@item 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. @file{/ondemand/}, @file{/flash/live/}, etc.).
+(e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
+the value parsed from the URI through the @code{rtmp_app} option, too.
@item playpath
It is the path or name of the resource to play with reference to the
-application specified in @var{app}, may be prefixed by "mp4:".
+application specified in @var{app}, may be prefixed by "mp4:". You
+can override the value parsed from the URI through the @code{rtmp_playpath}
+option, too.
+
+@end table
+
+Additionally, the following parameters can be set via command line options
+(or in code via @code{AVOption}s):
+@table @option
+
+@item rtmp_app
+Name of application to connect on the RTMP server. This option
+overrides the parameter specified in the URI.
+
+@item rtmp_buffer
+Set the client buffer time in milliseconds. The default is 3000.
+
+@item rtmp_conn
+Extra arbitrary AMF connection parameters, parsed from a string,
+e.g. like @code{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. @code{NB:myFlag:1}). This option may be used multiple
+times to construct arbitrary AMF sequences.
+
+@item rtmp_flashver
+Version of the Flash plugin used to run the SWF player. The default
+is LNX 9,0,124,2.
+
+@item rtmp_flush_interval
+Number of packets flushed in the same request (RTMPT only). The default
+is 10.
+
+@item rtmp_live
+Specify that the media is a live stream. No resuming or seeking in
+live streams is possible. The default value is @code{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 @code{live} and
+@code{recorded}.
+
+@item rtmp_pageurl
+URL of the web page in which the media was embedded. By default no
+value will be sent.
+
+@item rtmp_playpath
+Stream identifier to play or to publish. This option overrides the
+parameter specified in the URI.
+
+@item rtmp_swfurl
+URL of the SWF player for the media. By default no value will be sent.
+
+@item rtmp_tcurl
+URL of the target stream. Defaults to proto://host[:port]/app.
@end table
ffplay rtmp://myserver/vod/sample
@end example
+@section 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.
+
+@section 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.
+
+@section 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.
+
+@section 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.
+
+@section 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.
+
@section rtmp, rtmpe, rtmps, rtmpt, rtmpte
Real-Time Messaging Protocol and its variants supported through
@table @option
@item filter_src
Accept packets only from negotiated peer address and port.
+@item listen
+Act as a server, listening for an incoming connection.
@end table
When receiving data over UDP, the demuxer tries to reorder received packets
ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
@end example
+To receive a stream in realtime:
+
+@example
+ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
+@end example
+
@section sap
Session Announcement Protocol (RFC 2974). This is not technically a
@end table
+@section tls
+
+Transport Layer Security/Secure Sockets Layer
+
+The required syntax for a TLS/SSL url is:
+@example
+tls://@var{hostname}:@var{port}[?@var{options}]
+@end example
+
+@table @option
+
+@item listen
+Act as a server, listening for an incoming connection.
+
+@item cafile=@var{filename}
+Certificate authority file. The file must be in OpenSSL PEM format.
+
+@item cert=@var{filename}
+Certificate file. The file must be in OpenSSL PEM format.
+
+@item key=@var{filename}
+Private key file.
+
+@item verify=@var{0|1}
+Verify the peer's certificate.
+
+@end table
+
+Example command lines:
+
+To create a TLS/SSL server that serves an input stream.
+
+@example
+ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key}
+@end example
+
+To play back a stream from the TLS/SSL server using @command{ffplay}:
+
+@example
+ffplay tls://@var{hostname}:@var{port}
+@end example
+
@section udp
User Datagram Protocol.
udp://@var{hostname}:@var{port}[?@var{options}]
@end example
-@var{options} contains a list of &-seperated options of the form @var{key}=@var{val}.
-Follow the list of supported options.
+@var{options} contains a list of &-separated options of the form @var{key}=@var{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 @var{fifo_size} and
+@var{overrun_nonfatal} options are related to this buffer.
+
+The list of supported options follows.
@table @option
@item buffer_size=@var{size}
-set the UDP buffer size in bytes
+Set the UDP socket buffer size in bytes. This is used both for the
+receiving and the sending buffer size.
@item localport=@var{port}
-override the local UDP port to bind with
+Override the local UDP port to bind with.
@item localaddr=@var{addr}
Choose the local IP address. This is useful e.g. if sending multicast
which interface to send on by specifying the IP address of that interface.
@item pkt_size=@var{size}
-set the size in bytes of UDP packets
+Set the size in bytes of UDP packets.
@item reuse=@var{1|0}
-explicitly allow or disallow reusing UDP sockets
+Explicitly allow or disallow reusing UDP sockets.
@item ttl=@var{ttl}
-set the time to live value (for multicast only)
+Set the time to live value (for multicast only).
@item connect=@var{1|0}
Initialize the UDP socket with @code{connect()}. In this case, the
unreachable" is received.
For receiving, this gives the benefit of only receiving packets from
the specified peer address/port.
+
+@item sources=@var{address}[,@var{address}]
+Only receive packets sent to the multicast group from one of the
+specified sender IP addresses.
+
+@item block=@var{address}[,@var{address}]
+Ignore packets sent to the multicast group from the specified
+sender IP addresses.
+
+@item fifo_size=@var{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.
+
+@item overrun_nonfatal=@var{1|0}
+Survive in case of UDP receiving circular buffer overrun. Default
+value is 0.
@end table
-Some usage examples of the udp protocol with @command{ffmpeg} follow.
+Some usage examples of the UDP protocol with @command{ffmpeg} follow.
To stream over UDP to a remote endpoint:
@example