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FFMPEG-PROTOCOLS(1) FFMPEG-PROTOCOLS(1) NAME ffmpeg-protocols - FFmpeg protocols DESCRIPTION This document describes the input and output protocols provided by the libavformat library. PROTOCOL OPTIONS The libavformat library provides some generic global options, which can be set on all the protocols. In addition each protocol may support so-called private options, which are specific for that component. Options may be set by specifying -option value in the FFmpeg tools, or by setting the value explicitly in the "AVFormatContext" options or using the libavutil/opt.h API for programmatic use. The list of supported options follows: protocol_whitelist list (input) Set a ","-separated list of allowed protocols. "ALL" matches all proto- cols. Protocols prefixed by "-" are disabled. All protocols are al- lowed by default but protocols used by an another protocol (nested pro- tocols) are restricted to a per protocol subset. PROTOCOLS Protocols 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 en- abled by default. You can list all available ones using the configure op- tion "--list-protocols". You can disable all the protocols using the configure option "--dis- able-protocols", and selectively enable a protocol using the option "--en- able-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. All protocols accept the following options: rw_timeout Maximum time to wait for (network) read/write operations to complete, in microseconds. A description of the currently available protocols follows. amqp Advanced Message Queueing Protocol (AMQP) version 0-9-1 is a broker based publish-subscribe communication protocol. FFmpeg must be compiled with --enable-librabbitmq to support AMQP. A sepa- rate AMQP broker must also be run. An example open-source AMQP broker is RabbitMQ. After starting the broker, an FFmpeg client may stream data to the broker using the command: ffmpeg -re -i input -f mpegts amqp://[[user]:[password]@]hostname[:port][/vhost] Where hostname and port (default is 5672) is the address of the broker. The client may also set a user/password for authentication. The default for both fields is "guest". Name of virtual host on broker can be set with vhost. The default value is "/". Multiple subscribers may stream from the broker using the command: ffplay amqp://[[user]:[password]@]hostname[:port][/vhost] In RabbitMQ all data published to the broker flows through a specific ex- change, and each subscribing client has an assigned queue/buffer. When a packet arrives at an exchange, it may be copied to a client's queue depend- ing on the exchange and routing_key fields. The following options are supported: exchange Sets the exchange to use on the broker. RabbitMQ has several predefined exchanges: "amq.direct" is the default exchange, where the publisher and subscriber must have a matching routing_key; "amq.fanout" is the same as a broadcast operation (i.e. the data is forwarded to all queues on the fanout exchange independent of the routing_key); and "amq.topic" is similar to "amq.direct", but allows for more complex pattern match- ing (refer to the RabbitMQ documentation). routing_key Sets the routing key. The default value is "amqp". The routing key is used on the "amq.direct" and "amq.topic" exchanges to decide whether packets are written to the queue of a subscriber. pkt_size Maximum size of each packet sent/received to the broker. Default is 131072. Minimum is 4096 and max is any large value (representable by an int). When receiving packets, this sets an internal buffer size in FFmpeg. It should be equal to or greater than the size of the published packets to the broker. Otherwise the received message may be truncated causing decoding errors. connection_timeout The timeout in seconds during the initial connection to the broker. The default value is rw_timeout, or 5 seconds if rw_timeout is not set. delivery_mode mode Sets the delivery mode of each message sent to broker. The following values are accepted: persistent Delivery mode set to "persistent" (2). This is the default value. Messages may be written to the broker's disk depending on its setup. non-persistent Delivery mode set to "non-persistent" (1). Messages will stay in broker's memory unless the broker is under memory pressure. 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. The accepted options are: read_ahead_limit Amount in bytes that may be read ahead when seeking isn't supported. Range is -1 to INT_MAX. -1 for unlimited. Default is 65536. URL Syntax is cache:<URL> concat Physical concatenation protocol. Read and seek from many resources in sequence as if they were a unique re- source. 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 concate- nated, 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. concatf Physical concatenation protocol using a line break delimited list of re- sources. Read and seek from many resources in sequence as if they were a unique re- source. A URL accepted by this protocol has the syntax: concatf:<URL> where URL is the url containing a line break delimited list of resources to be concatenated, each one possibly specifying a distinct protocol. Special characters must be escaped with backslash or single quotes. See the "Quot- ing and escaping" section in the ffmpeg-utils(1) manual. For example to read a sequence of files split1.mpeg, split2.mpeg, split3.mpeg listed in separate lines within a file split.txt with ffplay use the command: ffplay concatf:split.txt Where split.txt contains the lines: split1.mpeg split2.mpeg split3.mpeg crypto AES-encrypted stream reading protocol. The accepted options are: key Set the AES decryption key binary block from given hexadecimal repre- sentation. 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///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png fd File descriptor access protocol. The accepted syntax is: fd: -fd <file_descriptor> If fd is not specified, by default the stdout file descriptor will be used for writing, stdin for reading. Unlike the pipe protocol, fd protocol has seek support if it corresponding to a regular file. fd protocol doesn't support pass file descriptor via URL for security. 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 re- action time, which is valuable if data transmission is slow. fd Set file descriptor. 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 (usu- ally 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 re- action time, which is valuable for files on slow medium. follow If set to 1, the protocol will retry reading at the end of the file, allowing reading files that still are being written. In order for this to terminate, you either need to use the rw_timeout option, or use the interrupt callback (for API users). seekable Controls if seekability is advertised on the file. 0 means non-seek- able, -1 means auto (seekable for normal files, non-seekable for named pipes). Many demuxers handle seekable and non-seekable resources differently, overriding this might speed up opening certain files at the cost of losing some features (e.g. accurate seeking). pkt_size Set the maximum packet size used for file I/O. A smaller value may re- duce memory usage. A higher value may increase throughput especially with networked filesystems. For reading, if explicitly set, it overrides the default internal buffer size (32 KB) and limits the maximum amount of data read per op- eration. For writing, this sets the size of each write operation. The default is 256 KB for regular files, 32 KB otherwise. 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 under- lying low level operation. By default it is set to -1, which means that the timeout is not specified. ftp-user Set a user to be used for authenticating to the FTP server. This is overridden by the user in the FTP URL. ftp-password Set a password to be used for authenticating to the FTP server. This is overridden by the password in the FTP URL, or by ftp-anonymous-password if no user is set. 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 re- source 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. gophers Gophers protocol. The Gopher protocol with TLS encapsulation. 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. http_proxy set HTTP proxy to tunnel through e.g. http://example.com:1234 headers Set custom HTTP headers, can override built in default headers. The value must be a string encoding the headers. content_type Set a specific content type for the POST messages or for listen mode. user_agent Override the User-Agent header. If not specified the protocol will use a string describing the libavformat build. ("Lavf/<version>") referer Set the Referer header. Include 'Referer: URL' header in HTTP request. multiple_requests Use persistent connections if set to 1, default is 0. request_size Limit the size of requests made. This is useful for some pathological servers that throttle unbounded range requests, as well as when expect- ing to seek frequently. Disabled (set to 0) by default. Note that if enabling this option, it's strongly recommended to also enable the multiple_requests option, as well as setting short_seek_size to the same value or higher. Doing so allows FFmpeg to reuse a single HTTP connection wherever possible. initial_request_size Limit the size of initial requests. Similar to "request_size", but only used during initial format parsing. Useful for formats like MXF or MOV that require frequent seeks during header parsing. Lasts until the de- muxer makes a read request larger than this size (without a seek in be- tween), after which the implementation will continue using requests as usual. Disabled (set to 0) by default. Note that if enabling this option, it's strongly recommended to also enable the multiple_requests option, as well as setting short_seek_size to the same value or higher. post_data Set custom HTTP post data. mime_type Export the MIME type. http_version Exports the HTTP response version number. Usually "1.0" or "1.1". 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. 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 applica- tion by reading the icy_metadata_headers and icy_metadata_packet op- tions. 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 con- tains 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. metadata Set an exported dictionary containing Icecast metadata from the bit- stream, if present. Only useful with the C API. auth_type Set HTTP authentication type. No option for Digest, since this method requires getting nonce parameters from the server first and can't be used straight away like Basic. noneChoose the HTTP authentication type automatically. This is the de- fault. basic Choose the HTTP basic authentication. Basic authentication sends a Base64-encoded string that contains a user name and password for the client. Base64 is not a form of en- cryption and should be considered the same as sending the user name and password in clear text (Base64 is a reversible encoding). If a resource needs to be protected, strongly consider using an authen- tication scheme other than basic authentication. HTTPS/TLS should be used with basic authentication. Without these additional secu- rity enhancements, basic authentication should not be used to pro- tect sensitive or valuable information. send_expect_100 Send an Expect: 100-continue header for POST. If set to 1 it will send, if set to 0 it won't, if set to -1 it will try to send if it is applic- able. Default value is -1. location An exported dictionary containing the content location. Only useful with the C API. 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 re- placed by autodetection in the future. reconnect Reconnect automatically when disconnected before EOF is hit. reconnect_at_eof If set then eof is treated like an error and causes reconnection, this is useful for live / endless streams. reconnect_on_network_error Reconnect automatically in case of TCP/TLS errors during connect. reconnect_on_http_error A comma separated list of HTTP status codes to reconnect on. The list can include specific status codes (e.g. '503') or the strings '4xx' / '5xx'. reconnect_streamed If set then even streamed/non seekable streams will be reconnected on errors. reconnect_delay_max Set the maximum delay in seconds after which to give up reconnecting. reconnect_max_retries Set the maximum number of times to retry a connection. Default unset. reconnect_delay_total_max Set the maximum total delay in seconds after which to give up recon- necting. respect_retry_after If enabled, and a Retry-After header is encountered, its requested re- connection delay will be honored, rather than using exponential back- off. Useful for 429 and 503 errors. Default enabled. 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 experimen- tal multi-client HTTP server. This is not yet implemented in ffmpeg.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> resource The resource requested by a client, when the experimental HTTP server is in use. reply_code The HTTP code returned to the client, when the experimental HTTP server is in use. short_seek_size Set the threshold, in bytes, for when a readahead should be preferred over a seek and new HTTP request. This is useful, for example, to make sure the same connection is used for reading large video packets with small audio packets in between. 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 do- main. 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. tls Establish a TLS (HTTPS) connection to Icecast. icecast://[<username>[:<password>]@]<server>:<port>/<mountpoint> ipfs InterPlanetary File System (IPFS) protocol support. One can access files stored on the IPFS network through so-called gateways. These are http(s) endpoints. This protocol wraps the IPFS native protocols (ipfs:// and ipns://) to be sent to such a gateway. Users can (and should) host their own node which means this protocol will use one's local gateway to access files on the IPFS network. This protocol accepts the following options: gateway Defines the gateway to use. When not set, the protocol will first try locating the local gateway by looking at $IPFS_GATEWAY, $IPFS_PATH and "$HOME/.ipfs/", in that order. One can use this protocol in 2 ways. Using IPFS: ffplay ipfs://<hash> Or the IPNS protocol (IPNS is mutable IPFS): ffplay ipns://<hash> 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>] If fd isn't specified, number is the number corresponding to the file de- scriptor 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 re- action time, which is valuable if data transmission is slow. fd Set file descriptor. Note that some formats (typically MOV), require the output protocol to be seekable, so they will fail with the pipe output protocol. prompeg Pro-MPEG Code of Practice #3 Release 2 FEC protocol. The Pro-MPEG CoP#3 FEC is a 2D parity-check forward error correction mecha- nism for MPEG-2 Transport Streams sent over RTP. This protocol must be used in conjunction with the "rtp_mpegts" muxer and the "rtp" protocol. The required syntax is: -f rtp_mpegts -fec prompeg=<option>=<val>... rtp://<hostname>:<port> The destination UDP ports are "port + 2" for the column FEC stream and "port + 4" for the row FEC stream. This protocol accepts the following options: l=n The number of columns (4-20, LxD <= 100) d=n The number of rows (4-20, LxD <= 100) Example usage: -f rtp_mpegts -fec prompeg=l=8:d=4 rtp://<hostname>:<port> rist Reliable Internet Streaming Transport protocol The accepted options are: rist_profile Supported values: simple mainThis one is default. advanced buffer_size Set internal RIST buffer size in milliseconds for retransmission of data. Default value is 0 which means the librist default (1 sec). Max- imum value is 30 seconds. fifo_size Size of the librist receiver output fifo in number of packets. This must be a power of 2. Defaults to 8192 (vs the librist default of 1024). overrun_nonfatal=1|0 Survive in case of librist fifo buffer overrun. Default value is 0. pkt_size Set maximum packet size for sending data. 1316 by default. log_level Set loglevel for RIST logging messages. You only need to set this if you explicitly want to enable debug level messages or packet loss simu- lation, otherwise the regular loglevel is respected. secret Set override of encryption secret, by default is unset. encryption Set encryption type, by default is disabled. Acceptable values are 128 and 256. 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. portThe 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 over- ride 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 over- rides 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, respec- tively. 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 pre- fixing 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 ar- bitrary AMF sequences. rtmp_enhanced_codecs Specify the list of codecs the client advertises to support in an en- hanced RTMP stream. This option should be set to a comma separated list of fourcc values, like "hvc1,av01,vp09" for multiple codecs or "hvc1" for only one codec. The specified list will be presented in the "four- CcLive" property of the Connect Command Message. 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 sub- scriber 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 pa- rameter 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. tcp_nodelay=1|0 Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0. Remark: Writing to the socket is currently not optimized to minimize system calls and reduces the efficiency / effect of TCP_NODELAY. tcp_keepalive=1|0 Enable the TCP keepalive mechanism to detect dead peers and help main- tain long-lived idle connections. Default value is 0. Only the basic keepalive option (SO_KEEPALIVE) can be enabled or dis- abled. Platform-specific tuning parameters such as TCP_KEEPIDLE, TCP_KEEPINTVL, or TCP_KEEPCNT are not configurable and will use the op- erating system's default values. 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 rtmp://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 fire- walls. 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 milliseconds of socket I/O operations used by the under- lying 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 au- thorization. By default libssh searches for keys in the ~/.ssh/ direc- tory. 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 configura- tion. 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>][?<options>] port specifies the RTP port to use. options contains a list of &-separated options of the form key=val. Stan- dard percent-encoding (and using the plus sign for space) can be used to escape keys and values. Options can also can be specified via command line options (or in code via "AVOption"s). The list of supported options follows. ttl=n Set the TTL (Time-To-Live) value (for multicast only). rtcpport=n Set the remote RTCP port to n. localport, local_rtpport, localrtpport=n Set the local RTP port to n. Using the localport option name is deprecated and should not be used. local_rtcpport, localrtcpport=n' Set the local RTCP port to n. pkt_size=n Set max packet size (in bytes) to n. buffer_size=size Set the maximum UDP socket buffer size in bytes. 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). localaddr=addr Local IP address of a network interface used for sending packets or joining multicast groups. timeout=n Set timeout (in microseconds) of socket I/O operations to n. 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 sup- porting 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". Muxer The following options are supported. 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. Default value is 0. rtsp_flags Set RTSP flags. The following values are accepted: latmUse MP4A-LATM packetization instead of MPEG4-GENERIC for AAC. rfc2190 Use RFC 2190 packetization instead of RFC 4629 for H.263. skip_rtcp Don't send RTCP sender reports. h264_mode0 Use mode 0 for H.264 in RTP. send_bye Send RTCP BYE packets when finishing. Default value is 0. min_port Set minimum local UDP port. Default value is 5000. max_port Set maximum local UDP port. Default value is 65000. buffer_size Set the maximum socket buffer size in bytes. pkt_size Set max send packet size (in bytes). Default value is 1472. Demuxer 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. httpUse HTTP tunneling as lower transport protocol, which is useful for passing proxies. https Use HTTPs tunneling as lower transport protocol, which is useful for passing proxies and widely used for security consideration. 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. satip_raw Export raw MPEG-TS stream instead of demuxing. The flag will simply write out the raw stream, with the original PAT/PMT/PIDs intact. Default value is none. allowed_media_types Set media types to accept from the server. The following flags are accepted: video audio data subtitle 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. listen_timeout Set maximum timeout (in seconds) to establish an initial connection. Setting listen_timeout > 0 sets rtsp_flags to listen. Default is -1 which means an infinite timeout when listen mode is set. reorder_queue_size Set number of packets to buffer for handling of reordered packets. timeout Set socket TCP I/O timeout in microseconds. user_agent Override User-Agent header. If not specified, it defaults to the libav- format identifier string. buffer_size Set the maximum socket buffer size in bytes. 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 re- spectively, 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 proto- col handler in libavformat, it is a muxer and demuxer. It is used for sig- nalling 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 an- nouncement multicast address 224.2.127.254 (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 libav- format 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://224.0.0.255?same_port=1 Similarly, for watching in ffplay: ffmpeg -re -i <input> -f sap sap://224.0.0.255 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 omit- ted, the default 224.2.127.254 (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 ad- dress: ffplay sap:// To play back the first stream announced on one the default IPv6 SAP multi- cast address: ffplay sap://[ff0e::2:7ffe] sctp Stream Control Transmission Protocol. The accepted URL syntax is: sctp://<host>:<port>[?<options>] options contains a list of &-separated options of the form key=val. Stan- dard percent-encoding (and using the plus sign for space) can be used to escape keys and values. Options can also can be specified via command line options (or in code via "AVOption"s). The list of supported options follows. listen If set to any value, listen for an incoming connection. Outgoing con- nection is done by default. max_streams Set the maximum number of streams. By default no limit is set. srt Haivision Secure Reliable Transport Protocol via libsrt. The supported syntax for a SRT URL is: srt://<hostname>:<port>[?<options>] options contains a list of &-separated options of the form key=val. Stan- dard percent-encoding (and using the plus sign for space) can be used to escape keys and values. Options can also can be specified via command line options (or in code via "AVOption"s). The list of supported options follows. connect_timeout=milliseconds Connection timeout; SRT cannot connect for RTT > 1500 msec (2 handshake exchanges) with the default connect timeout of 3 seconds. This option applies to the caller and rendezvous connection modes. The connect timeout is 10 times the value set for the rendezvous mode (which can be used as a workaround for this connection problem with earlier ver- sions). ffs=bytes Flight Flag Size (Window Size), in bytes. FFS is actually an internal parameter and you should set it to not less than recv_buffer_size and mss. The default value is relatively large, therefore unless you set a very large receiver buffer, you do not need to change this option. De- fault value is 25600. inputbw=bytes/seconds Sender nominal input rate, in bytes per seconds. Used along with oheadbw, when maxbw is set to relative (0), to calculate maximum send- ing rate when recovery packets are sent along with the main media stream: inputbw * (100 + oheadbw) / 100 if inputbw is not set while maxbw is set to relative (0), the actual input rate is evaluated inside the library. Default value is 0. iptos=tos IP Type of Service. Applies to sender only. Default value is 0xB8. ipttl=ttl IP Time To Live. Applies to sender only. Default value is 64. latency=microseconds Timestamp-based Packet Delivery Delay. Used to absorb bursts of missed packet retransmissions. This flag sets both rcvlatency and 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 set- ting peerlatency, when side is sender and rcvlatency when side is re- ceiver, and the bidirectional stream sending is not supported. listen_timeout=microseconds Set socket listen timeout. maxbw=bytes/seconds Maximum sending bandwidth, in bytes per seconds. -1 infinite (CSRTCC limit is 30mbps) 0 relative to input rate (see inputbw) >0 absolute limit value Default value is 0 (relative) mode=caller|listener|rendezvous Connection mode. caller opens client connection. listener starts server to listen for incoming connections. rendezvous use Rendez-Vous connection mode. Default value is caller. mss=bytes Maximum Segment Size, in bytes. Used for buffer allocation and rate calculation using a 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 set- tings. Default value is 1500. nakreport=1|0 If set to 1, Receiver will send `UMSG_LOSSREPORT` messages periodically until a lost packet is retransmitted or intentionally dropped. Default value is 1. oheadbw=percents Recovery bandwidth overhead above input rate, in percents. See in- putbw. Default value is 25%. passphrase=string HaiCrypt Encryption/Decryption Passphrase string, length from 10 to 79 characters. The passphrase is the shared secret between the sender and the receiver. It is used to generate the Key Encrypting Key using PBKDF2 (Password-Based Key Derivation Function). It is used only if pbkeylen is non-zero. It is used on the receiver only if the received data is encrypted. The configured passphrase cannot be recovered (write-only). enforced_encryption=1|0 If true, both connection parties must have the same password set (in- cluding empty, that is, with no encryption). If the password doesn't match or only one side is unencrypted, the connection is rejected. De- fault is true. kmrefreshrate=packets The number of packets to be transmitted after which the encryption key is switched to a new key. Default is -1. -1 means auto (0x1000000 in srt library). The range for this option is integers in the 0 - "INT_MAX". kmpreannounce=packets The interval between when a new encryption key is sent and when switchover occurs. This value also applies to the subsequent interval between when switchover occurs and when the old encryption key is de- commissioned. Default is -1. -1 means auto (0x1000 in srt library). The range for this option is integers in the 0 - "INT_MAX". snddropdelay=microseconds The sender's extra delay before dropping packets. This delay is added to the default drop delay time interval value. Special value -1: Do not drop packets on the sender at all. payload_size=bytes Sets the maximum declared size of a packet transferred during the sin- gle call to the sending function in Live mode. Use 0 if this value isn't used (which is default in file mode). Default is -1 (automatic), which typically means 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. pkt_size=bytes Alias for payload_size. peerlatency=microseconds The latency value (as described in rcvlatency) that is set by the sender side as a minimum value for the receiver. pbkeylen=bytes Sender encryption key length, in bytes. Only can be set to 0, 16, 24 and 32. Enable sender encryption if not 0. Not required on receiver (set to 0), key size obtained from sender in HaiCrypt handshake. De- fault value is 0. rcvlatency=microseconds The time that should elapse since the moment when the packet was sent and the moment when it's delivered to the receiver application in the receiving function. This time should be a buffer time large enough to cover the time spent for sending, unexpectedly extended RTT time, and the time needed to retransmit the lost UDP packet. The effective la- tency value will be the maximum of this options' value and the value of peerlatency set by the peer side. Before version 1.3.0 this option is only available as latency. recv_buffer_size=bytes Set UDP receive buffer size, expressed in bytes. send_buffer_size=bytes Set UDP send buffer size, expressed in bytes. timeout=microseconds Set raise error timeouts for read, write and connect operations. Note that the SRT library has internal timeouts which can be controlled sep- arately, the value set here is only a cap on those. tlpktdrop=1|0 Too-late Packet Drop. When enabled on receiver, it skips missing pack- ets that have not been delivered in time and delivers the following 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, the sender drops the older packets that have no chance of being delivered in time. It was automatically enabled in the sender if the receiver supports it. sndbuf=bytes Set send buffer size, expressed in bytes. rcvbuf=bytes Set receive buffer size, expressed in bytes. Receive buffer must not be greater than ffs. lossmaxttl=packets The value up to which the Reorder Tolerance may grow. When Reorder Tol- erance is > 0, then packet loss report is delayed until that number of packets come in. Reorder Tolerance increases every time a "belated" packet has come, but it wasn't due to retransmission (that is, when UDP packets tend to come out of order), with the difference between the latest sequence and this packet's sequence, and not more than the value of this option. By default it's 0, which means that this mechanism is turned off, and the loss report is always sent immediately upon experi- encing a "gap" in sequences. minversion 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. The version format in hex is 0xXXYYZZ for x.y.z in human readable form. streamid=string A string limited to 512 characters that can be set on the socket prior to connecting. This stream ID will be able to be retrieved by the lis- tener side from the socket that is returned from srt_accept and was connected by a socket with that set stream ID. SRT does not enforce any special interpretation of the contents of this string. This option doesnat make sense in Rendezvous connection; the result might be that simply one side will override the value from the other side and itas the matter of luck which one would win srt_streamid=string Alias for streamid to avoid conflict with ffmpeg command line option. smoother=live|file The type of Smoother used for the transmission for that socket, which is responsible for the transmission and congestion control. The Smoother type must be exactly the same on both connecting parties, oth- erwise the connection is rejected. messageapi=1|0 When set, this socket uses the Message API, otherwise it uses Buffer API. Note that in live mode (see transtype) thereas only message API available. In File mode you can chose to use one of two modes: Stream API (default, when this option is false). In this mode you may send as many data as you wish with one sending instruction, or even use dedicated functions that read directly from a file. The internal facil- ity 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 ex- actly 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 mes- sage is not complete (not all packets received or there was a packet loss) it will not be given up. transtype=live|file Sets the transmission type for the socket, in particular, setting this option sets multiple other parameters to their default values as re- quired for a particular transmission type. live: Set options as for live transmission. In this mode, you should send by one sending instruction only so many data that fit in one UDP packet, and limited to the value defined first in payload_size (1316 is default in this mode). There is no speed control in this mode, only the bandwidth control, if configured, in order to not exceed the bandwidth with the overhead transmission (retransmitted and control packets). file: Set options as for non-live transmission. See messageapi for fur- ther explanations linger=seconds The number of seconds that the socket waits for unsent data when clos- ing. Default is -1. -1 means auto (off with 0 seconds in live mode, on with 180 seconds in file mode). The range for this option is integers in the 0 - "INT_MAX". tsbpd=1|0 When true, use Timestamp-based Packet Delivery mode. The default behav- ior depends on the transmission type: enabled in live mode, disabled in file mode. For more information see: <https://github.com/Haivision/srt>. 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. If set to 0, extract till end of file. Examples: Extract a chapter from a DVD VOB file (start and end sectors obtained ex- ternally 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 Play a MPEG-TS file from start offset till end: subfile,,start,32815239,end,0,,:video.ts tee Writes the output to multiple protocols. The individual outputs are sepa- rated by | tee:file://path/to/local/this.avi|file://path/to/local/that.avi 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. Stan- dard percent-encoding (and using the plus sign for space) can be used to escape keys and values. Options can also can be specified via command line options (or in code via "AVOption"s). The list of supported options follows. listen=2|1|0 Listen for an incoming connection. 0 disables listen, 1 enables listen in single client mode, 2 enables listen in multi-client mode. Default value is 0. local_addr=addr Local IP address of a network interface used for tcp socket connect. local_port=port Local port used for tcp socket connect. 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. recv_buffer_size=bytes Set receive buffer size, expressed bytes. send_buffer_size=bytes Set send buffer size, expressed bytes. tcp_nodelay=1|0 Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0. Remark: Writing to the socket is currently not optimized to minimize system calls and reduces the efficiency / effect of TCP_NODELAY. tcp_mss=bytes Set maximum segment size for outgoing TCP packets, expressed in bytes. 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>] options contains a list of &-separated options of the form key=val. Stan- dard percent-encoding (and using the plus sign for space) can be used to escape keys and values. Options can also can be specified via command line options (or in code via "AVOption"s). The list of supported options follows. 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, 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 data- base, but it does not validate that the certificate actually matches the host name we are trying to connect to. (With other backends, the host name is validated as well.) This is disabled by default since it requires a CA database to be pro- vided 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 se- tups.) 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. http_proxy The HTTP proxy to tunnel through, e.g. "http://example.com:1234". The proxy must support the CONNECT method. 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=<server.key> To play back a stream from the TLS/SSL server using ffplay: ffplay tls://<hostname>:<port> dtls Datagram Transport Layer Security (DTLS) The required syntax for a DTLS URL is: dtls://<hostname>:<port>[?<options>] options contains a list of &-separated options of the form key=val. Stan- dard percent-encoding (and using the plus sign for space) can be used to escape keys and values. Options can also can be specified via command line options (or in code via "AVOption"s). DTLS shares most options with TLS, but operates over UDP instead of TCP. The list of supported options follows. 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, 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 data- base, but it does not validate that the certificate actually matches the host name we are trying to connect to. This is disabled by default since it requires a CA database to be pro- vided 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 se- tups.) key_file, key=filename A file containing the private key for the certificate. cert_pem=string Certificate PEM string key_pem=string Private key PEM string 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. mtu=size Set the Maximum Transmission Unit (MTU) for DTLS packets. use_srtp=1|0 Enable the use_srtp DTLS extension. This is used in WebRTC applica- tions to establish SRTP encryption keys through the DTLS handshake. De- fault is disabled. external_sock=1|0 Use an external socket instead of creating a new one. This option only makes sense to pass when interacting with the code via API, enabling this from CLI will cause immediate failure. Default is disabled. Example command lines: To create a DTLS server: ffmpeg -listen 1 -i dtls://<hostname>:<port> <output> To create a DTLS client and send data to server: ffmpeg -i <input> -f <format> dtls://<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. Stan- dard percent-encoding (and using the plus sign for space) can be used to escape keys and values. Options can also can be specified via command line options (or in code via "AVOption"s). 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 re- lated 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 32 KB for output, 384 KB for input. See also fifo_size. bitrate=bitrate If set to nonzero, the output will have the specified constant bitrate if the input has enough packets to sustain it. burst_bits=bits When using bitrate this specifies the maximum number of bits in packet bursts. localport=port Override the local UDP port to bind with. localaddr=addr Local IP address of a network interface used for sending packets or joining multicast groups. 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). dscp=dscp Set the 6-bit DSCP field for outgoing packets. 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 des- tination 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 ad- dress 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 from the specified addresses. In case of mul- ticast, also subscribe to multicast traffic coming from these addresses only. block=address[,address] Ignore packets sent from the specified addresses. In case of multicast, also exclude the source addresses in the multicast subscription. 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=65535 * 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. typeSelect socket type. stream Corresponds to SOCK_STREAM (reliable stream-oriented) datagram Corresponds to SOCK_DGRAM (unreliable packet-oriented) seqpacket Corresponds to SOCK_SEQPACKET (reliable packet-oriented) pkt_size Maximum packet size for packet-oriented sockets (SOCK_DGRAM and SOCK_SEQPACKET). If greater than zero, this value is used as "max_packet_size". Ignored for SOCK_STREAM. Default is 0. zmq ZeroMQ asynchronous messaging using the libzmq library. This library supports unicast streaming to multiple clients without relying on an external server. The required syntax for streaming or connecting to a stream is: zmq:tcp://ip-address:port Example: Create a localhost stream on port 5555: ffmpeg -re -i input -f mpegts zmq:tcp://127.0.0.1:5555 Multiple clients may connect to the stream using: ffplay zmq:tcp://127.0.0.1:5555 Streaming to multiple clients is implemented using a ZeroMQ Pub-Sub pat- tern. The server side binds to a port and publishes data. Clients connect to the server (via IP address/port) and subscribe to the stream. The order in which the server and client start generally does not matter. ffmpeg must be compiled with the --enable-libzmq option to support this protocol. Options can be set on the ffmpeg/ffplay command line. The following options are supported: pkt_size Forces the maximum packet size for sending/receiving data. The default value is 131,072 bytes. On the server side, this sets the maximum size of sent packets via ZeroMQ. On the clients, it sets an internal buffer size for receiving packets. Note that pkt_size on the clients should be equal to or greater than pkt_size on the server. Otherwise the received message may be truncated causing decoding errors. SEE ALSO ffmpeg(1), ffplay(1), ffprobe(1), libavformat(3) AUTHORS The FFmpeg developers. For details about the authorship, see the Git history of the project (https://git.ffmpeg.org/ffmpeg), e.g. by typing the command git log in the FFmpeg source directory, or browsing the online repository at <https://git.ffmpeg.org/ffmpeg>. Maintainers for the specific components are listed in the file MAINTAINERS in the source code tree. FFMPEG-PROTOCOLS(1)
NAME | DESCRIPTION | PROTOCOL OPTIONS | PROTOCOLS | SEE ALSO | AUTHORS
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