Protocols
*********

Protocols are configured elements in FFmpeg which allow to access resources which require the use of a particular protocol.

When you configure your FFmpeg build, all the supported protocols are enabled by default. You can list all available ones using the configure option "--list-protocols".

You can disable all the protocols using the configure option "--disable-protocols", and selectively enable a protocol using the option "--enable-protocol=PROTOCOL", or you can disable a particular protocol using the option "--disable-protocol=PROTOCOL".

The option "-protocols" of the ff* tools will display the list of supported protocols.

A description of the currently available protocols follows.

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

concat
======

Physical concatenation protocol.

Allow to read and seek from many resource 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.

file
====

File access protocol.

Allow to read from or read to a file.

For example to read from a file input.mpeg with ffmpeg use the command:

     ffmpeg -i file:input.mpeg output.mpeg

The ff* tools default to the file protocol, that is a resource specified with the name "FILE.mpeg" is interpreted as the URL "file:FILE.mpeg".

gopher
======

Gopher protocol.

hls
===

Read Apple HTTP Live Streaming compliant segmented stream as a uniform one. The M3U8 playlists describing the segments can be remote HTTP resources or local files, accessed using the standard file protocol. The nested protocol is declared by specifying "+PROTO" after the hls URI scheme name, where PROTO is either "file" or "http".

     hls+http://host/path/to/remote/resource.m3u8
     hls+file://path/to/local/resource.m3u8

Using this protocol is discouraged - the hls demuxer should work just as well (if not, please report the issues) and is more complete. To use the hls demuxer instead, simply use the direct URLs to the m3u8 files.

http
====

HTTP (Hyper Text Transfer Protocol).

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.

Allow to 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

Note that some formats (typically MOV), require the output protocol to be seekable, so they will fail with the pipe output protocol.

rtmp
====

Real-Time Messaging Protocol.

The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia content across a TCP/IP network.

The required syntax is:

     rtmp://SERVER[:PORT][/APP][/INSTANCE][/PLAYPATH]

The accepted parameters are:

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 AVOptions):

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.

rtmp_flush_interval
     Number of packets flushed in the same request (RTMPT only). The default is 10.

rtmp_live
     Specify that the media is a live stream. No resuming or seeking in live streams is possible. The default value is any, which means the subscriber first tries to play the live stream specified in the playpath. If a live stream of that name is not found, it plays the recorded stream. The other possible values are live and recorded.

rtmp_pageurl
     URL of the web page in which the media was embedded. By default no value will be sent.

rtmp_playpath
     Stream identifier to play or to publish. This option overrides the parameter specified in the URI.

rtmp_subscribe
     Name of live stream to subscribe to. By default no value will be sent. It is only sent if the option is specified or if rtmp_live is set to live.

rtmp_swfhash
     SHA256 hash of the decompressed SWF file (32 bytes).

rtmp_swfsize
     Size of the decompressed SWF file, required for SWFVerification.

rtmp_swfurl
     URL of the SWF player for the media. By default no value will be sent.

rtmp_swfverify
     URL to player swf file, compute hash/size automatically.

rtmp_tcurl
     URL of the target stream. Defaults to proto://host[:port]/app.

For example to read with ffplay a multimedia resource named "sample" from the application "vod" from an RTMP server "myserver":

     ffplay rtmp://myserver/vod/sample

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.

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 Protocol.

rtsp
====

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 RTSP server (http://github.com/revmischa/rtsp-server)).

The required syntax for a RTSP url is:

     rtsp://HOSTNAME[:PORT]/PATH

The following options (set on the ffmpeg/ffplay command line, or set in code via AVOptions or in avformat_open_input), are supported:

Flags for rtsp_transport:

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.

Flags for rtsp_flags:

filter_src
     Accept packets only from negotiated peer address and port.

listen
     Act as a server, listening for an incoming connection.

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.

Example command lines:

To 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

To watch a stream tunneled over HTTP:

     ffplay -rtsp_transport http rtsp://server/video.mp4

To 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

To 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]

tcp
===

Trasmission Control Protocol.

The required syntax for a TCP url is:

     tcp://HOSTNAME:PORT[?OPTIONS]

listen
     Listen for an incoming connection

          ffmpeg -i INPUT -f FORMAT tcp://HOSTNAME:PORT?listen
          ffplay tcp://HOSTNAME:PORT

tls
===

Transport Layer Security/Secure Sockets Layer

The required syntax for a TLS/SSL url is:

     tls://HOSTNAME:PORT[?OPTIONS]

listen
     Act as a server, listening for an incoming connection.

cafile=FILENAME
     Certificate authority file. The file must be in OpenSSL PEM format.

cert=FILENAME
     Certificate file. The file must be in OpenSSL PEM format.

key=FILENAME
     Private key file.

verify=0|1
     Verify the peer's certificate.

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 a 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 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 socket buffer size in bytes. This is used both for the receiving and the sending buffer size.

localport=PORT
     Override the local UDP port to bind with.

localaddr=ADDR
     Choose the local IP address. This is useful e.g. if sending multicast and the host has multiple interfaces, where the user can choose which interface to send on by specifying the IP address of that interface.

pkt_size=SIZE
     Set the size in bytes of UDP packets.

reuse=1|0
     Explicitly allow or disallow reusing UDP sockets.

ttl=TTL
     Set the time to live value (for multicast only).

connect=1|0
     Initialize the UDP socket with connect(). In this case, the destination address can't be changed with ff_udp_set_remote_url later. If the destination address isn't known at the start, this option can be specified in ff_udp_set_remote_url, too. This allows finding out the source address for the packets with getsockname, and makes writes return with AVERROR(ECONNREFUSED) if "destination unreachable" is received. For receiving, this gives the benefit of only receiving packets from the specified peer address/port.

sources=ADDRESS[,ADDRESS]
     Only receive packets sent to the multicast group from one of the specified sender IP addresses.

block=ADDRESS[,ADDRESS]
     Ignore packets sent to the multicast group from the specified sender IP addresses.

fifo_size=UNITS
     Set the UDP receiving circular buffer size, expressed as a number of packets with size of 188 bytes. If not specified defaults to 7*4096.

overrun_nonfatal=1|0
     Survive in case of UDP receiving circular buffer overrun. Default value is 0.

Some usage examples of the UDP protocol with ffmpeg follow.

To stream over UDP to a remote endpoint:

     ffmpeg -i INPUT -f FORMAT udp://HOSTNAME:PORT

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

To receive over UDP from a remote endpoint:

     ffmpeg -i udp://[MULTICAST-ADDRESS]:PORT