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
protocols. Protocols prefixed by "-" are disabled. All protocols
are allowed by default but protocols used by an another protocol
(nested protocols) 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
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.
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
separate 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 "/".
Muliple 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
exchange, 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 depending 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 matching (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
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.
concatf
Physical concatenation protocol using a line break delimited list of
resources.
Read and seek from many resources in sequence as if they were a unique
resource.
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 "Quoting 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
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///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" 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.
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-
seekable, -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).
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-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
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.
gophers
Gophers protocol.
The Gopher protocol with TLS encapsulation.
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 or for listen
mode.
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.
multiple_requests
Use persistent connections if set to 1, default is 0.
post_data
Set custom HTTP post data.
referer
Set the Referer header. Include 'Referer: URL' header in HTTP
request.
user_agent
Override the User-Agent header. If not specified the protocol will
use a string describing the libavformat build. ("Lavf/<version>")
reconnect_at_eof
If set then eof is treated like an error and causes reconnection,
this is useful for live / endless streams.
reconnect_streamed
If set then even streamed/non seekable streams will be reconnected
on errors.
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_delay_max
Sets the maximum delay in seconds after which to give up
reconnecting
mime_type
Export the MIME type.
http_version
Exports the HTTP response version number. Usually "1.0" or "1.1".
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 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>
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 applicable. Default value is -1.
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.
none
Choose the HTTP authentication type automatically. This is the
default.
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 encryption 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 authentication scheme other than
basic authentication. HTTPS/TLS should be used with basic
authentication. Without these additional security
enhancements, basic authentication should not be used to
protect sensitive or valuable information.
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.
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.
If a user doesn't have a node of their own then the public gateway
"https://dweb.link" is used by default.
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. If that fails
"https://dweb.link" will be used.
One can use this protocol in 2 ways. Using IPFS:
ffplay ipfs://QmbGtJg23skhvFmu9mJiePVByhfzu5rwo74MEkVDYAmF5T
Or the IPNS protocol (IPNS is mutable IPFS):
ffplay ipns://QmbGtJg23skhvFmu9mJiePVByhfzu5rwo74MEkVDYAmF5T
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.
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
mechanism 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
main
This 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).
Maximum 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 simulation, 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.
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.
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.
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
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 milliseconds 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.
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).
localport=n
Set the local RTP port to n.
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.
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.
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
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 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 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://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
omitted, 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
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.
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.
or
<options> srt://<hostname>:<port>
options contains a list of '-key val' options.
This protocol accepts the following options.
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 versions).
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. Default 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
sending 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 setting peerlatency, when side is sender
and rcvlatency when side is receiver, 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 settings. 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
inputbw. 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
(including empty, that is, with no encryption). If the password
doesn't match or only one side is unencrypted, the connection is
rejected. Default 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 decommissioned. 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
single 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. Default 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 latency 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 separately, the value set here is only a cap on those.
tlpktdrop=1|0
Too-late Packet Drop. When enabled on receiver, it skips missing
packets 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
Tolerance 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 experiencing 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 listener 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 doesnXt make sense in Rendezvous connection; the result
might be that simply one side will override the value from the
other side and itXs 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, otherwise 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) thereXs 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 facility will take care of any speed and congestion
control. When receiving, you can also receive as many data as
desired, the data not extracted will be waiting for the next call.
There is no boundary between data portions in the Stream mode.
Message API. In this mode your single sending instruction passes
exactly one piece of data that has boundaries (a message). Contrary
to Live mode, this message may span across multiple UDP packets and
the only size limitation is that it shall fit as a whole in the
sending buffer. The receiver shall use as large buffer as necessary
to receive the message, otherwise the message will not be given up.
When the message is not complete (not all packets received or there
was a packet loss) it will not be given up.
transtype=live|file
Sets the transmission type for the socket, in particular, setting
this option sets multiple other parameters to their default values
as required for a particular transmission type.
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
further explanations
linger=seconds
The number of seconds that the socket waits for unsent data when
closing. 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
behavior 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
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
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
separated 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.
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.
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>]
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 other
backends, 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.
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>
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 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).
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 from the specified addresses. In case of
multicast, 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.
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
pattern. 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)
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