Live (see `IP_TTL` option for IP) or IPv6 unicast hops (see -`IPV6_UNICAST_HOPS` for IPV6) depending on socket address family. Applies to sender only. - -- When getting, the returned value is the user preset for non-connected sockets and the actual -value for connected sockets. - -- *Sender: user configurable, default: 64* - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| ---------------- | ----- | ------- | ---------- | ------ | -------- | ------ | --- | ------ | -| `SRTO_IPV6ONLY` | 1.4.0 | pre | `int32_t` | | (system) | -1..1 | RW | GSD | - -- Set system socket flag `IPV6_V6ONLY`. 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 | Dir | Entity | -| --------------------- | ----- | ------- | ---------- | ------ | -------- | ------ | --- | ------ | -| `SRTO_KMREFRESHRATE` | 1.3.2 | pre | `int32_t` | pkts | 0x1000000| 0.. | RW | GSD | - -- The number of packets to be transmitted after which the Stream Encryption Key -(SEK), used to encrypt packets, will be switched to the new one. Note that -the old and new keys live in parallel for a certain period of time (see -`SRTO_KMPREANNOUNCE`) before and after the switchover. - -- Having a preannounce period before switchover ensures the new SEK is installed -at the receiver before the first packet encrypted with the new SEK is received. -The old key remains active after switchover in order to decrypt packets that -might still be in flight, or packets that have to be retransmitted. - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| --------------------- | ----- | ------- | ---------- | ------ | -------- | ------ | --- | ------ | -| `SRTO_KMPREANNOUNCE` | 1.3.2 | pre | `int32_t` | pkts | 0x1000 | 0.. * | RW | GSD | - -- The interval (defined in packets) between when a new Stream Encrypting Key -(SEK) is sent and when switchover occurs. This value also applies to the -subsequent interval between when switchover occurs and when the old SEK is -decommissioned. - -At `SRTO_KMPREANNOUNCE` packets before switchover the new key is sent -(repeatedly, if necessary, until it is confirmed by the receiver). - -At the switchover point (see `SRTO_KMREFRESHRATE`), the sender starts -encrypting and sending packets using the new key. The old key persists in case -it is needed to decrypt packets that were in the flight window, or -retransmitted packets. - -The old key is decommissioned at `SRTO_KMPREANNOUNCE` packets after switchover. - -The allowed range for this value is between 1 and half of the current value of -`SRTO_KMREFRESHRATE`. The minimum value should never be less than the flight -window (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 | Dir | Entity | -| --------------------- | ----- | ------- | ---------- | ------ | -------- | ------ | --- | ------ | -| `SRTO_KMSTATE` | 1.0.2 | | `int32_t` | enum | | | R | S | - -- Keying Material state. This is a legacy option that is equivalent to -`SRTO_SNDKMSTATE`, if the socket has set `SRTO_SENDER` to true, and -`SRTO_RCVKMSTATE` otherwise. This option is then equal to `SRTO_RCVKMSTATE` -always if your application disregards possible cooperation with a peer older -than 1.3.0, but then with the default value of `SRTO_ENFORCEDENCRYPTION` the -value returned by both options is always the same. See [SRT_KM_STATE](#srt-km-state) -for more details. - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| --------------------- | ----- | ------- | ---------- | ------ | -------- | ------ | --- | ------ | -| `SRTO_LATENCY` | 1.0.2 | pre | `int32_t` | ms | 120 * | 0.. | RW | GSD | - -- 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 1.2.0 was not supported. - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| -------------------- | ----- | ------- | ---------- | ------ | -------- | ------ | --- | ------ | -| `SRTO_LINGER` | | pre | linger | s | on, 180 | 0.. | RW | GSD | - -- Linger time on close (see [SO\_LINGER](http://man7.org/linux/man-pages/man7/socket.7.html)). - -- *SRT recommended value: off (0)*. - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| -------------------- | ----- | ------- | ---------- | ------- | -------- | ------ | --- | ------ | -| `SRTO_LOSSMAXTTL` | 1.2.0 | pre | `int32_t` | packets | 0 | 0.. | RW | GSD+ | - -- The value up to which the *Reorder Tolerance* may grow. The *Reorder Tolerance* -is the number of packets that must follow the experienced "gap" in sequence numbers -of incoming packets so that the loss report is sent (in a hope that the gap is due -to packet reordering rather than because of loss). The value of *Reorder Tolerance* -starts from 0 and is set to a greater value, when packet reordering is detected, that -is, when a "belated" packet (with sequence number older than the latest received) -has been received, but without retransmission flag. When this is detected the -*Reorder Tolerance* is set to the value of the interval between latest sequence -and this packet's sequence, but not more than the value set by `SRTO_LOSSMAXTTL`. -By default this value is set to 0, which means that this mechanism is off. - - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| -------------------- | ----- | ------- | ---------- | ------- | -------- | ------ | --- | ------ | -| `SRTO_MAXBW` | 1.0.5 | pre | `int64_t` | B/s | -1 | -1.. | RW | GSD | - -- Maximum send bandwidth. -- `-1`: infinite (the limit in Live Mode is 1Gbps) -- `0`: relative to input rate (see `SRTO_INPUTBW`) -- `>0`: absolute limit in B/s - -- *NOTE: This option has a default value of -1, regardless of the mode. -For live streams it is typically recommended to set the value 0 here and rely -on `SRTO_INPUTBW` and `SRTO_OHEADBW` options. However, if you want to do so, -you should make sure that your stream has a fairly constant bitrate, or that -changes are not abrupt, as high bitrate changes may work against the -measurement. SRT cannot ensure that this is always the case for a live stream, -therefore the default -1 remains even in live mode.* - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| -------------------- | ----- | ------- | ---------- | ------- | -------- | ------ | --- | ------ | -| `SRTO_MESSAGEAPI` | 1.3.0 | pre | bool | | true | | W | GSD | - -- When set, this socket uses the Message API[\*], otherwise it uses -Stream API. Note that in live mode (see `SRTO_TRANSTYPE` option) only the -message API is available. In File mode you can chose to use one of two modes -(note that the default for this option is changed with `SRTO_TRANSTYPE` -option): - - - Stream API (default for file mode). In this mode you may send - as many data as you wish with one sending instruction, or even use dedicated - functions that operate directly on a file. The internal facility will take care - of any speed and congestion control. When receiving, you can also receive as - many data as desired, the data not extracted will be waiting for the next call. - There is no boundary between data portions in 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 | Dir | Entity | -| -------------------- | ----- | ------- | ---------- | ------- | -------- | ------ | --- | ------ | -| `SRTO_MINVERSION` | 1.3.0 | pre | `int32_t` | version | 0 | * | W | GSD | - -- The minimum SRT version that is required from the peer. A connection to a -peer that does not satisfy the minimum version requirement will be rejected. -Format is explained at `SRTO_VERSION` option. - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| -------------------- | ----- | ------- | ---------- | ------- | -------- | ------ | --- | ------ | -| `SRTO_MSS` | | pre | `int32_t` | bytes | 1500 | 76.. | RW | GSD | - -- 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 | Dir | Entity | -| -------------------- | ----- | ------- | ---------- | ------- | -------- | ------ | --- | ------ | -| `SRTO_NAKREPORT` | 1.1.0 | pre | `bool` | | * | | RW | GSD+ | - -- When set to true, every report for a detected loss will be repeated when the -timeout for the expected retransmission of this loss has expired and the -missing packet still wasn't recovered, or wasn't conditionally dropped (see -`SRTO_TLPKTDROP`). - -- The default is true for Live mode and false for File mode (see `SRTO_TRANSTYPE`). - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| -------------------- | ----- | ------- | ---------- | ------- | -------- | ------ | --- | ------ | -| `SRTO_OHEADBW` | 1.0.5 | post | `int32_t` | % | 25 | 5..100 | W | GSD | - -- Recovery bandwidth overhead above input rate (see `SRTO_INPUTBW`), in percentage -of the input rate. It is effective only if `SRTO_MAXBW` is set to 0. - -- *Sender: user configurable, default: 25%.* - -- Recommendations: - - - *Overhead is intended to give you extra bandwidth for a case when -some packet has taken part of the bandwidth, but then was lost and has to be -retransmitted. Therefore the effective maximum bandwidth should be -appropriately higher than your stream's bitrate so that there's some room -for retransmission, but still limited so that the retransmitted packets -don't cause the bandwidth usage to skyrocket when larger groups of -packets were lost* - - - *Don't configure it too low and avoid 0 in case when you have -`SRTO_INPUTBW` option set to 0 (automatic) otherwise your stream will -choke and break quickly at any rising packet loss.* - -- ***To do: set-only. get should be supported.*** - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| -------------------- | ----- | ------- | ---------- | ------- | -------- | ------ | --- | ------ | -| `SRTO_PACKETFILTER` | 1.4.0 | pre | string | | "" | [512] | W | GSD | - -- Set up the packet filter. The string must match appropriate syntax for packet -filter setup. - -As there can only be one configuration for both parties, it is recommended that one party -defines the full configuration while the other only defines the matching packet filter type -(for example, one sets `fec,cols:10,rows:-5,layout:staircase` and the other -just `fec`). Both parties can also set this option to the same value. The packet filter function -will attempt to merge configuration definitions, but if the options specified are in -conflict, the connection will be rejected. - -For details, see [Packet Filtering & FEC](packet-filtering-and-fec.md). - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| -------------------- | ----- | ------- | ---------- | ------- | -------- | ------ | --- | ------ | -| `SRTO_PASSPHRASE` | 0.0.0 | pre | string | | "" |[10..79]| W | GSD | - -- Sets the passphrase for encryption. This enables encryption on this party (or -disables it, 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](http://en.wikipedia.org/wiki/PBKDF2) -(Password-Based Key Derivation Function 2). It is used on the receiver only if -the received data is encrypted. -- Note that since the bidirectional support, there's only one initial SEK to encrypt -the stream (new keys after refreshing will be updated independently) and there's no -distinction between "service party that defines the password" and "client party that -is required to set matching password" - both parties are equivalent and in order to -have a working encrypted connection, they have to simply set the same passphrase, -otherwise the connection is rejected by default (see also `SRTO_ENFORCEDENCRYPTION`). - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| -------------------- | ----- | ------- | ---------- | ------- | -------- | ------ | --- | ------ | -| `SRTO_PAYLOADSIZE` | 1.3.0 | pre | `int32_t` | bytes | * | 0.. | W | GSD | - -- Sets the maximum declared size of a single call to sending function in Live -mode. When set to 0, there's no limit for a single sending call. - -- For Live mode: Default value is 1316, can be increased up to 1456. Note that -with the `SRTO_PACKETFILTER` option additional header space is usually required, -which decreases the maximum possible value for `SRTO_PAYLOADSIZE`. - -- For File mode: Default value is 0 and it's recommended not to be changed. - - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| -------------------- | ----- | ------- | ---------- | ------- | -------- | ------ | --- | ------ | -| `SRTO_PBKEYLEN` | 0.0.0 | pre | `int32_t` | bytes | 0 | * | RW | GSD | - -- Sender encryption key length. - -- Possible values: - - 0 (`PBKEYLEN` not set) - - 16 (effective default) = AES-128 - - 24 = AES-192 - - 32 = AES-256 - -- The use is slightly different in 1.2.0 (HSv4) and since 1.3.0 (HSv5): - - - HSv4: This is set on the sender and enables encryption, if not 0. The receiver - shall not set it and will agree on the length as defined by the sender. - - - 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 - three intended methods of defining it, 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 on both parties. - - - **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. - - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| -------------------- | ----- | ------- | ---------- | ------- | -------- | ------ | --- | ------ | -| `SRTO_PEERIDLETIMEO` | 1.3.3 | pre | `int32_t` | ms | 5000 | 0.. | RW | GSD+ | - -- The maximum time in `[ms]` to wait until 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 | Dir | Entity | -| -------------------- | ----- | ------- | ---------- | ------- | -------- | ------ | --- | ------ | -| `SRTO_PEERLATENCY` | 1.3.0 | pre | `int32_t` | ms | 0 | 0.. | RW | GSD | - -- The latency value (as described in `SRTO_RCVLATENCY`) that is set by the sender -side as a minimum value for the receiver. - -- Note that when reading, the value will report the preset value on a non-connected -socket, and the effective value on a connected socket. - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| -------------------- | ----- | ------- | ---------- | ------- | -------- | ------ | --- | ------ | -| `SRTO_PEERVERSION` | 1.1.0 | | `int32_t` | * | | | R | GS | - -- SRT version used by the peer. The value 0 is returned if not connected, SRT -handshake not yet performed (HSv4 only), or if peer is not SRT. See `SRTO_VERSION` -for the version format. - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| -------------------- | ----- | ------- | ---------- | ------- | ---------- | ------ | --- | ------ | -| `SRTO_RCVBUF` | | pre | `int32_t` | bytes | 8192 bufs | * | RW | GSD+ | - - -- Receive Buffer Size, in bytes. Note, however, that the internal setting of this -value is in the number of buffers, each one of size equal to SRT payload size, -which is the value of `SRTO_MSS` decreased by UDP and SRT header sizes (28 and 16). -The value set here will be effectively aligned to the multiple of payload size. - -- Minimum value: 32 buffers (46592 with default value of `SRTO_MSS`). -- Maximum value: `SRTO_FC` number of buffers (receiver buffer must not be greater than FC size). - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| ----------------- | ----- | ------- | ---------- | ------- | ---------- | ------ | --- | ------ | -| `SRTO_RCVDATA` | | | `int32_t` | pkts | | | R | S | - -- Size of the available data in the receive buffer. - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| ----------------- | ----- | ------- | ---------- | ------- | ---------- | ------ | --- | ------ | -| `SRTO_RCVKMSTATE` | 1.2.0 | | `int32_t` | enum | | | R | S | - -- KM state on the agent side when it's a receiver. - -- Values defined in enum `SRT_KM_STATE` (see [SRT_KM_STATE](#srt-km-state)) - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| ----------------- | ----- | ------- | ---------- | ------- | ---------- | ------ | --- | ------ | -| `SRTO_RCVLATENCY` | 1.3.0 | pre | `int32_t` | msec | * | 0.. | RW | GSD | - -- Latency value in the receiving direction. This value is only significant when -`SRTO_TSBPDMODE` is set to true. - -- Latency refers to the time that elapses from the moment a packet is sent -to the moment when it's delivered to a receiver application. The SRT latency -setting should be a time buffer large enough to cover the time spent for -sending, unexpectedly extended RTT time, and the time needed to retransmit any -lost UDP packet. The effective latency value will be the maximum between the -`SRTO_RCVLATENCY` 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`. Note that the real latency value may be slightly different -than this setting due to the impossibility of perfectly measuring exactly the -same point in time at both parties simultaneously. What is important with -latency is that its actual value, once set with the connection, is kept constant -throughout the duration of a connection. - -- Default value: 120 in Live mode, 0 in File mode (see `SRTO_TRANSTYPE`). - -- Note that when reading, the value will report the preset value on a non-connected -socket, and the effective value on a connected socket. - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| ----------------- | ----- | ------- | ---------- | ------- | ---------- | ------ | --- | ------ | -| `SRTO_RCVSYN` | | post | `bool` | | true | | RW | GSI | - -- When true, sets blocking mode on reading function when it's not ready to -perform the operation. When false ("non-blocking mode"), the reading function -will in this case report error `SRT_EASYNCRCV` and return immediately. Details -depend on the tested entity: - -- On a connected socket or group this applies to a receiving function -(`srt_recv` and others) and a situation when there are no data available for -reading. The readiness state for this operation can be tested by checking the -`SRT_EPOLL_IN` flag on the aforementioned socket or group. - -- On a freshly created socket or group that is about to be connected to a peer -listener this applies to any `srt_connect` call (and derived), which in -"non-blocking mode" always return immediately. The connected state for that -socket or group can be tested by checking the `SRT_EPOLL_OUT` flag. NOTE -that a socket that failed to connect doesn't change the `SRTS_CONNECTING` -state and can be found out only by testing `SRT_EPOLL_ERR` flag. - -- On a listener socket this applies to `srt_accept` call. The readiness state -for this operation can be tested by checking the `SRT_EPOLL_IN` flag on -this listener socket. This flag is also derived from the listener socket -by the accepted socket or group, although the meaning of this flag is -effectively different. - -- Note that when this flag is set only on a group, it applies to a -specific receiving operation being done on that group (i.e. it is not -derived from the socket of which the group is a member). - - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| ----------------- | ----- | ------- | ---------- | ------- | ---------- | ------ | --- | ------ | -| `SRTO_RCVTIMEO` | | post | `int32_t` | ms | -1 | -1, 0..| RW | GSI | - -- Limit the time up to which the receiving operation will block (see -`SRTO_RCVSYN` for details), so when this time is exceeded, it will behave as -if in "non-blocking mode". The -1 value means no time limit. - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| ----------------- | ----- | ------- | ---------- | ------- | ---------- | ------ | --- | ------ | -| `SRTO_RENDEZVOUS` | | pre | `bool` | | false | | RW | S | - -- Use Rendezvous connection mode (both sides must set this and both must use the -procedure of `srt_bind` and then `srt_connect` (or `srt_rendezvous`) to one aother. - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| --------------------- | ----- | ------- | ------ | ------ | ------- | ------ | --- | ------ | -| `SRTO_RETRANSMITALGO` | 1.5.0 | pre | `int` | | 0 | [0, 1] | W | GSD | - -- Retransmission algorithm to use (SENDER option): - - 0 - Default (retranmsit on every loss report). - - 1 - Reduced retransmissions (not more often than once per RTT) - reduced bandwidth consumption. - -- This option is effective only on the sending side. It influences the -decision as to whether particular reported lost packets should be retransmitted -at a certain time or not. - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| ----------------- | ----- | ------- | ---------- | ------- | ---------- | ------ | --- | ------ | -| `SRTO_REUSEADDR` | | pre | `bool` | | true | | RW | GSD | - -- 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 | Dir | Entity | -| ----------------- | ----- | ------- | ---------- | ------- | ---------- | ------ | --- | ------ | -| `SRTO_SENDER` | 1.0.4 | pre | `bool` | | false | | W | S | - -- Set sender side. The side that sets this flag is expected to be a sender. -This flag is only required when communicating with a receiver that uses SRT -version less than 1.3.0 (and hence *HSv4* handshake), in which case if not -set properly, the TSBPD mode (see `SRTO_TSBPDMODE`) or encryption will not work. -Setting `SRTO_MINVERSION` to 1.3.0 is therefore recommended. - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| ----------------- | ----- | ------- | ---------- | ------- | --------- | ------ | --- | ------ | -| `SRTO_CONGESTION` | 1.3.0 | pre | `string` | | "live" | * | W | S | - -- The type of congestion controller used for the transmission for that socket. - -- Its type must be exactly the same on both connecting parties, otherwise the -connection is rejected - **however** you may also change the value of this -option for the accepted socket in the listener callback (see `srt_listen_callback`) -if an appropriate instruction was given in the Stream ID. - -- Currently supported congestion controllers are designated as "live" and "file" - -- Note that it is not recommended to change this option manually, but you should -rather change the whole set of options through `SRTO_TRANSTYPE` option. - - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| ----------------- | ----- | ------- | ---------- | ------- | --------- | ------ | --- | ------ | -| `SRTO_SNDBUF` | | pre | `int32_t` | bytes |8192 bufs | * | RW | GSD+ | - -- Sender Buffer Size. See `SRTO_RCVBUF` for more information. - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| ----------------- | ----- | ------- | ---------- | ------- | --------- | ------ | --- | ------ | -| `SRTO_SNDDATA` | | | `int32_t` | pkts | | | R | S | - -- Size of the unacknowledged data in send buffer. - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| -------------------- | ----- | ------- | ---------- | ------- | --------- | ------ | --- | ------ | -| `SRTO_SNDDROPDELAY` | 1.3.2 | pre | `int32_t` | ms | * | -1.. | W | GSD+ | - -- Sets an extra delay before TLPKTDROP is triggered on the data sender. -This delay is added to the default drop delay time interval value. Keep in mind -that the longer the delay, the more probable it becomes that packets would be -retransmitted uselessly because they will be dropped by the receiver anyway. - -- TLPKTDROP discards packets reported as lost if it is already too late -to send them (the receiver would discard them even if received). The delay -before TLPKTDROP mechanism is triggered consists of the SRT latency -(`SRTO_PEERLATENCY`), plus `SRTO_SNDDROPDELAY`, plus `2 * interval between -sending ACKs` (the default `interval between sending ACKs` is 10 milliseconds). -The minimum delay is `1000 + 2 * interval between sending ACKs` milliseconds. - -- **Special value -1**: Do not drop packets on the sender at all (retransmit them always when requested). - -- Default: 0 in Live mode, -1 in File mode. - - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| -------------------- | ----- | ------- | ---------- | ------- | --------- | ------ | --- | ------ | -| `SRTO_SNDKMSTATE` | 1.2.0 | post | `int32_t` | enum | | | R | S | - -- Peer KM state on receiver side for `SRTO_KMSTATE` - -- Values defined in enum `SRT_KM_STATE` (see [SRT_KM_STATE](#srt-km-state)) - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| -------------------- | ----- | ------- | ---------- | ------- | --------- | ------ | --- | ------ | -| `SRTO_SNDSYN` | | post | `bool` | | true | | RW | GSI | - -- When true, sets blocking mode on writing function when it's not ready to -perform the operation. When false ("non-blocking mode"), the writing function -will in this case report error `SRT_EASYNCSND` and return immediately. - -- On a connected socket or group this applies to a sending function -(`srt_send` and others) and a situation when there's no free space in -the sender buffer, caused by inability to send all the scheduled data over -the network. Readiness for this operation can be tested by checking the -`SRT_EPOLL_OUT` flag. - -- On a freshly created socket or group it will have no effect until the socket -enters a connected state. - -- On a listener socket it will be derived by the accepted socket or group, -but will have no effect on the listener socket itself. - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| -------------------- | ----- | ------- | ---------- | ------- | --------- | ------ | --- | ------ | -| `SRTO_SNDTIMEO` | | post | `int32_t` | ms | -1 | -1.. | RW | GSI | - -- limit the time up to which the sending operation will block (see -`SRTO_SNDSYN` for details), so when this time is exceeded, it will behave as -if in "non-blocking mode". The -1 value means no time limit. - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| -------------------- | ----- | ------- | ---------- | ------- | --------- | ------ | --- | ------ | -| `SRTO_STATE` | | | `int32_t` | enum | | | R | S | - -- Returns the current socket state, same as `srt_getsockstate`. - - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| -------------------- | ----- | ------- | ---------- | ------- | --------- | ------ | --- | ------ | -| `SRTO_STREAMID` | 1.3.0 | pre | `string` | | "" | [512] | RW | GSD | - -- A string that can be set on the socket prior to connecting. The listener side -will be able to retrieve this stream ID from the socket that is returned from -`srt_accept` (for a connected socket with that stream ID). You usually use SET -on the socket used for `srt_connect`, and GET on the socket retrieved from -`srt_accept`. This string can be used completely free-form. However, it's highly -recommended to follow the [SRT Access Control guidlines](AccessControl.md). - -- As this uses internally the `std::string` type, there are additional functions -for it in the legacy/C++ API (udt.h): `srt::setstreamid` and -`srt::getstreamid`. - -- This option is not useful for a Rendezvous connection, since once side would -override the value from the other side resulting in an arbitrary winner. Also -in this connection both peers are known from upside to one another and both -have equivalent roles in the connection. - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| -------------------------- | ----- | ------- | ---------- | ------- | --------- | ------ | --- | ------ | -| `SRTO_ENFORCEDENCRYPTION` | 1.3.2 | pre | `bool` | | true | | W | GSD | - -- This option enforces that both connection parties have the same passphrase -set, or both do not set the passphrase, otherwise the connection is rejected. - -- When this option is set to FALSE **on both connection parties**, the -connection is allowed even if the passphrase differs on both parties, -or it was set only on one party. Note that the party that has set a passphrase -is still allowed to send data over the network. However, the receiver will not -be able to decrypt that data and will not deliver it to the application. The -party that has set no passphrase can send (unencrypted) data that will be -successfully received by its peer. - -- This option can be used in some specific situations when the user knows -both parties of the connection, so there's no possible situation of a rogue -sender and can be useful in situations where it is important to know whether a -connection is possible. The inability to decrypt an incoming transmission can -be then reported as a different kind of problem. - -**IMPORTANT**: There is unusual and unobvious behavior when this flag is TRUE -on the caller and FALSE on the listener, and the passphrase was mismatched. On -the listener side the connection will be established and broken right after, -resulting in a short-lived "spurious" connection report on the listener socket. -This way, a socket will be available for retrieval from an `srt_accept` call -for a very short time, after which it will be removed from the listener backlog -just as if no connection attempt was made at all. If the application is fast -enough to react on an incoming connection, it will retrieve it, only to learn -that it is already broken. This also makes possible a scenario where -`SRT_EPOLL_IN` is reported on a listener socket, but then an `srt_accept` call -reports an `SRT_EASYNCRCV` error. How fast the connection gets broken depends -on the network parameters - in particular, whether the `UMSG_SHUTDOWN` message -sent by the caller is delivered (which takes one RTT in this case) or missed -during the interval from its creation up to the connection timeout (default = 5 -seconds). It is therefore strongly recommended that you only set this flag to -FALSE on the listener when you are able to ensure that it is also set to FALSE -on the caller side. - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| ----------------- | ----- | ------- | ---------- | ------- | --------- | ------ | --- | ------ | -| `SRTO_TLPKTDROP` | 1.0.6 | pre | `bool` | | * | | RW | GSD | - -- Too-late Packet Drop. When enabled on receiver, it skips missing packets that -have not been delivered in time and delivers the subsequent packets to the -application when their time-to-play has come. It also sends a fake ACK to the -sender. When enabled on sender and enabled on the receiving peer, sender drops -the older packets that have no chance to be delivered in time. It is automatically -enabled in sender if receiver supports it. - -- Default: true in Live mode, false in File mode (see `SRTO_TRANSTYPE`) - ---- - -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| ----------------- | ----- | ------- | ---------- | ------- | --------- | ------ | --- | ------ | -| `SRTO_TRANSTYPE` | 1.3.0 | pre | `int32_t` | enum |`SRTT_LIVE`| * | W | S | - -- Sets the transmission type for the socket, in particular, setting this option -sets multiple other parameters to their default values as required for a -particular transmission type. - -- Values defined by enum `SRT_TRANSTYPE` (see above for possible values) - ---- +Transmission types +------------------ -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| ----------------- | ----- | ------- | ---------- | ------- | --------- | ------ | --- | ------ | -| `SRTO_TSBPDMODE` | 0.0.0 | pre | `bool` | | * | | W | S | +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. -- When true, use Timestamp-based Packet Delivery mode. In this mode the -packet's time is assigned at the sending time (or allowed to be predefined), -transmitted in the packet's header and restored on the receiver side so that -time distances between consecutive packets are preserved when delivering to -the application. +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. -- Default: true in Live mode, false in File mode (see `SRTO_TRANSTYPE`). +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) -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| ----------------- | ----- | ------- | ---------- | ------- | --------- | ------ | --- | ------ | -| `SRTO_UDP_RCVBUF` | | pre | `int32_t` | bytes | 8192 bufs | * | RW | GSD+ | +This makes possible a total of three data transmission methods: -- UDP Socket Receive Buffer Size. Configured in bytes, maintained in packets -based on MSS value. Receive buffer must not be greater than FC size. +* [Live](#transmission-method-live) +* [Buffer](#transmission-method-buffer) +* [Message](#transmission-method-message) ---- +### Terminology -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| ----------------- | ----- | ------- | ---------- | ------- | --------- | ------ | --- | ------ | -| `SRTO_UDP_SNDBUF` | | pre | `int32_t` | bytes | 65536 | * | RW | GSD+ | +The following terms are used in the description of transmission types: -- UDP Socket Send Buffer Size. Configured in bytes, maintained in packets based -on `SRTO_MSS` value. +**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. -| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | -| ----------------- | ----- | ------- | ---------- | ------- | --------- | ------ | --- | ------ | -| `SRTO_VERSION` | 1.1.0 | | `int32_t` | | | | R | S | +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-funtions.md](API-funtions.md) +for details. -- Local SRT version. This is the highest local version supported if not - connected, or the highest version supported by the peer if connected. +The following types of operations are involved: -- The version format in hex is 0x00XXYYZZ for x.y.z in human readable form. -For example, version 1.4.2 is encoded as `0x010402`. +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. -Transmission types ------------------- +2. Writing data: `srt_send`, `srt_sendmsg`, `srt_sendmsg2`, `srt_sendfile`. -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. +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. -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. +3. Accepting an incoming connection: `srt_accept` -There are two options to set a particular type: +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. -* `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) +4. Connecting: `srt_connect` and its derivatives -This makes possible a total of three data transmission methods: +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. -* 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. +**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 parameters: +Setting `SRTO_TRANSTYPE` to `SRTT_LIVE` sets the following [parameters](APISocketOptions.md): * `SRTO_TSBPDMODE` = true * `SRTO_RCVLATENCY` = 120 @@ -1631,60 +552,63 @@ 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. +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`). +lost packets (if TSBPD mode is off - see [`SRTO_TSBPDMODE`](APISocketOptions.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 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 + +* `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 +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 + +* `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 lossreport, when the lost + +* `SRTO_NAKREPORT`: Turns on repeated sending of loss reports, 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 +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_PAYLOADSIZE`: Default value is for MPEG TS; if you are going + +* `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. +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. +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 parameters: +Setting `SRTO_TRANSTYPE` to `SRTT_FILE` sets the following [parameters](APISocketOptions.md): * `SRTO_TSBPDMODE` = false * `SRTO_RCVLATENCY` = 0 @@ -1697,7 +621,7 @@ Setting `SRTO_TRANSTYPE` to `SRTT_FILE` sets the following parameters: 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 +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 @@ -1707,17 +631,16 @@ 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. +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 +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 +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 @@ -1726,67 +649,71 @@ 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. +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 `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 +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 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. +**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 [APISocketOptions.md](APISocketOptions.md) + + +[Return to top](#srt-api) + diff --git a/docs/APISocketOptions.md b/docs/APISocketOptions.md new file mode 100644 index 000000000..05a2608de --- /dev/null +++ b/docs/APISocketOptions.md @@ -0,0 +1,1510 @@ +SRT Socket Options +================== + +There is a general method of setting options on a socket in the SRT C API, similar +to the system `setsockopt/getsockopt` functions. + +**NOTE**: This document replaces the socket option description originally +in [api.md](https://github.com/Haivision/srt/blob/master/docs/API.md) + +1. [Types used in socket options](#types-used-in-socket-options) + - [List of options](#list-of-options) + - [Option Descriptions](#option-descriptions) +2. [Enumeration types used in options](#enumeration-types-used-in-options) + - [`SRT_TRANSTYPE`](#1-srt_transtype) + - [`SRT_KM_STATE`](#2-srt_km_state) +3. [Getting and setting options](#getting-and-setting-options) + + +Types used in socket options +---------------------------- + +Possible types of socket options are: + +* `int32_t` - This type can usually be treated as an `int` equivalent since it +does not change size on 64-bit systems. For clarity, options use this fixed size +integer. In some cases the value is expressed using an enumeration type (see below). + +* `int64_t` - Some options need the parameter specified as 64-bit integer + +* `bool` - Requires the use of a boolean type (`` for C, or built-in +for C++). When *setting* an option, passing the value through an `int` type is +also properly recognized. When *getting* an option, however, you should use the +`bool` type, although you can risk passing a variable of `int` type initialized +with 0 and then checking if the resulting value is equal to 0 (just don't compare +the result with 1). + +* `string` - When *setting* an option, pass the character array pointer as value +and the string length as length. When *getting*, pass an array of sufficient size +(as specified in the size variable). Every option with this type that can be +read should specify the maximum length of that array. + +* `linger` - Linger structure. Used exclusively with `SRTO_LINGER`. + + +Enumeration types used in options +--------------------------------- + +### 1. `SRT_TRANSTYPE` + +Used by `SRTO_TRANSTYPE` option: + +* `SRTT_LIVE`: Live mode. +* `SRTT_FILE`: File mode. + +See [Transmission types](#transmission-types) below for details. + + +### 2. `SRT_KM_STATE` + +The defined encryption state as performed by the Key Material Exchange, used +by `SRTO_RCVKMSTATE`, `SRTO_SNDKMSTATE` and `SRTO_KMSTATE` options: + +- `SRT_KM_S_UNSECURED`: no encryption/decryption. If this state is only on +the receiver, received encrypted packets will be dropped. + +- `SRT_KM_S_SECURING`: pending security (HSv4 only). This is a temporary state +used only if the connection uses HSv4 and the Key Material Exchange is +not finished yet. On HSv5 this is not possible because the Key Material +Exchange for the initial key is done in the handshake. + +- `SRT_KM_S_SECURED`: KM exchange was successful and the data will be sent +encrypted and will be decrypted by the receiver. This state is only possible on +both sides in both directions simultaneously. + +- `SRT_KM_S_NOSECRET`: If this state is in the sending direction (`SRTO_SNDKMSTATE`), +then it means that the sending party has set a passphrase, but the peer did not. +In this case the sending party can receive unencrypted packets from the peer, but +packets it sends to the peer will be encrypted and the peer will not be able to +decrypt them. This state is only possible in HSv5. + +- `SRT_KM_S_BADSECRET`: The password is wrong (set differently on each party); +encrypted payloads won't be decrypted in either direction. + +Note that with the default value of `SRTO_ENFORCEDENCRYPTION` option (true), +the state is equal on both sides in both directions, and it can be only +`SRT_KM_S_UNSECURED` or `SRT_KM_S_SECURED` (in other cases the connection +is rejected). Otherwise it may happen that either both sides have different +passwords and the state is `SRT_KM_S_BADSECRET` in both directions, or only +one party has set a password, in which case the KM state is as follows: + +| | `SRTO_RCVKMSTATE` | `SRTO_SNDKMSTATE` | +|--------------------------|----------------------|----------------------| +| Party with no password: | `SRT_KM_S_NOSECRET` | `SRT_KM_S_UNSECURED` | +| Party with password: | `SRT_KM_S_UNSECURED` | `SRT_KM_S_NOSECRET` | + + +Getting and setting options +--------------------------- + +Legacy version: + + int srt_getsockopt(SRTSOCKET socket, int level, SRT_SOCKOPT optName, void* optval, int& optlen); + int srt_setsockopt(SRTSOCKET socket, int level, SRT_SOCKOPT optName, const void* optval, int optlen); + +New version: + + int srt_getsockflag(SRTSOCKET socket, SRT_SOCKOPT optName, void* optval, int& optlen); + int srt_setsockflag(SRTSOCKET socket, SRT_SOCKOPT optName, const void* optval, int optlen); + +In the legacy version, there's an additional unused `level` parameter. It was +there in the original UDT API just to mimic the system `setsockopt` function, +but it's ignored. + +Some options require a value of type `bool` while others require type `integer`, +which is not the same -- they differ in size, and mistaking them may end up +causing a crash. This must be kept in mind especially in any C wrapper. For +convenience, the *setting* option function may accept both `int32_t` and `bool` +types, but this is not so in the case of *getting* an option value. + +**UDT project legacy note**: Almost all options from the UDT library are +derived (there are a few deleted, including some deprecated already in UDT). +Many new SRT options have been added. All options are available exclusively +with the `SRTO_` prefix. Old names are provided as alias names in the `udt.h` +legacy/C++ API file. Note the translation rules: +* `UDT_` prefix from UDT options was changed to the prefix `SRTO_` +* `UDP_` prefix from UDT options was changed to the prefix `SRTO_UDP_` +* `SRT_` prefix in older SRT versions was changed to `SRTO_` + +The [table below](#list-of-options) provides a complete list of SRT options and +their characteristics according to the following legend: + +1. **Since**: Defines the SRT version when this option was first introduced. If this field +is empty, it's an option derived from UDT. "Version 0.0.0" is the oldest version +of SRT ever created and put into use. + +2. **Binding**: Defines limitation on setting the option. The field is empty if the option +is not settable (see **Dir** column): + + - `pre`: A connecting socket (both as caller and rendezvous) must be set +prior to calling `srt_connect()` or `srt_bind()` and never changed thereafter. +A listener socket should be set to "listening" 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 (as well as on an accepted socket). Setting this flag on a listening +socket is effective only on that socket itself. Note though that there are some +post-bound options that have important meaning when set prior to connecting. + +3. **Type**: The data type of the option (see above). + +4. **Units**: Roughly specified unit, if the value defines things like length or time. +It can also define more precisely what kind of specialization can be used +when the type is integer: + + - `enum`: the possible values are defined in an enumeration type + - `flags`: the integer value is a collection of bit flags + +5. **Default**: The exact default value, if it can be easily specified. A more complicated +default state of a particular option will be explained in the [description](#option-descriptions) +(when marked by asterisk). For non-settable options this field is empty. + +6. **Range**: If a value of an integer type has a limited range, or only a certain value +allowed, it will be specified here (otherwise empty). A range value can be +specified as: + + - `X-... `: specifies only a minimum value + - `X-Y,Z `: values between X and Y are allowed, and additionally Z + + - If the value is of `string` type, this field will contain its maximum size +in square brackets. + + - If the range contains additionally an asterisk, it means that more elaborate +restrictions on the value apply, as explained in the [description](#option-descriptions). + +7. **Dir**: Option direction: W if can be set, R if can be retrieved, RW if both. + +6. **Entity**: This describes whether the option can be set on the socket or the group. +The G and S options may appear together, in which case both possibilities apply. +The D and I options, mutually exclusive, appear always with G. +The + marker can only coexist with GS. Possible specifications are: + + - S: This option can be set on a single socket (exclusively, if not GS) + + - G: This option can be set on a group (exclusively, if not GS) + + - D: If set on a group, it will be derived by the member socket + + - I: If set on a group, it will be taken and managed exclusively by the group + + - +: This option is also allowed to be set individually on a group member + socket through a configuration object in `SRT_SOCKGROUPCONFIG` prepared by + `srt_create_config`. Note that this setting may override the setting derived + from the group. + +## List of options + +The following table lists SRT socket options in alphabetical order. Option details are given further below. + +| Option Name | Since | Binding | Type | Units | Default | Range | Dir |Entity | +| :----------------------------------------------------- | :---: | :-----: | :-------: | :-----: | :--------: | :------: |:---:|:-----:| +| [`SRTO_BINDTODEVICE`](#SRTO_BINDTODEVICE) | 1.5.0 | pre | `string` | | | | RW | GSD+ | +| [`SRTO_CONGESTION`](#SRTO_CONGESTION) | 1.3.0 | pre | `string` | | "live" | * | W | S | +| [`SRTO_CONNTIMEO`](#SRTO_CONNTIMEO) | 1.1.2 | pre | `int32_t` | msec | 3000 | 0.. | W | GSD+ | +| [`SRTO_DRIFTTRACER`](#SRTO_DRIFTTRACER) | 1.5.0 | post | `bool` | | true | | RW | GSD | +| [`SRTO_ENFORCEDENCRYPTION`](#SRTO_ENFORCEDENCRYPTION) | 1.3.2 | pre | `bool` | | true | | W | GSD | +| [`SRTO_EVENT`](#SRTO_EVENT) | | | `int32_t` | flags | | | R | S | +| [`SRTO_FC`](#SRTO_FC) | | pre | `int32_t` | pkts | 25600 | 32.. | RW | GSD | +| [`SRTO_GROUPCONNECT`](#SRTO_GROUPCONNECT) | 1.5.0 | pre | `int32_t` | | 0 | 0...1 | W | S | +| [`SRTO_GROUPSTABTIMEO`](#SRTO_GROUPSTABTIMEO) | | pre | `int32_t` | ms | 40 | 10-... | W | GSD+ | +| [`SRTO_GROUPTYPE`](#SRTO_GROUPTYPE) | | pre | `int32_t` | enum | | | R | S | +| [`SRTO_INPUTBW`](#SRTO_INPUTBW) | 1.0.5 | post | `int64_t` | B/s | 0 | 0.. | W | GSD | +| [`SRTO_IPTOS`](#SRTO_IPTOS) | 1.0.5 | pre | `int32_t` | | (system) | 0..255 | RW | GSD | +| [`SRTO_IPTTL`](#SRTO_IPTTL) | 1.0.5 | pre | `int32_t` | hops | (system) | 1..255 | RW | GSD | +| [`SRTO_IPV6ONLY`](#SRTO_IPV6ONLY) | 1.4.0 | pre | `int32_t` | | (system) | -1..1 | RW | GSD | +| [`SRTO_ISN`](#SRTO_ISN) | 1.3.0 | | `int32_t` | | | | R | S | +| [`SRTO_KMPREANNOUNCE`](#SRTO_KMPREANNOUNCE) | 1.3.2 | pre | `int32_t` | pkts | 0x1000 | 0.. * | RW | GSD | +| [`SRTO_KMREFRESHRATE`](#SRTO_KMREFRESHRATE) | 1.3.2 | pre | `int32_t` | pkts | 0x1000000 | 0.. | RW | GSD | +| [`SRTO_KMSTATE`](#SRTO_KMSTATE) | 1.0.2 | | `int32_t` | enum | | | R | S | +| [`SRTO_LATENCY`](#SRTO_LATENCY) | 1.0.2 | pre | `int32_t` | ms | 120 * | 0.. | RW | GSD | +| [`SRTO_LINGER`](#SRTO_LINGER) | | pre | `linger` | s | on, 180 | 0.. | RW | GSD | +| [`SRTO_LOSSMAXTTL`](#SRTO_LOSSMAXTTL) | 1.2.0 | pre | `int32_t` | packets | 0 | 0.. | RW | GSD+ | +| [`SRTO_MAXBW`](#SRTO_MAXBW) | 1.0.5 | pre | `int64_t` | B/s | -1 | -1.. | RW | GSD | +| [`SRTO_MESSAGEAPI`](#SRTO_MESSAGEAPI) | 1.3.0 | pre | `bool` | | true | | W | GSD | +| [`SRTO_MINVERSION`](#SRTO_MINVERSION) | 1.3.0 | pre | `int32_t` | version | 0 | * | W | GSD | +| [`SRTO_MSS`](#SRTO_MSS) | | pre | `int32_t` | bytes | 1500 | 76.. | RW | GSD | +| [`SRTO_NAKREPORT`](#SRTO_NAKREPORT) | 1.1.0 | pre | `bool` | | * | | RW | GSD+ | +| [`SRTO_OHEADBW`](#SRTO_OHEADBW) | 1.0.5 | post | `int32_t` | % | 25 | 5..100 | W | GSD | +| [`SRTO_PACKETFILTER`](#SRTO_PACKETFILTER) | 1.4.0 | pre | `string` | | "" | [512] | W | GSD | +| [`SRTO_PASSPHRASE`](#SRTO_PASSPHRASE) | 0.0.0 | pre | `string` | | "" | [10..79] | W | GSD | +| [`SRTO_PAYLOADSIZE`](#SRTO_PAYLOADSIZE) | 1.3.0 | pre | `int32_t` | bytes | * | * | W | GSD | +| [`SRTO_PBKEYLEN`](#SRTO_PBKEYLEN) | 0.0.0 | pre | `int32_t` | bytes | 0 | * | RW | GSD | +| [`SRTO_PEERIDLETIMEO`](#SRTO_PEERIDLETIMEO) | 1.3.3 | pre | `int32_t` | ms | 5000 | 0.. | RW | GSD+ | +| [`SRTO_PEERLATENCY`](#SRTO_PEERLATENCY) | 1.3.0 | pre | `int32_t` | ms | 0 | 0.. | RW | GSD | +| [`SRTO_PEERVERSION`](#SRTO_PEERVERSION) | 1.1.0 | | `int32_t` | * | | | R | GS | +| [`SRTO_RCVBUF`](#SRTO_RCVBUF) | | pre | `int32_t` | bytes | 8192 bufs | * | RW | GSD+ | +| [`SRTO_RCVDATA`](#SRTO_RCVDATA) | | | `int32_t` | pkts | | | R | S | +| [`SRTO_RCVKMSTATE`](#SRTO_RCVKMSTATE) | 1.2.0 | | `int32_t` | enum | | | R | S | +| [`SRTO_RCVLATENCY`](#SRTO_RCVLATENCY) | 1.3.0 | pre | `int32_t` | msec | * | 0.. | RW | GSD | +| [`SRTO_RCVSYN`](#SRTO_RCVSYN) | | post | `bool` | | true | | RW | GSI | +| [`SRTO_RCVTIMEO`](#SRTO_RCVTIMEO) | | post | `int32_t` | ms | -1 | -1, 0.. | RW | GSI | +| [`SRTO_RENDEZVOUS`](#SRTO_RENDEZVOUS) | | pre | `bool` | | false | | RW | S | +| [`SRTO_RETRANSMITALGO`](#SRTO_RETRANSMITALGO) | 1.5.0 | pre | `int` | | 0 | [0, 1] | W | GSD | +| [`SRTO_REUSEADDR`](#SRTO_REUSEADDR) | | pre | `bool` | | true | | RW | GSD | +| [`SRTO_SENDER`](#SRTO_SENDER) | 1.0.4 | pre | `bool` | | false | | W | S | +| [`SRTO_SNDBUF`](#SRTO_SNDBUF) | | pre | `int32_t` | bytes | 8192 bufs | * | RW | GSD+ | +| [`SRTO_SNDDATA`](#SRTO_SNDDATA) | | | `int32_t` | pkts | | | R | S | +| [`SRTO_SNDDROPDELAY`](#SRTO_SNDDROPDELAY) | 1.3.2 | pre | `int32_t` | ms | * | -1.. | W | GSD+ | +| [`SRTO_SNDKMSTATE`](#SRTO_SNDKMSTATE) | 1.2.0 | post | `int32_t` | enum | | | R | S | +| [`SRTO_SNDSYN`](#SRTO_SNDSYN) | | post | `bool` | | true | | RW | GSI | +| [`SRTO_SNDTIMEO`](#SRTO_SNDTIMEO) | | post | `int32_t` | ms | -1 | -1.. | RW | GSI | +| [`SRTO_STATE`](#SRTO_STATE) | | | `int32_t` | enum | | | R | S | +| [`SRTO_STREAMID`](#SRTO_STREAMID) | 1.3.0 | pre | `string` | | "" | [512] | RW | GSD | +| [`SRTO_TLPKTDROP`](#SRTO_TLPKTDROP) | 1.0.6 | pre | `bool` | | * | | RW | GSD | +| [`SRTO_TRANSTYPE`](#SRTO_TRANSTYPE) | 1.3.0 | pre | `int32_t` | enum |`SRTT_LIVE` | * | W | S | +| [`SRTO_TSBPDMODE`](#SRTO_TSBPDMODE) | 0.0.0 | pre | `bool` | | * | | W | S | +| [`SRTO_UDP_RCVBUF`](#SRTO_UDP_RCVBUF) | | pre | `int32_t` | bytes | 8192 bufs | * | RW | GSD+ | +| [`SRTO_UDP_SNDBUF`](#SRTO_UDP_SNDBUF) | | pre | `int32_t` | bytes | 65536 | * | RW | GSD+ | +| [`SRTO_VERSION`](#SRTO_VERSION) | 1.1.0 | | `int32_t` | | | | R | S | + + + +### Option Descriptions + +#### SRTO_BINDTODEVICE + + +| OptName | Since | Binding | Type | Units | Default | Range | Dir |Entity| +| --------------------- | ----- | ------- | -------- | ------ | -------- | ------ |-----|------| +| `SRTO_BINDTODEVICE` | 1.5.0 | pre | `string` | | | | RW | GSD+ | + +- Refers to the `SO_BINDTODEVICE` system socket option for `SOL_SOCKET` level. +This effectively limits the packets received by this socket to only those +that are targeted to that device. The device is specified by name passed as +string. The setting becomes effective after binding the socket (including +default-binding when connecting). + +- NOTE: This option is only available on Linux and available there by default. +On all other platforms setting this option will always fail. + +- NOTE: With the default system configuration, this option is only available +for a process that runs as root. Otherwise the function that applies the setting +(`srt_bind`, `srt_connect` etc.) will fail. + + +[Return to list](#list-of-options) + + +#### SRTO_CONGESTION + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| ----------------- | ----- | ------- | ---------- | ------- | --------- | ------ | --- | ------ | +| `SRTO_CONGESTION` | 1.3.0 | pre | `string` | | "live" | * | W | S | + +- The type of congestion controller used for the transmission for that socket. + +- Its type must be exactly the same on both connecting parties, otherwise the +connection is rejected - **however** you may also change the value of this +option for the accepted socket in the listener callback (see `srt_listen_callback`) +if an appropriate instruction was given in the Stream ID. + +- Currently supported congestion controllers are designated as "live" and "file" + +- Note that it is not recommended to change this option manually, but you should +rather change the whole set of options using the [`SRTO_TRANSTYPE`](#SRTO_TRANSTYPE) option. + + +#### SRTO_CONNTIMEO + + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| ------------------ | ----- | ------- | --------- | ------ | -------- | ------ | --- | ------ | +| `SRTO_CONNTIMEO` | 1.1.2 | pre | `int32_t` | msec | 3000 | 0.. | W | GSD+ | + +- Connect timeout. This option applies to the caller and rendezvous connection +modes. For the rendezvous mode (see `SRTO_RENDEZVOUS`) the effective connection timeout +will be 10 times the value set with `SRTO_CONNTIMEO`. + + +[Return to list](#list-of-options) + + +#### SRTO_DRIFTTRACER + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| ----------------- | ----- | ------- | --------- | ------ | -------- | ------ | --- | ------ | +| `SRTO_DRIFTTRACER`| 1.5.0 | post | `bool` | | true | | RW | GSD | + +- Enables or disables time drift tracer (receiver). + + +[Return to list](#list-of-options) + + + +#### SRTO_ENFORCEDENCRYPTION + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| -------------------------- | ----- | ------- | ---------- | ------- | --------- | ------ | --- | ------ | +| `SRTO_ENFORCEDENCRYPTION` | 1.3.2 | pre | `bool` | | true | | W | GSD | + +- This option enforces that both connection parties have the same passphrase +set, or both do not set the passphrase, otherwise the connection is rejected. + +- When this option is set to FALSE **on both connection parties**, the +connection is allowed even if the passphrase differs on both parties, +or it was set only on one party. Note that the party that has set a passphrase +is still allowed to send data over the network. However, the receiver will not +be able to decrypt that data and will not deliver it to the application. The +party that has set no passphrase can send (unencrypted) data that will be +successfully received by its peer. + +- This option can be used in some specific situations when the user knows +both parties of the connection, so there's no possible situation of a rogue +sender and can be useful in situations where it is important to know whether a +connection is possible. The inability to decrypt an incoming transmission can +be then reported as a different kind of problem. + +**IMPORTANT**: There is unusual and unobvious behavior when this flag is TRUE +on the caller and FALSE on the listener, and the passphrase was mismatched. On +the listener side the connection will be established and broken right after, +resulting in a short-lived "spurious" connection report on the listener socket. +This way, a socket will be available for retrieval from an `srt_accept` call +for a very short time, after which it will be removed from the listener backlog +just as if no connection attempt was made at all. If the application is fast +enough to react on an incoming connection, it will retrieve it, only to learn +that it is already broken. This also makes possible a scenario where +`SRT_EPOLL_IN` is reported on a listener socket, but then an `srt_accept` call +reports an `SRT_EASYNCRCV` error. How fast the connection gets broken depends +on the network parameters -- in particular, whether the `UMSG_SHUTDOWN` message +sent by the caller is delivered (which takes one RTT in this case) or missed +during the interval from its creation up to the connection timeout (default = 5 +seconds). It is therefore strongly recommended that you only set this flag to +FALSE on the listener when you are able to ensure that it is also set to FALSE +on the caller side. + + +[Return to list](#list-of-options) + + + +#### SRTO_EVENT + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| ----------------- | ----- | ------- | --------- | ------ | -------- | ------ | --- | ------ | +| `SRTO_EVENT` | | | `int32_t` | flags | | | R | S | + +- Returns bit flags set according to the current active events on the socket. +- Possible values are those defined in `SRT_EPOLL_OPT` enum (a combination of +`SRT_EPOLL_IN`, `SRT_EPOLL_OUT` and `SRT_EPOLL_ERR`). + + +[Return to list](#list-of-options) + + + +#### SRTO_FC + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| ----------------- | ----- | ------- | --------- | ------ | -------- | ------ | --- | ------ | +| `SRTO_FC` | | pre | `int32_t` | pkts | 25600 | 32.. | RW | GSD | + +- Flight Flag Size (maximum number of bytes that can be sent without +being acknowledged) + + +[Return to list](#list-of-options) + + + +#### SRTO_GROUPCONNECT + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| -------------------- | ----- | ------- | --------- | ------ | -------- | ------ | --- | ------ | +| `SRTO_GROUPCONNECT` | 1.5.0 | pre | `int32_t` | | 0 | 0...1 | W | S | + +- When this flag is set to 1 on a listener socket, it allows this socket to +accept group connections. When set to the default 0, group connections will be +rejected. Keep in mind that if the `SRTO_GROUPCONNECT` flag is set to 1 (i.e. +group connections are allowed) `srt_accept` may return a socket **or** a group +ID. A call to `srt_accept` on a listener socket that has group connections +allowed must take this into consideration. It's up to the caller of this +function to make this distinction and to take appropriate action depending on +the type of entity returned. + +- When this flag is set to 1 on an accepted socket that is passed to the +listener callback handler, it means that this socket is created for a group +connection and it will become a member of a group. Note that in this case +only the first connection within the group will result in reporting from +`srt_accept` (further connections are handled in the background), and this +function will return the group, not this socket ID. + + +[Return to list](#list-of-options) + + + +#### SRTO_GROUPSTABTIMEO + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| --------------------- | ----- | ------- | ---------- | ------ | -------- | ------ | --- | ------ | +| `SRTO_GROUPSTABTIMEO` | | pre | `int32_t` | ms | 40 | 10-... | W | GSD+ | + +- This setting is used for groups of type `SRT_GTYPE_BACKUP`. It defines the stability +timeout, which is the maximum interval between two consecutive packets retrieved from +the peer on the currently active link. These two packets can be of any type, +but this setting usually refers to control packets while the agent is a sender. +Idle links exchange only keepalive messages once per second, so they do not +count. Note that this option is meaningless on sockets that are not members of +the Backup-type group. + +- This value should be set with a thoroughly selected balance and correspond to +the maximum stretched response time between two consecutive ACK messages. By default +ACK messages are sent every 10ms (so this interval is not dependent on the network +latency), and so should be the interval between two consecutive received ACK +messages. Note, however, that the network jitter on the public internet causes +these intervals to be stretched, even to multiples of that interval. Both large +and small values of this option have consequences: + +- Large values of this option prevent overreaction on highly stretched response +times, but introduce a latency penalty - the latency must be greater +than this value (otherwise switching to another link won't preserve +smooth signal sending). Large values will also contribute to higher packet +bursts sent at the moment when an idle link is activated. + +- Smaller values of this option respect low latency requirements very +well, but may cause overreaction on even slightly stretched response times. This is +unwanted, as a link switch should ideally happen only when the currently active +link is really broken, as every link switch costs extra overhead (it counts +for 100% for a time of one ACK interval). + +- Note that the value of this option is not allowed to exceed the value of +`SRTO_PEERIDLETIMEO`. Usually it is only meaningful if you change the latter +option, as the default value of it is way above any sensible value of +`SRTO_GROUPSTABTIMEO`. + + + +[Return to list](#list-of-options) + + +#### SRTO_GROUPTYPE + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| -------------------- | ----- | ------- | ---------- | ------ | -------- | ------ | --- | ------ | +| `SRTO_GROUPTYPE` | | pre | `int32_t` | enum | | | R | S | + +- This option is read-only and it is intended to be called inside the listener +callback handler (see `srt_listen_callback`). Possible values are defined in +the `SRT_GROUP_TYPE` enumeration type. + +- This option returns the group type that is declared in the incoming connection. +If the incoming connection is not going to make a group-member connection, then +the value returned is `SRT_GTYPE_UNDEFINED`. If this option is read in any other +context than inside the listener callback handler, the value is undefined. + + +[Return to list](#list-of-options) + + + +#### SRTO_INPUTBW + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| ---------------- | ----- | ------- | ---------- | ------ | -------- | ------ | --- | ------ | +| `SRTO_INPUTBW` | 1.0.5 | post | `int64_t` | B/s | 0 | 0.. | W | GSD | + +- This option is effective only if `SRTO_MAXBW` is set to 0 (relative). It +controls the maximum bandwidth together with `SRTO_OHEADBW` option according +to the formula: `MAXBW = INPUTBW * (100 + OHEADBW) / 100`. When this option +is set to 0 (automatic) then the real INPUTBW value will be estimated from +the rate of the input (cases when the application calls the `srt_send*` +function) during transmission. + +- *Recommended: set this option to the anticipated bitrate of your live stream +and keep the default 25% value for `SRTO_OHEADBW`*. + + +[Return to list](#list-of-options) + + + +#### SRTO_IPTOS + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| ---------------- | ----- | ------- | ---------- | ------ | -------- | ------ | --- | ------ | +| `SRTO_IPTOS` | 1.0.5 | pre | `int32_t` | | (system) | 0..255 | RW | GSD | + +- IPv4 Type of Service (see `IP_TOS` option for IP) or IPv6 Traffic Class (see `IPV6_TCLASS` +of IPv6) depending on socket address family. Applies to sender only. + +- When *getting*, the returned value is the user preset for non-connected sockets +and the actual value for connected sockets. + +- *Sender: user configurable, default: 0xB8* + + +[Return to list](#list-of-options) + + + +#### SRTO_IPTTL + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| ---------------- | ----- | ------- | ---------- | ------ | -------- | ------ | --- | ------ | +| `SRTO_IPTTL` | 1.0.5 | pre | `int32_t` | hops | (system) | 1..255 | RW | GSD | + +- IPv4 Time To Live (see `IP_TTL` option for IP) or IPv6 unicast hops (see +`IPV6_UNICAST_HOPS` for IPv6) depending on socket address family. Applies to sender only. + +- When *getting*, the returned value is the user preset for non-connected sockets +and the actual value for connected sockets. + +- *Sender: user configurable, default: 64* + + +[Return to list](#list-of-options) + + + +#### SRTO_IPV6ONLY + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| ---------------- | ----- | ------- | ---------- | ------ | -------- | ------ | --- | ------ | +| `SRTO_IPV6ONLY` | 1.4.0 | pre | `int32_t` | | (system) | -1..1 | RW | GSD | + +- Set system socket flag `IPV6_V6ONLY`. 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. + + +[Return to list](#list-of-options) + + +#### SRTO_ISN + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| ---------------- | ----- | ------- | ---------- | ------ | -------- | ------ | --- | ------ | +| `SRTO_ISN` | 1.3.0 | | `int32_t` | | | | R | S | + +- The value of the ISN (Initial Sequence Number), which is the first sequence + number put on the first UDP packets sent that are carrying an SRT data payload. + +- *This value is useful for developers of some more complicated methods of flow +control, possibly with multiple SRT sockets at a time. It is not intended to be +used in any regular development.* + + +[Return to list](#list-of-options) + + + +#### SRTO_KMPREANNOUNCE + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| --------------------- | ----- | ------- | ---------- | ------ | -------- | ------ | --- | ------ | +| `SRTO_KMPREANNOUNCE` | 1.3.2 | pre | `int32_t` | pkts | 0x1000 | 0.. * | RW | GSD | + +- The interval (defined in packets) between when a new Stream Encrypting Key +(SEK) is sent and when switchover occurs. This value also applies to the +subsequent interval between when switchover occurs and when the old SEK is +decommissioned. + +At `SRTO_KMPREANNOUNCE` packets before switchover the new key is sent +(repeatedly, if necessary, until it is confirmed by the receiver). + +At the switchover point (see `SRTO_KMREFRESHRATE`), the sender starts +encrypting and sending packets using the new key. The old key persists in case +it is needed to decrypt packets that were in the flight window, or +retransmitted packets. + +The old key is decommissioned at `SRTO_KMPREANNOUNCE` packets after switchover. + +The allowed range for this value is between 1 and half of the current value of +`SRTO_KMREFRESHRATE`. The minimum value should never be less than the flight +window (i.e. the number of packets that have already left the sender but have +not yet arrived at the receiver). + + +[Return to list](#list-of-options) + + + +#### SRTO_KMREFRESHRATE + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| --------------------- | ----- | ------- | ---------- | ------ | -------- | ------ | --- | ------ | +| `SRTO_KMREFRESHRATE` | 1.3.2 | pre | `int32_t` | pkts | 0x1000000| 0.. | RW | GSD | + +- The number of packets to be transmitted after which the Stream Encryption Key +(SEK), used to encrypt packets, will be switched to the new one. Note that +the old and new keys live in parallel for a certain period of time (see +`SRTO_KMPREANNOUNCE`) before and after the switchover. + +- Having a preannounce period before switchover ensures the new SEK is installed +at the receiver before the first packet encrypted with the new SEK is received. +The old key remains active after switchover in order to decrypt packets that +might still be in flight, or packets that have to be retransmitted. + + +[Return to list](#list-of-options) + + + +#### SRTO_KMSTATE + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| --------------------- | ----- | ------- | ---------- | ------ | -------- | ------ | --- | ------ | +| `SRTO_KMSTATE` | 1.0.2 | | `int32_t` | enum | | | R | S | + +- Keying Material state. This is a legacy option that is equivalent to +`SRTO_SNDKMSTATE`, if the socket has set `SRTO_SENDER` to true, and +`SRTO_RCVKMSTATE` otherwise. This option is then equal to `SRTO_RCVKMSTATE` +always if your application disregards possible cooperation with a peer older +than 1.3.0, but then with the default value of `SRTO_ENFORCEDENCRYPTION` the +value returned by both options is always the same. See [`SRT_KM_STATE`](#2-srt_km_state) +for more details. + + +[Return to list](#list-of-options) + + + +#### SRTO_LATENCY + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| --------------------- | ----- | ------- | ---------- | ------ | -------- | ------ | --- | ------ | +| `SRTO_LATENCY` | 1.0.2 | pre | `int32_t` | ms | 120 * | 0.. | RW | GSD | + +- 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. +Bidirectional stream sending in version 1.2.0 was not supported. + + +[Return to list](#list-of-options) + + + +#### SRTO_LINGER + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| -------------------- | ----- | ------- | ---------- | ------ | -------- | ------ | --- | ------ | +| `SRTO_LINGER` | | pre | `linger` | s | on, 180 | 0.. | RW | GSD | + +- Linger time on close (see [SO\_LINGER](http://man7.org/linux/man-pages/man7/socket.7.html)). + +- *SRT recommended value: off (0)*. + + +[Return to list](#list-of-options) + + + +#### SRTO_LOSSMAXTTL + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| -------------------- | ----- | ------- | ---------- | ------- | -------- | ------ | --- | ------ | +| `SRTO_LOSSMAXTTL` | 1.2.0 | pre | `int32_t` | packets | 0 | 0.. | RW | GSD+ | + +- The value up to which the *Reorder Tolerance* may grow. The *Reorder Tolerance* +is the number of packets that must follow the experienced "gap" in sequence numbers +of incoming packets so that the loss report is sent (in the hope that the gap is due +to packet reordering rather than because of loss). The value of *Reorder Tolerance* +starts from 0 and is set to a greater value when packet reordering is detected +This happens when a "belated" packet, with sequence number older than the latest +received, has been received, but without retransmission flag. When this is detected +the *Reorder Tolerance* is set to the value of the interval between latest sequence +and this packet's sequence, but not more than the value set by `SRTO_LOSSMAXTTL`. +By default this value is set to 0, which means that this mechanism is off. + + + +[Return to list](#list-of-options) + + + +#### SRTO_MAXBW + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| -------------------- | ----- | ------- | ---------- | ------- | -------- | ------ | --- | ------ | +| `SRTO_MAXBW` | 1.0.5 | pre | `int64_t` | B/s | -1 | -1.. | RW | GSD | + +- Maximum send bandwidth. +- `-1`: infinite (the limit in Live Mode is 1 Gbps) +- ` 0`: relative to input rate (see [`SRTO_INPUTBW`](#SRTO_INPUTBW)) +- `>0`: absolute limit in B/s + +- *NOTE: This option has a default value of -1, regardless of the mode. +For live streams it is typically recommended to set the value 0 here and rely +on `SRTO_INPUTBW` and `SRTO_OHEADBW` options. However, if you want to do so, +you should make sure that your stream has a fairly constant bitrate, or that +changes are not abrupt, as high bitrate changes may work against the +measurement. SRT cannot ensure that this is always the case for a live stream, +therefore the default -1 remains even in live mode.* + + +[Return to list](#list-of-options) + + + +#### SRTO_MESSAGEAPI + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| -------------------- | ----- | ------- | ---------- | ------- | -------- | ------ | --- | ------ | +| `SRTO_MESSAGEAPI` | 1.3.0 | pre | `bool` | | true | | W | GSD | + +- When set, this socket uses the Message API[\*], otherwise it uses the +Stream API. Note that in live mode (see [`SRTO_TRANSTYPE`](#SRTO_TRANSTYPE) option) only the +Message API is available. In File mode you can chose to use one of two modes +(note that the default for this option is changed with `SRTO_TRANSTYPE` +option): + + - **Stream API** (default for file mode): In this mode you may send + as many data as you wish with one sending instruction, or even use dedicated + functions that operate directly on a file. The internal facility will take care + of any speed and congestion control. When receiving, you can also receive as + many data as desired. The data not extracted will be waiting for the next call. + There is no boundary between data portions in Stream mode. + + - **Message API**: In this mode your single sending instruction passes exactly + one piece of data that has boundaries (a message). Contrary to Live mode, + this message may span multiple UDP packets, and the only size limitation + is that it shall fit as a whole in the sending buffer. The receiver shall use + as large a buffer as necessary to receive the message, otherwise reassembling + and delivering the message might not be possible. When the message is not + complete (not all packets received or there was a packet loss) it will not be + copied to the application's buffer. Messages that are sent later, but were + earlier reassembled by the receiver, will be delivered once ready, if the + `inorder` flag was set to false. + See [`srt_sendmsg`](https://github.com/Haivision/srt/blob/master/docs/API.md#sending-and-receiving)). + +- As a comparison to the standard system protocols, the Stream API does +transmission similar to TCP, whereas the Message API functions like the +SCTP protocol. + + +[Return to list](#list-of-options) + + + +#### SRTO_MINVERSION + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| -------------------- | ----- | ------- | ---------- | ------- | -------- | ------ | --- | ------ | +| `SRTO_MINVERSION` | 1.3.0 | pre | `int32_t` | version | 0 | * | W | GSD | + +- The minimum SRT version that is required from the peer. A connection to a +peer that does not satisfy the minimum version requirement will be rejected. +See [`SRTO_VERSION`](#SRTO_VERSION) for the version format. + + +[Return to list](#list-of-options) + + + +#### SRTO_MSS + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| -------------------- | ----- | ------- | ---------- | ------- | -------- | ------ | --- | ------ | +| `SRTO_MSS` | | pre | `int32_t` | bytes | 1500 | 76.. | RW | GSD | + +- Maximum Segment Size. Used for buffer allocation and rate calculation using +packet counter assuming fully filled packets. Each party can set its own MSS +value independently. During a handshake the parties exchange MSS values, and +the lowest is used. + +*Generally on the internet MSS is 1500 by default. This is the maximum +size of a UDP packet and can be only decreased, unless you have some unusual +dedicated network settings. MSS is not to be confused with the size of the UDP +payload or SRT payload - this size is the size of the IP packet, including the +UDP and SRT headers* + + +[Return to list](#list-of-options) + + + +#### SRTO_NAKREPORT + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| -------------------- | ----- | ------- | ---------- | ------- | -------- | ------ | --- | ------ | +| `SRTO_NAKREPORT` | 1.1.0 | pre | `bool` | | * | | RW | GSD+ | + +- When set to true, every report for a detected loss will be repeated when the +timeout for the expected retransmission of this loss has expired and the +missing packet still wasn't recovered, or wasn't conditionally dropped (see +[`SRTO_TLPKTDROP`](#SRTO_TLPKTDROP)). + +- The default is true for Live mode, and false for File mode (see [`SRTO_TRANSTYPE`](#SRTO_TRANSTYPE)). + + +[Return to list](#list-of-options) + + + +#### SRTO_OHEADBW + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| -------------------- | ----- | ------- | ---------- | ------- | -------- | ------ | --- | ------ | +| `SRTO_OHEADBW` | 1.0.5 | post | `int32_t` | % | 25 | 5..100 | W | GSD | + +- Recovery bandwidth overhead above input rate (see `[`SRTO_INPUTBW`](#SRTO_INPUTBW)`), +in percentage of the input rate. It is effective only if `SRTO_MAXBW` is set to 0. + +- *Sender: user configurable, default: 25%.* + +- Recommendations: + + - *Overhead is intended to give you extra bandwidth for the case when a packet + has taken part of the bandwidth, but then was lost and has to be retransmitted. + Therefore the effective maximum bandwidth should be appropriately higher than + your stream's bitrate so that there's some room for retransmission, but still + limited so that the retransmitted packets don't cause the bandwidth usage to + skyrocket when larger groups of packets are lost* + + - *Don't configure it too low and avoid 0 in the case when you have the + `SRTO_INPUTBW` option set to 0 (automatic). Otherwise your stream will choke + and break quickly at any rise in packet loss.* + +- ***To do: set-only; get should be supported.*** + + +[Return to list](#list-of-options) + + + +#### SRTO_PACKETFILTER + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| -------------------- | ----- | ------- | ---------- | ------- | -------- | ------ | --- | ------ | +| `SRTO_PACKETFILTER` | 1.4.0 | pre | `string` | | "" | [512] | W | GSD | + +- Set up the packet filter. The string must match appropriate syntax for packet +filter setup. + +As there can only be one configuration for both parties, it is recommended that +one party defines the full configuration while the other only defines the matching +packet filter type (for example, one sets `fec,cols:10,rows:-5,layout:staircase` +and the other just `fec`). Both parties can also set this option to the same value. +The packet filter function will attempt to merge configuration definitions, but if +the options specified are in conflict, the connection will be rejected. + +For details, see [Packet Filtering & FEC](packet-filtering-and-fec.md). + + +[Return to list](#list-of-options) + + + +#### SRTO_PASSPHRASE + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| -------------------- | ----- | ------- | ---------- | ------- | -------- | ------ | --- | ------ | +| `SRTO_PASSPHRASE` | 0.0.0 | pre | `string` | | "" |[10..79]| W | GSD | + +- Sets the passphrase for encryption. This enables encryption on this party (or +disables it, if an 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](http://en.wikipedia.org/wiki/PBKDF2) +(Password-Based Key Derivation Function 2). It is used on the receiver only if +the received data is encrypted. + +- Note that since the introduction of bidirectional support, there's only one +initial SEK to encrypt the stream (new keys after refreshing will be updated +independently), and there's no distinction between "service party that defines +the password" and "client party that is required to set matching password" - both +parties are equivalent, and in order to have a working encrypted connection, they +have to simply set the same passphrase. Otherwise the connection is rejected by +default (see also [`SRTO_ENFORCEDENCRYPTION`](#SRTO_ENFORCEDENCRYPTION)). + + +[Return to list](#list-of-options) + + + +#### SRTO_PAYLOADSIZE + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| -------------------- | ----- | ------- | ---------- | ------- | -------- | ------ | --- | ------ | +| `SRTO_PAYLOADSIZE` | 1.3.0 | pre | `int32_t` | bytes | * | * | W | GSD | + +- Sets the maximum declared size of a single call to sending function in Live +mode. When set to 0, there's no limit for a single sending call. + +- For Live mode: Default value is 1316, but can be increased up to 1456. Note that +with the `SRTO_PACKETFILTER` option additional header space is usually required, +which decreases the maximum possible value for `SRTO_PAYLOADSIZE`. + +- For File mode: Default value is 0 and it's recommended not to be changed. + + + +[Return to list](#list-of-options) + + + +#### SRTO_PBKEYLEN + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| -------------------- | ----- | ------- | ---------- | ------- | -------- | ------ | --- | ------ | +| `SRTO_PBKEYLEN` | 0.0.0 | pre | `int32_t` | bytes | 0 | * | RW | GSD | + +- Sender encryption key length. + +- Possible values: + - 0 =`PBKEYLEN` (default value) + - 16 = AES-128 (effective value) + - 24 = AES-192 + - 32 = AES-256 + +- The use is slightly different in 1.2.0 (HSv4), and since 1.3.0 (HSv5): + + - **HSv4**: This is set on the sender and enables encryption, if not 0. The receiver + shall not set it and will agree on the length as defined by the sender. + + - **HSv5**: The "default value" for `PBKEYLEN` is 0, which means that the + `PBKEYLEN` won't be advertised. The "effective value" for `PBKEYLEN` is 16, but + this applies only when neither party has set the value explicitly (i.e. when + both are initially at the default value of 0). If any party *has* set an + explicit value (16, 24, 32) it will be advertised in the handshake. If the other + party remains at the default 0, it will accept the peer's value. The situation + where both parties set a value should be treated carefully. Actually there are + three intended methods of defining it, and all other uses are considered + undefined behavior: + + - **Unidirectional**: the sender shall set `PBKEYLEN` and the receiver shall + not alter the default value 0. The effective `PBKEYLEN` will be the one set + on the sender. The receiver need not know the sender's `PBKEYLEN`, just the + passphrase, `PBKEYLEN` will be correctly passed. + + - **Bidirectional in Caller-Listener arrangement**: it is recommended to use + a rule whereby you will be setting the `PBKEYLEN` exclusively either on the + Listener or on the Caller. The value set on the Listener will win, if set on + both parties. + + - **Bidirectional in Rendezvous arrangement**: you have to know the passphrases + for both parties, as well as `PBKEYLEN`. 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 the Listener in a Caller-Listener arrangement. In Rendezvous it's a + matter of luck which party becomes the Responder. + + + +[Return to list](#list-of-options) + + + +#### SRTO_PEERIDLETIMEO + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| -------------------- | ----- | ------- | ---------- | ------- | -------- | ------ | --- | ------ | +| `SRTO_PEERIDLETIMEO` | 1.3.3 | pre | `int32_t` | ms | 5000 | 0.. | RW | GSD+ | + +- The maximum time in `[ms]` to wait until another packet is received from a peer +since the last such packet reception. If this time is passed, the connection is +considered broken on timeout. + + +[Return to list](#list-of-options) + + + +#### SRTO_PEERLATENCY + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| -------------------- | ----- | ------- | ---------- | ------- | -------- | ------ | --- | ------ | +| `SRTO_PEERLATENCY` | 1.3.0 | pre | `int32_t` | ms | 0 | 0.. | RW | GSD | + +- The latency value (as described in `SRTO_RCVLATENCY`) that is set by the sender +side as a minimum value for the receiver. + +- Note that when reading, the value will report the preset value on a non-connected +socket, and the effective value on a connected socket. + + +[Return to list](#list-of-options) + + + +#### SRTO_PEERVERSION + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| -------------------- | ----- | ------- | ---------- | ------- | -------- | ------ | --- | ------ | +| `SRTO_PEERVERSION` | 1.1.0 | | `int32_t` | * | | | R | GS | + +- SRT version used by the peer. The value 0 is returned if not connected, SRT +handshake not yet performed (HSv4 only), or if peer is not SRT. +See [`SRTO_VERSION`](#SRTO_VERSION) for the version format. + + +[Return to list](#list-of-options) + + + +#### SRTO_RCVBUF + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| -------------------- | ----- | ------- | ---------- | ------- | ---------- | ------ | --- | ------ | +| `SRTO_RCVBUF` | | pre | `int32_t` | bytes | 8192 bufs | * | RW | GSD+ | + + +- Receive Buffer Size, in bytes. Note, however, that the internal setting of this +value is in the number of buffers, each one of size equal to SRT payload size, +which is the value of `SRTO_MSS` decreased by UDP and SRT header sizes (28 and 16). +The value set here will be effectively aligned to the multiple of payload size. + +- Minimum value: 32 buffers (46592 with default value of `SRTO_MSS`). +- Maximum value: `SRTO_FC` number of buffers (receiver buffer must not be greater + than the Flight Flag size). + + +[Return to list](#list-of-options) + + + +#### SRTO_RCVDATA + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| ----------------- | ----- | ------- | ---------- | ------- | ---------- | ------ | --- | ------ | +| `SRTO_RCVDATA` | | | `int32_t` | pkts | | | R | S | + +- Size of the available data in the receive buffer. + + +[Return to list](#list-of-options) + + + +#### SRTO_RCVKMSTATE + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| ----------------- | ----- | ------- | ---------- | ------- | ---------- | ------ | --- | ------ | +| `SRTO_RCVKMSTATE` | 1.2.0 | | `int32_t` | enum | | | R | S | + +- KM state on the agent side when it's a receiver. + +- Values defined in enum [`SRT_KM_STATE`](#2-srt_km_state). + + +[Return to list](#list-of-options) + + + +#### SRTO_RCVLATENCY + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| ----------------- | ----- | ------- | ---------- | ------- | ---------- | ------ | --- | ------ | +| `SRTO_RCVLATENCY` | 1.3.0 | pre | `int32_t` | msec | * | 0.. | RW | GSD | + +- Latency value in the receiving direction. This value is only significant when +`SRTO_TSBPDMODE` is set to true. + +- Latency refers to the time that elapses from the moment a packet is sent +to the moment when it's delivered to a receiver application. The SRT latency +setting should be a buffer large enough to cover the time spent for sending, +unexpectedly extended RTT time, and the time needed to retransmit any +lost UDP packet. The effective latency value will be the maximum between the +`SRTO_RCVLATENCY` 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`. Note that the real latency value may be slightly different +than this setting due to the impossibility of perfectly measuring exactly the +same point in time at both parties simultaneously. What is important with +latency is that its actual value, once set with the connection, is kept constant +throughout the duration of a connection. + +- Default value: 120 in Live mode, 0 in File mode (see [`SRTO_TRANSTYPE`](#SRTO_TRANSTYPE)). + +- Note that when reading, the value will report the preset value on a non-connected +socket, and the effective value on a connected socket. + + +[Return to list](#list-of-options) + + + +#### SRTO_RCVSYN + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| ----------------- | ----- | ------- | ---------- | ------- | ---------- | ------ | --- | ------ | +| `SRTO_RCVSYN` | | post | `bool` | | true | | RW | GSI | + +- When true, sets blocking mode on reading function when it's not ready to +perform the operation. When false ("non-blocking mode"), the reading function +will in this case report error `SRT_EASYNCRCV` and return immediately. Details +depend on the tested entity: + +- On a connected socket or group this applies to a receiving function +(`srt_recv` and others) and a situation when there are no data available for +reading. The readiness state for this operation can be tested by checking the +`SRT_EPOLL_IN` flag on the aforementioned socket or group. + +- On a freshly created socket or group that is about to be connected to a peer +listener this applies to any `srt_connect` call (and derived), which in +"non-blocking mode" always returns immediately. The connected state for that +socket or group can be tested by checking the `SRT_EPOLL_OUT` flag. Note +that a socket that failed to connect doesn't change the `SRTS_CONNECTING` +state and can be found out only by testing the `SRT_EPOLL_ERR` flag. + +- On a listener socket this applies to `srt_accept` call. The readiness state +for this operation can be tested by checking the `SRT_EPOLL_IN` flag on +this listener socket. This flag is also derived from the listener socket +by the accepted socket or group, although the meaning of this flag is +effectively different. + +- Note that when this flag is set only on a group, it applies to a +specific receiving operation being done on that group (i.e. it is not +derived from the socket of which the group is a member). + + + +[Return to list](#list-of-options) + + + +#### SRTO_RCVTIMEO + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| ----------------- | ----- | ------- | ---------- | ------- | ---------- | ------ | --- | ------ | +| `SRTO_RCVTIMEO` | | post | `int32_t` | ms | -1 | -1, 0..| RW | GSI | + +- Limits the time up to which the receiving operation will block (see +[`SRTO_RCVSYN`](#SRTO_RCVSYN) for details), such that when this time is exceeded, +it will behave as if in "non-blocking mode". The -1 value means no time limit. + + +[Return to list](#list-of-options) + + + +#### SRTO_RENDEZVOUS + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| ----------------- | ----- | ------- | ---------- | ------- | ---------- | ------ | --- | ------ | +| `SRTO_RENDEZVOUS` | | pre | `bool` | | false | | RW | S | + +- Use Rendezvous connection mode (both sides must set this and both must use the +procedure of `srt_bind` and then `srt_connect` (or `srt_rendezvous`) to one another. + + +[Return to list](#list-of-options) + + + +#### SRTO_RETRANSMITALGO + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| --------------------- | ----- | ------- | ------ | ------ | ------- | ------ | --- | ------ | +| `SRTO_RETRANSMITALGO` | 1.5.0 | pre | `int` | | 0 | [0, 1] | W | GSD | + +- Retransmission algorithm to use (SENDER option): + - 0 - Default (retransmit on every loss report). + - 1 - Reduced retransmissions (not more often than once per RTT); reduced + bandwidth consumption. + +- This option is effective only on the sending side. It influences the decision +as to whether particular reported lost packets should be retransmitted at a +certain time or not. + + +[Return to list](#list-of-options) + + + +#### SRTO_REUSEADDR + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| ----------------- | ----- | ------- | ---------- | ------- | ---------- | ------ | --- | ------ | +| `SRTO_REUSEADDR` | | pre | `bool` | | true | | RW | GSD | + +- When true, allows the SRT socket to use the binding address used already by +another SRT socket in the same application. Note that SRT socket uses an +intermediate object called Multiplexer to access the underlying UDP sockets, +so multiple SRT sockets may share one UDP socket, and the packets received by this +UDP socket will be correctly dispatched to the SRT socket to which they are +currently destined. This has some similarities to the `SO_REUSEADDR` system socket +option, although it's only used inside SRT. + +- *TODO: This option weirdly only allows the socket used in **bind()** to use the +local address that another socket is already using, but not to disallow another +socket in the same application to use the binding address that the current +socket is already using. What it actually changes is that when given an address in +**bind()** is already used by another socket, this option will make the binding +fail instead of adding the socket to the shared group of that socket that +already has bound this address - but it will not disallow another socket to reuse +its address.* + + +[Return to list](#list-of-options) + + + +#### SRTO_SENDER + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| ----------------- | ----- | ------- | ---------- | ------- | ---------- | ------ | --- | ------ | +| `SRTO_SENDER` | 1.0.4 | pre | `bool` | | false | | W | S | + +- Set sender side. The side that sets this flag is expected to be a sender. This +flag is only required when communicating with a receiver that uses SRT version +less than 1.3.0 (and hence *HSv4* handshake), in which case if not set properly, +the TSBPD mode (see [`SRTO_TSBPDMODE`](#SRTO_TSBPDMODE)) or encryption will not +work. Setting `SRTO_MINVERSION` to 1.3.0 is therefore recommended. + + +[Return to list](#list-of-options) + + + +#### SRTO_SNDBUF + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| ----------------- | ----- | ------- | ---------- | ------- | --------- | ------ | --- | ------ | +| `SRTO_SNDBUF` | | pre | `int32_t` | bytes |8192 bufs | * | RW | GSD+ | + +- Sender Buffer Size. See [`SRTO_RCVBUF`](#SRTO_RCVBUF) for more information. + + +[Return to list](#list-of-options) + + + +#### SRTO_SNDDATA + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| ----------------- | ----- | ------- | ---------- | ------- | --------- | ------ | --- | ------ | +| `SRTO_SNDDATA` | | | `int32_t` | pkts | | | R | S | + +- Size of the unacknowledged data in send buffer. + + +[Return to list](#list-of-options) + + + +#### SRTO_SNDDROPDELAY + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| -------------------- | ----- | ------- | ---------- | ------- | --------- | ------ | --- | ------ | +| `SRTO_SNDDROPDELAY` | 1.3.2 | pre | `int32_t` | ms | * | -1.. | W | GSD+ | + +- Sets an extra delay before `TLPKTDROP` is triggered on the data sender. +This delay is added to the default drop delay time interval value. Keep in mind +that the longer the delay, the more probable it becomes that packets would be +retransmitted uselessly because they will be dropped by the receiver anyway. + +- `TLPKTDROP` discards packets reported as lost if it is already too late to send +them (the receiver would discard them even if received). The delay before the +`TLPKTDROP` mechanism is triggered consists of the SRT latency (`SRTO_PEERLATENCY`), +plus `SRTO_SNDDROPDELAY`, plus `2 * interval between sending ACKs` (where the +default `interval between sending ACKs` is 10 milliseconds). +The minimum delay is `1000 + 2 * interval between sending ACKs` milliseconds. + +- **Special value -1**: Do not drop packets on the sender at all (retransmit them + always when requested). + +- Default: 0 in Live mode, -1 in File mode. + + + +[Return to list](#list-of-options) + + + +#### SRTO_SNDKMSTATE + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| -------------------- | ----- | ------- | ---------- | ------- | --------- | ------ | --- | ------ | +| `SRTO_SNDKMSTATE` | 1.2.0 | post | `int32_t` | enum | | | R | S | + +- Peer KM state on receiver side for `SRTO_KMSTATE` + +- Values defined in enum [`SRT_KM_STATE`](#2-srt_km_state). + + +[Return to list](#list-of-options) + + + +#### SRTO_SNDSYN + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| -------------------- | ----- | ------- | ---------- | ------- | --------- | ------ | --- | ------ | +| `SRTO_SNDSYN` | | post | `bool` | | true | | RW | GSI | + +- When true, sets blocking mode on writing function when it's not ready to +perform the operation. When false ("non-blocking mode"), the writing function +will in this case report error `SRT_EASYNCSND` and return immediately. + +- On a connected socket or group this applies to a sending function +(`srt_send` and others) and a situation when there's no free space in +the sender buffer, caused by inability to send all the scheduled data over +the network. Readiness for this operation can be tested by checking the +`SRT_EPOLL_OUT` flag. + +- On a freshly created socket or group it will have no effect until the socket +enters a connected state. + +- On a listener socket it will be derived by the accepted socket or group, +but will have no effect on the listener socket itself. + + +[Return to list](#list-of-options) + + + +#### SRTO_SNDTIMEO + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| -------------------- | ----- | ------- | ---------- | ------- | --------- | ------ | --- | ------ | +| `SRTO_SNDTIMEO` | | post | `int32_t` | ms | -1 | -1.. | RW | GSI | + +- limit the time up to which the sending operation will block (see +`SRTO_SNDSYN` for details), so when this time is exceeded, it will behave as +if in "non-blocking mode". The -1 value means no time limit. + + +[Return to list](#list-of-options) + + + +#### SRTO_STATE + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| -------------------- | ----- | ------- | ---------- | ------- | --------- | ------ | --- | ------ | +| `SRTO_STATE` | | | `int32_t` | enum | | | R | S | + +- Returns the current socket state, same as `srt_getsockstate`. + + + +[Return to list](#list-of-options) + + + +#### SRTO_STREAMID + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| -------------------- | ----- | ------- | ---------- | ------- | --------- | ------ | --- | ------ | +| `SRTO_STREAMID` | 1.3.0 | pre | `string` | | "" | [512] | RW | GSD | + +- A string that can be set on the socket prior to connecting. The listener side +will be able to retrieve this stream ID from the socket that is returned from +`srt_accept` (for a connected socket with that stream ID). You usually use SET +on the socket used for `srt_connect`, and GET on the socket retrieved from +`srt_accept`. This string can be used completely free-form. However, it's highly +recommended to follow the [SRT Access Control guidlines](AccessControl.md). + +- As this uses internally the `std::string` type, there are additional functions +for it in the legacy/C++ API (udt.h): `srt::setstreamid` and `srt::getstreamid`. + +- This option is not useful for a Rendezvous connection, since one side would +override the value from the other side resulting in an arbitrary winner. Also +in this connection both peers are known to one another and both have equivalent +roles in the connection. + + +[Return to list](#list-of-options) + + + +#### SRTO_TLPKTDROP + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| ----------------- | ----- | ------- | ---------- | ------- | --------- | ------ | --- | ------ | +| `SRTO_TLPKTDROP` | 1.0.6 | pre | `bool` | | * | | RW | GSD | + +- Too-late Packet Drop. When enabled on receiver, it skips missing packets that +have not been delivered in time and delivers the subsequent packets to the +application when their time-to-play has come. It also sends a fake ACK to the +sender. When enabled on sender and enabled on the receiving peer, sender drops +the older packets that have no chance to be delivered in time. It is automatically +enabled in sender if receiver supports it. + +- Default: true in Live mode, false in File mode (see [`SRTO_TRANSTYPE`](#SRTO_TRANSTYPE)) + + +[Return to list](#list-of-options) + + + +#### SRTO_TRANSTYPE + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| ----------------- | ----- | ------- | ---------- | ------- | --------- | ------ | --- | ------ | +| `SRTO_TRANSTYPE` | 1.3.0 | pre | `int32_t` | enum |`SRTT_LIVE`| * | W | S | + +- Sets the transmission type for the socket, in particular, setting this option +sets multiple other parameters to their default values as required for a +particular transmission type. + +- Values defined by enum `SRT_TRANSTYPE` (see above for possible values) + + +[Return to list](#list-of-options) + + + +#### SRTO_TSBPDMODE + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| ----------------- | ----- | ------- | ---------- | ------- | --------- | ------ | --- | ------ | +| `SRTO_TSBPDMODE` | 0.0.0 | pre | `bool` | | * | | W | S | + +- When true, use Timestamp-based Packet Delivery mode. In this mode the +packet's time is assigned at the sending time (or allowed to be predefined), +transmitted in the packet's header, and then restored on the receiver side so that +the time intervals between consecutive packets are preserved when delivering to +the application. + +- Default: true in Live mode, false in File mode (see [`SRTO_TRANSTYPE`](#SRTO_TRANSTYPE)). + + +[Return to list](#list-of-options) + + + +#### SRTO_UDP_RCVBUF + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| ----------------- | ----- | ------- | ---------- | ------- | --------- | ------ | --- | ------ | +| `SRTO_UDP_RCVBUF` | | pre | `int32_t` | bytes | 8192 bufs | * | RW | GSD+ | + +- UDP Socket Receive Buffer Size. Configured in bytes, maintained in packets +based on MSS value. Receive buffer must not be greater than FC size. + + +[Return to list](#list-of-options) + + + +#### SRTO_UDP_SNDBUF + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| ----------------- | ----- | ------- | ---------- | ------- | --------- | ------ | --- | ------ | +| `SRTO_UDP_SNDBUF` | | pre | `int32_t` | bytes | 65536 | * | RW | GSD+ | + +- UDP Socket Send Buffer Size. Configured in bytes, maintained in packets based +on `SRTO_MSS` value. + + +[Return to list](#list-of-options) + + + +#### SRTO_VERSION + +| OptName | Since | Binding | Type | Units | Default | Range | Dir | Entity | +| ----------------- | ----- | ------- | ---------- | ------- | --------- | ------ | --- | ------ | +| `SRTO_VERSION` | 1.1.0 | | `int32_t` | | | | R | S | + +- Local SRT version. This is the highest local version supported if not + connected, or the highest version supported by the peer if connected. + +- The version format in hex is `0x00XXYYZZ` for x.y.z in human readable form. +For example, version 1.4.2 is encoded as `0x010402`. + + +[Return to list](#list-of-options) + + +