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Introduction into srt-live-transmit

The srt-live-transmit tool is a universal data transport tool, which's intention is to transport data between SRT and other medium. At the same time it is just a sample application to show some of the powerful features of SRT. We encourage you to use SRT library itself integrated into your products.

The srt-live-transmit can be both used as a universal SRT-to-something-else flipper, as well as a testing tool for SRT.

The general usage is the following:

srt-live-transmit <input-uri> <output-uri> [options]

The following medium types are handled by srt-live-transmit:

  • SRT - use SRT for reading or writing, in listener, caller or rendezvous mode, with possibly additional parameters
  • UDP - read or write the given UDP address (also multicast)
  • Local file - read or store the stream into the file
  • Process's pipeline - use the process's stdin and stdout standard streams

Any medium can be used with any direction, although some of them may have special direction-dependent cases.

Mind that the URI has a standard syntax:

scheme://HOST:PORT/PATH?PARAM1=VALUE1&PARAM2=VALUE2&...

The first parameter is introduced with a ? and all following can be appended with an & character.

If you specify only the path (no :// specified), then the scheme defaults to file. The path can be also specified as relative this way. Note also that empty host (scheme://:PORT) defaults to 0.0.0.0, and an empty port (when there's no :PORT part) defaults to port number 0.

Special options for particular medium may be specified in PARAM items. All options are medium-specific, although there may happen some options common for multiple media types.

Note also that the HOST part is always tried to be resolved as a name, if its form is not directly the IPv4 address.

Example for Smoke Testing

First we need to start up the srt-live-transmit app, listening for unicast UDP TS input on port 1234 and making SRT available on port 4201. Note, these are randomly chosen ports. We also open the app in verbose mode for debugging:

srt-live-transmit udp://:1234 srt://:4201 -v

Now we need to generate a UDP stream. ffmpeg can be used to generate bars and tone as follows, doing a simple unicast push to our listening srt-live-transmit application:

ffmpeg -f lavfi -re -i smptebars=duration=300:size=1280x720:rate=30 -f lavfi -re -i sine=frequency=1000:duration=60:sample_rate=44100 -pix_fmt yuv420p -c:v libx264 -b:v 1000k -g 30 -keyint_min 120 -profile:v baseline -preset veryfast -f mpegts "udp://127.0.0.1:1234?pkt_size=1316"

You should see the stream connect in srt-live-transmit.

Now you can test in VLC (make sure you're using the latest version!) - just go to file -> open network stream and enter srt://127.0.0.1:4201 and you should see bars and tone right away.

If you're having trouble, make sure this works, then add complexity one step at a time (multicast, push vs listen, etc.).

URI Syntaxis

Transmission mediums are specified as the standard URI format:

SCHEME://HOST:PORT?PARAM1=VALUE1&PARAM2=VALUE2&...

The applications supports the following schemes:

  • file - for file or standard input and output
  • udp - UDP output (unicast and multicast)
  • srt - SRT connection

Note that this application doesn't support file as a medium, but this can be handled by other applications from this project.

Medium: FILE (including standard process pipes)

NB! File mode, except file://con, is not supported in the srt-file-transmit tool!

The general syntax is: file:///global/path/to/the/file. No parameters in the URL are extracted. There's one (non-standard!) special case, though:

file://con

That is, con is used as a HOST part of the URI. If you use this URI for <input-uri>, then the data will be read from the standard input. If <output-uri>, the data will be send to the standard output. Be careful with options being specified together with having standard output as output URI - some of them are not allowed as the extra output controlled by options might interfere with the data output.

Medium: UDP

UDP can only be used in listening mode for input, and in calling mode for output. Multicast Streaming is also possible, whithout any special declaration. Just use an IP address from the multicast range. The specification and meaning of the fields in the URI depend on the mode.

The PORT part is always mandatory and it designates either the port number for the target host or the port number to be bound to read from.

For sending to unicast:

udp://TARGET:PORT?parameters...

  • The HOST part (here: TARGET) is mandatory and designates the target host

  • The iptos parameter designates the Type-Of-Service (TOS) field for outgoing packets via IP_TOS socket option.

  • The ttl parameter will set time-to-live value for outgoing packets via IP_TTL socket options.

For receiving from unicast:

udp://LOCALADDR:PORT?parameters...
  • The HOST part (here: LOCALADDR) designates the local interface to bind. It's optional (can be empty) and defaults to 0.0.0.0 (INADDR_ANY).

For multicast the scheme is:

udp://GROUPADDR:PORT?parameters...
  • The HOST part (here: GROUPADDR) is mandatory always and designates the target multicast group. The @ character is handled in this case, but it's not necessary, as the IGMP addresses are recognized by their mask.

For sending to a multicast group:

  • The iptos parameter designates the Type-Of-Service (TOS) field for outgoing packets via IP_TOS socket option.

  • The ttl parameter will set time-to-live value for outgoing packets via IP_MULTICAST_TTL socket options.

  • The adapter parameter can be used to specify the adapter to be set through IP_MULTICAST_IF option to override the default device used for sending

For receiving from a multicast group:

  • The adapter parameter can be used to specify the adapter through which the given multicast group can be reached (it's used to bind the socket)

  • The source parameter enforces the use of IP_ADD_SOURCE_MEMBERSHIP instead of IP_ADD_MEMBERSHIP and the value is set to imr_sourceaddr field.

Explanations for the symbols and terms used above can be found in POSIX manual pages, like ip(7) and on Microsoft docs pages under IPPROTO_IP.

Medium: SRT

Most important about SRT is that it can be either input or output and in both these cases it can work in listener, caller and rendezvous mode. SRT also handles several parameters special way, in addition to standard SRT options that can be set through the parameters.

SRT can be connected using one of three connection modes:

  • caller: the "agent" (this application) sends the connection request to the peer, which must be listener, and this way it initiates the connection.

  • listener: the "agent" waits to be contacted by any peer caller. Note that a listener can accept multiple callers, but srt-live-transmit does not use this ability; after the first connection, it no longer accepts new connections.

  • rendezvous: A one-to-one only connection where both parties are equivalent and both connect to one another simultaneously. Whoever happened to start first (or succeeded to punch through the firewall) is meant to have initiated the connection.

This mode can be specified explicitly using the mode parameter. When it's not specified, then it is "deduced" the following way:

  • srt://:1234 - the port is specified (1234), but host is empty. This assumes listener mode.
  • srt://remote.host.com:1234 - both host and port are specified. This assumes caller mode.

When the mode parameter is specified explicitly, then the interpretation of the host part is the following:

  • For caller, it's always the destination host address. If this is empty, it is resolved to 0.0.0.0, which usually should mean connecting to the local host

  • For listener, it defines the IP address of the local device on which the socket should listen, e.g.:

srt://10.10.10.100:5001?mode=listener

An alternative method to specify this IP address is the adapter parameter:

srt://:5001?adapter=10.10.10.100

The rendezvous mode is not deduced and it has to be specified explicitly. Note also special cases of the host and port parts specified in the URI:

  • CALLER: the host and port parts are mandatory and specify the remote host and port to be contacted.
    • The port parameter can be used to enforce the local outgoing port (not to be confused with remote port!).
    • The adapter parameter is not used.
  • LISTENER: the port part is mandatory and it specifies the local listening port.
    • The adapter parameter can be used to specify the adapter.
    • The host part, if specified, can be also used to set the adapter - although in this case mode=listener must be set explicitly.
    • The port parameter is not used.
  • RENDEZVOUS: the host and port parts are mandatory.
    • The host part specifies the remote host to contact.
    • The port part specifies both local and remote port. Note that the local port is this way both listening port and outgoing port.
    • The adapter parameter can be used to specify the adapter.
    • The port parameter is not used.

Some parameters handled for SRT medium are specific, all others are socket options. The following parameters are handled special way by srt-live-transmit:

  • mode: enforce caller, listener or rendezvous mode
  • port: enforce the outgoing port (the port number that will be set in the UDP packet as a source port when sent from this host). This can be used only in caller mode.
  • blocking: sets the SRTO_RCVSYN for input medium or SRTO_SNDSYN for output medium
  • timeout: sets SRTO_RCVTIMEO for input medium or SRTO_SNDTIMEO for output medium
  • adapter: sets the adapter for listening in listener or rendezvous mode

All other parameters are SRT socket options. Here are some most characteristic options:

  • latency: Sets the maximum accepted transmission latency and should be >= 2.5 times the RTT (default: 120ms; when both parties set different values, the maximum of the two is used for both)
  • passphrase: Sets the password for the encrypted transmission.
  • pbkeylen: Crypto key len in bytes {16,24,32} Default: 16 (128-bit)
  • tlpktdrop: Whether to drop packets that are not delivered on time. Default is on.
  • conntimeo: Connection timeout (in ms). Caller default: 3000, rendezvous (x 10)

For the complete list of options, please refer to the SRT header file srt.h and search for SRT_SOCKOPT enum type. Please note that the set of available options may be version dependent. All options are available under the lowercase name of the option without the SRTO_ prefix. For example, SRTO_PASSPHRASE can be set using a passphrase parameter. The mapping table srt_options can be found in common/socketoptions.hpp file.

Important thing about the options (which holds true also for options for TCP and UDP, even though it's not described anywhere explicitly) is that there are two categories of options:

  • PRE options: these options must be set to the socket prior to connecting and they cannot be altered after the connection is made. A PRE option set to a listening socket will be also derived by the socket returned by srt_accept().
  • POST options: these options can be set to a socket at any time. The option set to a listening socket will not be derived by an accepted socket.

You don't have to worry about that actually - the application is aware of this and it sets these options at appropriate time.

Note also that blocking option has no practical use for users. Normally the non-blocking mode is used only when you have an event-driven application that needs a common signal bar for multiple event sources, or you prefer fibers to threads, when working with multiple SRT sockets in one application. The srt-live-transmit application isn't defined this way. This makes that the practical result of non-blocking mode here is that it uses polling on exactly one socket with infinite timeout. Every reading and writing operation will then return always without blocking, but when they report the "again" situation the application will stall on srt_epoll_wait() call. This option then exists for the testing purposes, as well as educational, to serve as an example of how your application should use the non-blocking mode.

Command-Line Options

The following options are available in the application. Note that some may affect specifically only selected type of medium.

Options usually have values and they are set using colon: for example, -t:60. Alternatively you can also separate them by a space, but this space must be part of the parameter and not extracted by a shell (using " " quotes or backslash).

  • -timeout, -t, -to - Sets the timeout for any activity from any medium (in seconds). Default is 0 for infinite (that is, turn this mechanism off). The mechanism is such that the SIGALRM is set up to be called after the given time and it's reset after every reading succeeded. When the alarm expires due to no reading activity in defined time, it will break the application. Notes:
    • The alarm is set up after the reading loop has started, not when the application has started. That is, a caller will still wait the standard timeout to connect, and a listener may wait infinitely until some peer connects; only after the connection is established is the alarm counting started.
    • The timeout mechanism doesn't work on Windows at all. It behaves as if the timeout was set to -1 and it's not modifiable.
  • -timeout-mode, -tm - Timeout mode used. Default is 0 - timeout will happen after the specified time. Mode 1 cancels the timeout if the connection was established.
  • -chunk, -c - use given size of the buffer. The default size is 1456 bytes, which is the maximum payload size for a single SRT packet.
  • -verbose, -v - Display additional information on the standard output. Note that it's not allowed to be combined with output specified as file://con.
  • -statsout - SRT statistics output: filename. Without this option specified, the statistics will be printed to the standard output.
  • -pf, -statspf - SRT statistics print format. Values: json, csv, default.
  • -s, -stats, -stats-report-frequency - The frequency of SRT statistics collection, in milliseconds.
  • -loglevel - lowest logging level for SRT, one of: fatal, error, warning, note, debug (default: error)
  • -logfa - selected FAs in SRT to be logged (default: all is enabled, that is, you can filter out log messages from only wanted FAs using this option).
  • -logfile:logs.txt - Output of logs is written to file logs.txt instead of being printed to stderr.
  • -help, -h - Show help.
  • -version - Show version info.

Testing Considerations

Before starting any test with srt-live-transmit please make sure your video source works properly. For example: if you use VLC as a test player, send a UDP stream directly to it before routing it through srt-live-transmit.

For any MPEG-TS UDP based source make sure it has packet sizes of 1316 bytes. When using ffmpeg like in the "Example for Smoke Testing" section above set the pkt_size=1316 parameter in case your input is a continuous data stream like from a file, camera or data-generator.

When leaving the LAN for testing, please keep an eye on statistics and make sure your round-trip-time (RTT) is not drifting. It's recommended to set the latency 3 to 4 times higher than RTT. Especially on wireless links such as WLAN, Line-of-Sight Radio (LOS) and mobile links such as LTE/4G or 5G the RTT can vary a lot.

If you perform tests on the public Internet, consider checking your firewall rules. The SRT listener must be reachable on the chosen UDP port. Same applies to routers using NAT. Please set a port forwarding rule with protocol UDP to the local IP address of the SRT listener.

The direction of initiating the stream doesn't need to be the same as the stream direction. The sender of a stream can be a SRT listener or SRT caller as long as the receiving end uses the opposite mode. Typically you use the SRT listener on the end, which is easier to configure in terms of firewall/router setup. It also makes sense to leave the Sender in listener mode when trying to connect from various end points with possibly unknown IP addresses.