Protocols
*********

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

When you configure your Libav 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 av* tools will display the list of supported protocols.

A description of the currently available protocols follows.

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 avplay use the command:

     avplay 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 avconv use the command:

     avconv -i file:input.mpeg output.mpeg

The av* 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).

This protocol accepts the following options:

mime_type
     Export the MIME type.

icy
     If set to 1 request ICY (SHOUTcast) metadata from the server. If the server supports this, the metadata has to be retrieved by the application by reading the icy_metadata_headers and icy_metadata_packet options. The default is 0.

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.

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.
     avconv -i input.flv -f avi -y md5:output.avi.md5

     # Write the MD5 hash of the encoded AVI file to stdout.
     avconv -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 avconv:

     cat test.wav | avconv -i pipe:0
     # ...this is the same as...
     cat test.wav | avconv -i pipe:

For writing to stdout with avconv:

     avconv -i test.wav -f avi pipe:1 | cat > test.avi
     # ...this is the same as...
     avconv -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://[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 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. (When publishing, the default is FMLE/3.0 (compatible; <libavformat version>).)

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

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

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

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

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

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

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

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

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

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

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

     avplay rtmp://myserver/vod/sample

To publish to a password protected server, passing the playpath and app names separately:

     avconv -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.

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 avconv:

     avconv -re -i myfile -f flv rtmp://myserver/live/mystream

To play the same stream using avplay:

     avplay "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 avconv/avplay 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 avplay, 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:

     avplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4

To watch a stream tunneled over HTTP:

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

To send a stream in realtime to a RTSP server, for others to watch:

     avconv -re -i INPUT -f rtsp -muxdelay 0.1 rtsp://server/live.sdp

To receive a stream in realtime:

     avconv -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:

     avconv -re -i INPUT -f sap sap://224.0.0.255?same_port=1

Similarly, for watching in avplay:

     avconv -re -i INPUT -f sap sap://224.0.0.255

And for watching in avplay, over IPv6:

     avconv -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:

     avplay sap://

To play back the first stream announced on one the default IPv6 SAP multicast address:

     avplay 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

          avconv -i INPUT -f FORMAT tcp://HOSTNAME:PORT?listen
          avplay tcp://HOSTNAME:PORT

tls
===

Transport Layer Security (TLS) / Secure Sockets Layer (SSL)

The required syntax for a TLS url is:

     tls://HOSTNAME:PORT

The following parameters can be set via command line options (or in code via AVOptions):

ca_file
     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.

tls_verify=1|0
     If enabled, try to verify the peer that we are communicating with. Note, if using OpenSSL, this currently only makes sure that the peer certificate is signed by one of the root certificates in the CA database, but it does not validate that the certificate actually matches the host name we are trying to connect to. (With GnuTLS, the host name is validated as well.)

     This is disabled by default since it requires a CA database to be provided by the caller in many cases.

cert_file
     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
     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.

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. Follow the list of supported options.

buffer_size=SIZE
     set the UDP buffer size in bytes

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.

Some usage examples of the udp protocol with avconv follow.

To stream over UDP to a remote endpoint:

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

To stream in mpegts format over UDP using 188 sized UDP packets, using a large input buffer:

     avconv -i INPUT -f mpegts udp://HOSTNAME:PORT?pkt_size=188&buffer_size=65535

To receive over UDP from a remote endpoint:

     avconv -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 AVOptions):

timeout
     Timeout in ms.

listen
     Create the Unix socket in listening mode.