FFMPEG-PROTOCOLS(1) - man page online | user commands
NAMEffmpeg-protocols - FFmpeg protocols
DESCRIPTIONThis document describes the input and output protocols provided by the libavformat library.
PROTOCOLSProtocols are configured elements in FFmpeg that enable access to resources that require specific protocols. When you configure your FFmpeg build, all the supported protocols are enabled by default. You can list all available ones using the configure option "--list-protocols". You can disable all the protocols using the configure option "--disable-protocols", and selectively enable a protocol using the option "--enable-protocol=PROTOCOL", or you can disable a particular protocol using the option "--disable-protocol=PROTOCOL". The option "-protocols" of the ff* tools will display the list of supported protocols. A description of the currently available protocols follows. async Asynchronous data filling wrapper for input stream. Fill data in a background thread, to decouple I/O operation from demux thread. async:<URL> async:http://host/resource async:cache:http://host/resource bluray Read BluRay playlist. The accepted options are: angle BluRay angle chapter Start chapter (1...N) playlist Playlist to read (BDMV/PLAYLIST/?????.mpls) Examples: Read longest playlist from BluRay mounted to /mnt/bluray: bluray:/mnt/bluray Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2: -playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray cache Caching wrapper for input stream. Cache the input stream to temporary file. It brings seeking capability to live streams. cache:<URL> concat Physical concatenation protocol. Read and seek from many resources in sequence as if they were a unique resource. A URL accepted by this protocol has the syntax: concat:<URL1>|<URL2>|...|<URLN> where URL1, URL2, ..., URLN are the urls of the resource to be concatenated, each one possibly specifying a distinct protocol. For example to read a sequence of files split1.mpeg, split2.mpeg, split3.mpeg with ffplay use the command: ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg Note that you may need to escape the character "|" which is special for many shells. crypto AES-encrypted stream reading protocol. The accepted options are: key Set the AES decryption key binary block from given hexadecimal representation. iv Set the AES decryption initialization vector binary block from given hexadecimal representation. Accepted URL formats: crypto:<URL> crypto+<URL> data Data in-line in the URI. See <http://en.wikipedia.org/wiki/Data_URI_scheme>. For example, to convert a GIF file given inline with ffmpeg: ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywA ↲ AAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png file File access protocol. Read from or write to a file. A file URL can have the form: file:<filename> where filename is the path of the file to read. An URL that does not have a protocol prefix will be assumed to be a file URL. Depending on the build, an URL that looks like a Windows path with the drive letter at the beginning will also be assumed to be a file URL (usually not the case in builds for unix-like systems). For example to read from a file input.mpeg with ffmpeg use the command: ffmpeg -i file:input.mpeg output.mpeg This protocol accepts the following options: truncate Truncate existing files on write, if set to 1. A value of 0 prevents truncating. Default value is 1. blocksize Set I/O operation maximum block size, in bytes. Default value is "INT_MAX", which results in not limiting the requested block size. Setting this value reasonably low improves user termination request reaction time, which is valuable for files on slow medium. ftp FTP (File Transfer Protocol). Read from or write to remote resources using FTP protocol. Following syntax is required. ftp://[user[:password]@]server[:port]/path/to/remote/resource.mpeg This protocol accepts the following options. timeout Set timeout in microseconds of socket I/O operations used by the underlying low level operation. By default it is set to -1, which means that the timeout is not specified. ftp-anonymous-password Password used when login as anonymous user. Typically an e-mail address should be used. ftp-write-seekable Control seekability of connection during encoding. If set to 1 the resource is supposed to be seekable, if set to 0 it is assumed not to be seekable. Default value is 0. NOTE: Protocol can be used as output, but it is recommended to not do it, unless special care is taken (tests, customized server configuration etc.). Different FTP servers behave in different way during seek operation. ff* tools may produce incomplete content due to server limitations. gopher Gopher protocol. hls Read Apple HTTP Live Streaming compliant segmented stream as a uniform one. The M3U8 playlists describing the segments can be remote HTTP resources or local files, accessed using the standard file protocol. The nested protocol is declared by specifying "+proto" after the hls URI scheme name, where proto is either "file" or "http". hls+http://host/path/to/remote/resource.m3u8 hls+file://path/to/local/resource.m3u8 Using this protocol is discouraged - the hls demuxer should work just as well (if not, please report the issues) and is more complete. To use the hls demuxer instead, simply use the direct URLs to the m3u8 files. http HTTP (Hyper Text Transfer Protocol). This protocol accepts the following options: seekable Control seekability of connection. If set to 1 the resource is supposed to be seekable, if set to 0 it is assumed not to be seekable, if set to -1 it will try to autodetect if it is seekable. Default value is -1. chunked_post If set to 1 use chunked Transfer-Encoding for posts, default is 1. content_type Set a specific content type for the POST messages. headers Set custom HTTP headers, can override built in default headers. The value must be a string encoding the headers. multiple_requests Use persistent connections if set to 1, default is 0. post_data Set custom HTTP post data. user-agent user_agent Override the User-Agent header. If not specified the protocol will use a string describing the libavformat build. ("Lavf/<version>") timeout Set timeout in microseconds of socket I/O operations used by the underlying low level operation. By default it is set to -1, which means that the timeout is not specified. mime_type Export the MIME type. icy If set to 1 request ICY (SHOUTcast) metadata from the server. If the server supports this, the metadata has to be retrieved by the application by reading the icy_metadata_headers and icy_metadata_packet options. The default is 1. icy_metadata_headers If the server supports ICY metadata, this contains the ICY-specific HTTP reply headers, separated by newline characters. icy_metadata_packet If the server supports ICY metadata, and icy was set to 1, this contains the last non- empty metadata packet sent by the server. It should be polled in regular intervals by applications interested in mid-stream metadata updates. cookies Set the cookies to be sent in future requests. The format of each cookie is the same as the value of a Set-Cookie HTTP response field. Multiple cookies can be delimited by a newline character. offset Set initial byte offset. end_offset Try to limit the request to bytes preceding this offset. method When used as a client option it sets the HTTP method for the request. When used as a server option it sets the HTTP method that is going to be expected from the client(s). If the expected and the received HTTP method do not match the client will be given a Bad Request response. When unset the HTTP method is not checked for now. This will be replaced by autodetection in the future. listen If set to 1 enables experimental HTTP server. This can be used to send data when used as an output option, or read data from a client with HTTP POST when used as an input option. If set to 2 enables experimental mutli-client HTTP server. This is not yet implemented in ffmpeg.c or ffserver.c and thus must not be used as a command line option. # Server side (sending): ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://<server>:<port> # Client side (receiving): ffmpeg -i http://<server>:<port> -c copy somefile.ogg # Client can also be done with wget: wget http://<server>:<port> -O somefile.ogg # Server side (receiving): ffmpeg -listen 1 -i http://<server>:<port> -c copy somefile.ogg # Client side (sending): ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://<server>:<port> # Client can also be done with wget: wget --post-file=somefile.ogg http://<server>:<port> HTTP Cookies Some HTTP requests will be denied unless cookie values are passed in with the request. The cookies option allows these cookies to be specified. At the very least, each cookie must specify a value along with a path and domain. HTTP requests that match both the domain and path will automatically include the cookie value in the HTTP Cookie header field. Multiple cookies can be delimited by a newline. The required syntax to play a stream specifying a cookie is: ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain ↲ .com/somestream.m3u8 Icecast Icecast protocol (stream to Icecast servers) This protocol accepts the following options: ice_genre Set the stream genre. ice_name Set the stream name. ice_description Set the stream description. ice_url Set the stream website URL. ice_public Set if the stream should be public. The default is 0 (not public). user_agent Override the User-Agent header. If not specified a string of the form "Lavf/<version>" will be used. password Set the Icecast mountpoint password. content_type Set the stream content type. This must be set if it is different from audio/mpeg. legacy_icecast This enables support for Icecast versions < 2.4.0, that do not support the HTTP PUT method but the SOURCE method. icecast://[<username>[:<password>]@]<server>:<port>/<mountpoint> mmst MMS (Microsoft Media Server) protocol over TCP. mmsh MMS (Microsoft Media Server) protocol over HTTP. The required syntax is: mmsh://<server>[:<port>][/<app>][/<playpath>] md5 MD5 output protocol. Computes the MD5 hash of the data to be written, and on close writes this to the designated output or stdout if none is specified. It can be used to test muxers without writing an actual file. Some examples follow. # Write the MD5 hash of the encoded AVI file to the file output.avi.md5. ffmpeg -i input.flv -f avi -y md5:output.avi.md5 # Write the MD5 hash of the encoded AVI file to stdout. ffmpeg -i input.flv -f avi -y md5: Note that some formats (typically MOV) require the output protocol to be seekable, so they will fail with the MD5 output protocol. pipe UNIX pipe access protocol. Read and write from UNIX pipes. The accepted syntax is: pipe:[<number>] number is the number corresponding to the file descriptor of the pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If number is not specified, by default the stdout file descriptor will be used for writing, stdin for reading. For example to read from stdin with ffmpeg: cat test.wav | ffmpeg -i pipe:0 # ...this is the same as... cat test.wav | ffmpeg -i pipe: For writing to stdout with ffmpeg: ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi # ...this is the same as... ffmpeg -i test.wav -f avi pipe: | cat > test.avi This protocol accepts the following options: blocksize Set I/O operation maximum block size, in bytes. Default value is "INT_MAX", which results in not limiting the requested block size. Setting this value reasonably low improves user termination request reaction time, which is valuable if data transmission is slow. Note that some formats (typically MOV), require the output protocol to be seekable, so they will fail with the pipe output protocol. rtmp Real-Time Messaging Protocol. The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia content across a TCP/IP network. The required syntax is: rtmp://[<username>:<password>@]<server>[:<port>][/<app>][/<instance>][/<playpath>] The accepted parameters are: username An optional username (mostly for publishing). password An optional password (mostly for publishing). server The address of the RTMP server. port The number of the TCP port to use (by default is 1935). app It is the name of the application to access. It usually corresponds to the path where the application is installed on the RTMP server (e.g. /ondemand/, /flash/live/, etc.). You can override the value parsed from the URI through the "rtmp_app" option, too. playpath It is the path or name of the resource to play with reference to the application specified in app, may be prefixed by "mp4:". You can override the value parsed from the URI through the "rtmp_playpath" option, too. listen Act as a server, listening for an incoming connection. timeout Maximum time to wait for the incoming connection. Implies listen. Additionally, the following parameters can be set via command line options (or in code via "AVOption"s): rtmp_app Name of application to connect on the RTMP server. This option overrides the parameter specified in the URI. rtmp_buffer Set the client buffer time in milliseconds. The default is 3000. rtmp_conn Extra arbitrary AMF connection parameters, parsed from a string, e.g. like "B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0". Each value is prefixed by a single character denoting the type, B for Boolean, N for number, S for string, O for object, or Z for null, followed by a colon. For Booleans the data must be either 0 or 1 for FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or 1 to end or begin an object, respectively. Data items in subobjects may be named, by prefixing the type with 'N' and specifying the name before the value (i.e. "NB:myFlag:1"). This option may be used multiple times to construct arbitrary AMF sequences. rtmp_flashver Version of the Flash plugin used to run the SWF player. The default is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible; <libavformat version>).) rtmp_flush_interval Number of packets flushed in the same request (RTMPT only). The default is 10. rtmp_live Specify that the media is a live stream. No resuming or seeking in live streams is possible. The default value is "any", which means the subscriber first tries to play the live stream specified in the playpath. If a live stream of that name is not found, it plays the recorded stream. The other possible values are "live" and "recorded". rtmp_pageurl URL of the web page in which the media was embedded. By default no value will be sent. rtmp_playpath Stream identifier to play or to publish. This option overrides the parameter specified in the URI. rtmp_subscribe Name of live stream to subscribe to. By default no value will be sent. It is only sent if the option is specified or if rtmp_live is set to live. rtmp_swfhash SHA256 hash of the decompressed SWF file (32 bytes). rtmp_swfsize Size of the decompressed SWF file, required for SWFVerification. rtmp_swfurl URL of the SWF player for the media. By default no value will be sent. rtmp_swfverify URL to player swf file, compute hash/size automatically. rtmp_tcurl URL of the target stream. Defaults to proto://host[:port]/app. For example to read with ffplay a multimedia resource named "sample" from the application "vod" from an RTMP server "myserver": ffplay rtmp://myserver/vod/sample To publish to a password protected server, passing the playpath and app names separately: ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rt ↲ mp://username:password@myserver/ rtmpe Encrypted Real-Time Messaging Protocol. The Encrypted Real-Time Messaging Protocol (RTMPE) is used for streaming multimedia content within standard cryptographic primitives, consisting of Diffie-Hellman key exchange and HMACSHA256, generating a pair of RC4 keys. rtmps Real-Time Messaging Protocol over a secure SSL connection. The Real-Time Messaging Protocol (RTMPS) is used for streaming multimedia content across an encrypted connection. rtmpt Real-Time Messaging Protocol tunneled through HTTP. The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used for streaming multimedia content within HTTP requests to traverse firewalls. rtmpte Encrypted Real-Time Messaging Protocol tunneled through HTTP. The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE) is used for streaming multimedia content within HTTP requests to traverse firewalls. rtmpts Real-Time Messaging Protocol tunneled through HTTPS. The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used for streaming multimedia content within HTTPS requests to traverse firewalls. libsmbclient libsmbclient permits one to manipulate CIFS/SMB network resources. Following syntax is required. smb://[[domain:]user[:password@]]server[/share[/path[/file]]] This protocol accepts the following options. timeout Set timeout in miliseconds of socket I/O operations used by the underlying low level operation. By default it is set to -1, which means that the timeout is not specified. truncate Truncate existing files on write, if set to 1. A value of 0 prevents truncating. Default value is 1. workgroup Set the workgroup used for making connections. By default workgroup is not specified. For more information see: <http://www.samba.org/>. libssh Secure File Transfer Protocol via libssh Read from or write to remote resources using SFTP protocol. Following syntax is required. sftp://[user[:password]@]server[:port]/path/to/remote/resource.mpeg This protocol accepts the following options. timeout Set timeout of socket I/O operations used by the underlying low level operation. By default it is set to -1, which means that the timeout is not specified. truncate Truncate existing files on write, if set to 1. A value of 0 prevents truncating. Default value is 1. private_key Specify the path of the file containing private key to use during authorization. By default libssh searches for keys in the ~/.ssh/ directory. Example: Play a file stored on remote server. ffplay sftp://user:password@server_address:22/home/user/resource.mpeg librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte Real-Time Messaging Protocol and its variants supported through librtmp. Requires the presence of the librtmp headers and library during configuration. You need to explicitly configure the build with "--enable-librtmp". If enabled this will replace the native RTMP protocol. This protocol provides most client functions and a few server functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT), encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled variants of these encrypted types (RTMPTE, RTMPTS). The required syntax is: <rtmp_proto>://<server>[:<port>][/<app>][/<playpath>] <options> where rtmp_proto is one of the strings "rtmp", "rtmpt", "rtmpe", "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and server, port, app and playpath have the same meaning as specified for the RTMP native protocol. options contains a list of space- separated options of the form key=val. See the librtmp manual page (man 3 librtmp) for more information. For example, to stream a file in real-time to an RTMP server using ffmpeg: ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream To play the same stream using ffplay: ffplay "rtmp://myserver/live/mystream live=1" rtp Real-time Transport Protocol. The required syntax for an RTP URL is: rtp://hostname[:port][?option=val...] port specifies the RTP port to use. The following URL options are supported: ttl=n Set the TTL (Time-To-Live) value (for multicast only). rtcpport=n Set the remote RTCP port to n. localrtpport=n Set the local RTP port to n. localrtcpport=n' Set the local RTCP port to n. pkt_size=n Set max packet size (in bytes) to n. connect=0|1 Do a "connect()" on the UDP socket (if set to 1) or not (if set to 0). sources=ip[,ip] List allowed source IP addresses. block=ip[,ip] List disallowed (blocked) source IP addresses. write_to_source=0|1 Send packets to the source address of the latest received packet (if set to 1) or to a default remote address (if set to 0). localport=n Set the local RTP port to n. This is a deprecated option. Instead, localrtpport should be used. Important notes: 1. If rtcpport is not set the RTCP port will be set to the RTP port value plus 1. 2. If localrtpport (the local RTP port) is not set any available port will be used for the local RTP and RTCP ports. 3. If localrtcpport (the local RTCP port) is not set it will be set to the local RTP port value plus 1. rtsp Real-Time Streaming Protocol. RTSP is not technically a protocol handler in libavformat, it is a demuxer and muxer. The demuxer supports both normal RTSP (with data transferred over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with data transferred over RDT). The muxer can be used to send a stream using RTSP ANNOUNCE to a server supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's <https://github.com/revmischa/rtsp-server>). The required syntax for a RTSP url is: rtsp://<hostname>[:<port>]/<path> Options can be set on the ffmpeg/ffplay command line, or set in code via "AVOption"s or in "avformat_open_input". The following options are supported. initial_pause Do not start playing the stream immediately if set to 1. Default value is 0. rtsp_transport Set RTSP transport protocols. It accepts the following values: udp Use UDP as lower transport protocol. tcp Use TCP (interleaving within the RTSP control channel) as lower transport protocol. udp_multicast Use UDP multicast as lower transport protocol. http Use HTTP tunneling as lower transport protocol, which is useful for passing proxies. Multiple lower transport protocols may be specified, in that case they are tried one at a time (if the setup of one fails, the next one is tried). For the muxer, only the tcp and udp options are supported. rtsp_flags Set RTSP flags. The following values are accepted: filter_src Accept packets only from negotiated peer address and port. listen Act as a server, listening for an incoming connection. prefer_tcp Try TCP for RTP transport first, if TCP is available as RTSP RTP transport. Default value is none. allowed_media_types Set media types to accept from the server. The following flags are accepted: video audio data By default it accepts all media types. min_port Set minimum local UDP port. Default value is 5000. max_port Set maximum local UDP port. Default value is 65000. timeout Set maximum timeout (in seconds) to wait for incoming connections. A value of -1 means infinite (default). This option implies the rtsp_flags set to listen. reorder_queue_size Set number of packets to buffer for handling of reordered packets. stimeout Set socket TCP I/O timeout in microseconds. user-agent Override User-Agent header. If not specified, it defaults to the libavformat identifier string. When receiving data over UDP, the demuxer tries to reorder received packets (since they may arrive out of order, or packets may get lost totally). This can be disabled by setting the maximum demuxing delay to zero (via the "max_delay" field of AVFormatContext). When watching multi-bitrate Real-RTSP streams with ffplay, the streams to display can be chosen with "-vst" n and "-ast" n for video and audio respectively, and can be switched on the fly by pressing "v" and "a". Examples The following examples all make use of the ffplay and ffmpeg tools. · Watch a stream over UDP, with a max reordering delay of 0.5 seconds: ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4 · Watch a stream tunneled over HTTP: ffplay -rtsp_transport http rtsp://server/video.mp4 · Send a stream in realtime to a RTSP server, for others to watch: ffmpeg -re -i <input> -f rtsp -muxdelay 0.1 rtsp://server/live.sdp · Receive a stream in realtime: ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp <output> sap Session Announcement Protocol (RFC 2974). This is not technically a protocol handler in libavformat, it is a muxer and demuxer. It is used for signalling of RTP streams, by announcing the SDP for the streams regularly on a separate port. Muxer The syntax for a SAP url given to the muxer is: sap://<destination>[:<port>][?<options>] The RTP packets are sent to destination on port port, or to port 5004 if no port is specified. options is a "&"-separated list. The following options are supported: announce_addr=address Specify the destination IP address for sending the announcements to. If omitted, the announcements are sent to the commonly used SAP announcement multicast address 18.104.22.168 (sap.mcast.net), or ff0e::2:7ffe if destination is an IPv6 address. announce_port=port Specify the port to send the announcements on, defaults to 9875 if not specified. ttl=ttl Specify the time to live value for the announcements and RTP packets, defaults to 255. same_port=0|1 If set to 1, send all RTP streams on the same port pair. If zero (the default), all streams are sent on unique ports, with each stream on a port 2 numbers higher than the previous. VLC/Live555 requires this to be set to 1, to be able to receive the stream. The RTP stack in libavformat for receiving requires all streams to be sent on unique ports. Example command lines follow. To broadcast a stream on the local subnet, for watching in VLC: ffmpeg -re -i <input> -f sap sap://22.214.171.124?same_port=1 Similarly, for watching in ffplay: ffmpeg -re -i <input> -f sap sap://126.96.36.199 And for watching in ffplay, over IPv6: ffmpeg -re -i <input> -f sap sap://[ff0e::1:2:3:4] Demuxer The syntax for a SAP url given to the demuxer is: sap://[<address>][:<port>] address is the multicast address to listen for announcements on, if omitted, the default 188.8.131.52 (sap.mcast.net) is used. port is the port that is listened on, 9875 if omitted. The demuxers listens for announcements on the given address and port. Once an announcement is received, it tries to receive that particular stream. Example command lines follow. To play back the first stream announced on the normal SAP multicast address: ffplay sap:// To play back the first stream announced on one the default IPv6 SAP multicast address: ffplay sap://[ff0e::2:7ffe] sctp Stream Control Transmission Protocol. The accepted URL syntax is: sctp://<host>:<port>[?<options>] The protocol accepts the following options: listen If set to any value, listen for an incoming connection. Outgoing connection is done by default. max_streams Set the maximum number of streams. By default no limit is set. srtp Secure Real-time Transport Protocol. The accepted options are: srtp_in_suite srtp_out_suite Select input and output encoding suites. Supported values: AES_CM_128_HMAC_SHA1_80 SRTP_AES128_CM_HMAC_SHA1_80 AES_CM_128_HMAC_SHA1_32 SRTP_AES128_CM_HMAC_SHA1_32 srtp_in_params srtp_out_params Set input and output encoding parameters, which are expressed by a base64-encoded representation of a binary block. The first 16 bytes of this binary block are used as master key, the following 14 bytes are used as master salt. subfile Virtually extract a segment of a file or another stream. The underlying stream must be seekable. Accepted options: start Start offset of the extracted segment, in bytes. end End offset of the extracted segment, in bytes. Examples: Extract a chapter from a DVD VOB file (start and end sectors obtained externally and multiplied by 2048): subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB Play an AVI file directly from a TAR archive: subfile,,start,183241728,end,366490624,,:archive.tar tcp Transmission Control Protocol. The required syntax for a TCP url is: tcp://<hostname>:<port>[?<options>] options contains a list of &-separated options of the form key=val. The list of supported options follows. listen=1|0 Listen for an incoming connection. Default value is 0. timeout=microseconds Set raise error timeout, expressed in microseconds. This option is only relevant in read mode: if no data arrived in more than this time interval, raise error. listen_timeout=milliseconds Set listen timeout, expressed in milliseconds. The following example shows how to setup a listening TCP connection with ffmpeg, which is then accessed with ffplay: ffmpeg -i <input> -f <format> tcp://<hostname>:<port>?listen ffplay tcp://<hostname>:<port> tls Transport Layer Security (TLS) / Secure Sockets Layer (SSL) The required syntax for a TLS/SSL url is: tls://<hostname>:<port>[?<options>] The following parameters can be set via command line options (or in code via "AVOption"s): ca_file, cafile=filename A file containing certificate authority (CA) root certificates to treat as trusted. If the linked TLS library contains a default this might not need to be specified for verification to work, but not all libraries and setups have defaults built in. The file must be in OpenSSL PEM format. tls_verify=1|0 If enabled, try to verify the peer that we are communicating with. Note, if using OpenSSL, this currently only makes sure that the peer certificate is signed by one of the root certificates in the CA database, but it does not validate that the certificate actually matches the host name we are trying to connect to. (With GnuTLS, the host name is validated as well.) This is disabled by default since it requires a CA database to be provided by the caller in many cases. cert_file, cert=filename A file containing a certificate to use in the handshake with the peer. (When operating as server, in listen mode, this is more often required by the peer, while client certificates only are mandated in certain setups.) key_file, key=filename A file containing the private key for the certificate. listen=1|0 If enabled, listen for connections on the provided port, and assume the server role in the handshake instead of the client role. Example command lines: To create a TLS/SSL server that serves an input stream. ffmpeg -i <input> -f <format> tls://<hostname>:<port>?listen&cert=<server.crt>&key=<s ↲ erver.key> To play back a stream from the TLS/SSL server using ffplay: ffplay tls://<hostname>:<port> udp User Datagram Protocol. The required syntax for an UDP URL is: udp://<hostname>:<port>[?<options>] options contains a list of &-separated options of the form key=val. In case threading is enabled on the system, a circular buffer is used to store the incoming data, which allows one to reduce loss of data due to UDP socket buffer overruns. The fifo_size and overrun_nonfatal options are related to this buffer. The list of supported options follows. buffer_size=size Set the UDP maximum socket buffer size in bytes. This is used to set either the receive or send buffer size, depending on what the socket is used for. Default is 64KB. See also fifo_size. localport=port Override the local UDP port to bind with. localaddr=addr Choose the local IP address. This is useful e.g. if sending multicast and the host has multiple interfaces, where the user can choose which interface to send on by specifying the IP address of that interface. pkt_size=size Set the size in bytes of UDP packets. reuse=1|0 Explicitly allow or disallow reusing UDP sockets. ttl=ttl Set the time to live value (for multicast only). connect=1|0 Initialize the UDP socket with "connect()". In this case, the destination address can't be changed with ff_udp_set_remote_url later. If the destination address isn't known at the start, this option can be specified in ff_udp_set_remote_url, too. This allows finding out the source address for the packets with getsockname, and makes writes return with AVERROR(ECONNREFUSED) if "destination unreachable" is received. For receiving, this gives the benefit of only receiving packets from the specified peer address/port. sources=address[,address] Only receive packets sent to the multicast group from one of the specified sender IP addresses. block=address[,address] Ignore packets sent to the multicast group from the specified sender IP addresses. fifo_size=units Set the UDP receiving circular buffer size, expressed as a number of packets with size of 188 bytes. If not specified defaults to 7*4096. overrun_nonfatal=1|0 Survive in case of UDP receiving circular buffer overrun. Default value is 0. timeout=microseconds Set raise error timeout, expressed in microseconds. This option is only relevant in read mode: if no data arrived in more than this time interval, raise error. broadcast=1|0 Explicitly allow or disallow UDP broadcasting. Note that broadcasting may not work properly on networks having a broadcast storm protection. Examples · Use ffmpeg to stream over UDP to a remote endpoint: ffmpeg -i <input> -f <format> udp://<hostname>:<port> · Use ffmpeg to stream in mpegts format over UDP using 188 sized UDP packets, using a large input buffer: ffmpeg -i <input> -f mpegts udp://<hostname>:<port>?pkt_size=188&buffer_size=6553 ↲ 5 · Use ffmpeg to receive over UDP from a remote endpoint: ffmpeg -i udp://[<multicast-address>]:<port> ... unix Unix local socket The required syntax for a Unix socket URL is: unix://<filepath> The following parameters can be set via command line options (or in code via "AVOption"s): timeout Timeout in ms. listen Create the Unix socket in listening mode.
SEE ALSOffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavformat(3)
AUTHORSThe FFmpeg developers. For details about the authorship, see the Git history of the project (git://source.ffmpeg.org/ffmpeg), e.g. by typing the command git log in the FFmpeg source directory, or browsing the online repository at <http://source.ffmpeg.org>. Maintainers for the specific components are listed in the file MAINTAINERS in the source code tree.
|This manual||Reference||Other manuals|
|ffmpeg-protocols(1)||referred by||ffmpeg(1) | ffmpeg-all(1) | ffplay(1) | ffplay-all(1) | ffprobe(1) | ffprobe-all(1) | ffserver(1) | ffserver-all(1) | libavformat(3)|
|refer to||ffmpeg(1) | ffplay(1) | ffprobe(1) | ffserver(1) | libavformat(3)|