librtmp − RTMPDump Real-Time Messaging Protocol API
RTMPDump RTMP (librtmp, -lrtmp)
The Real-Time Messaging Protocol (RTMP) is used for streaming
multimedia content across a TCP/IP network. This API 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 basic
RTMP specification has been published by Adobe but this API was
reverse-engineered without use of the Adobe specification. As such, it may
deviate from any published specifications but it usually duplicates the
actual behavior of the original Adobe clients.
The RTMPDump software package includes a basic client utility program
some sample servers, and a library used to provide programmatic access
to the RTMP protocol. This man page gives an overview of the RTMP
library routines. These routines are found in the -lrtmp library. Many
other routines are also available, but they are not documented yet.
The basic interaction is as follows. A session handle is created using
and initialized using
All session parameters are provided using
The network connection is established using
and then the RTMP session is established using
The stream is read using
A client can publish a stream by calling
call, and then using
after the session is established.
While a stream is playing it may be paused and unpaused using
The stream playback position can be moved using
returns 0 bytes, the stream is complete and may be closed using
The session handle is freed using
All data is transferred using FLV format. The basic session requires
an RTMP URL. The RTMP URL format is of the form
Plain rtmp, as well as tunneled and encrypted sessions are supported.
Additional options may be specified by appending space-separated
key=value pairs to the URL. Special characters in values may need
to be escaped to prevent misinterpretation by the option parser.
The escape encoding uses a backslash followed by two hexadecimal digits
representing the ASCII value of the character. E.g., spaces must
be escaped as \20 and backslashes must be escaped as \5c.
These options define how to connect to the media server.
Use the specified SOCKS4 proxy.
These options define the content of the RTMP Connect request packet.
If correct values are not provided, the media server will reject the
Name of application to connect to on the RTMP server. Overrides
the app in the RTMP URL. Sometimes the librtmp URL parser cannot
determine the app name automatically, so it must be given explicitly
using this option.
URL of the target stream. Defaults to rtmp[t][e|s]://host[:port]/app.
URL of the web page in which the media was embedded. By default no
value will be sent.
URL of the SWF player for the media. By default no value will be sent.
Version of the Flash plugin used to run the SWF player. The
default is "LNX 10,0,32,18".
Append arbitrary AMF data to the Connect message. The type
must be B for Boolean, N for number, S for string, O for object, or Z
for null. 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, e.g.
NB:myFlag:1. This option may be used multiple times to construct arbitrary
AMF sequences. E.g.
conn=B:1 conn=S:authMe conn=O:1 conn=NN:code:1.23 conn=NS:flag:ok conn=O:0
These options take effect after the Connect request has succeeded.
Overrides the playpath parsed from the RTMP URL. Sometimes the
rtmpdump URL parser cannot determine the correct playpath
automatically, so it must be given explicitly using this option.
If the value is 1 or TRUE, issue a set_playlist command before sending the
play command. The playlist will just contain the current playpath. If the
value is 0 or FALSE, the set_playlist command will not be sent. The
default is FALSE.
Specify that the media is a live stream. No resuming or seeking in
live streams is possible.
Name of live stream to subscribe to. Defaults to
seconds into the stream. Not valid for live streams.
seconds into the stream.
Set buffer time to
milliseconds. The default is 30000.
Timeout the session after
seconds without receiving any data from the server. The default is 120.
These options handle additional authentication requests from the server.
Key for SecureToken response, used if the server requires SecureToken
JSON token used by legacy Justin.tv servers. Invokes NetStream.Authenticate.UsherToken
If the value is 1 or TRUE, the SWF player is retrieved from the
for performing SWF Verification. The SWF hash and size (used in the
verification step) are computed automatically. Also the SWF information is
cached in a
file in the user's home directory, so that it doesn't need to be retrieved
and recalculated every time. The .swfinfo file records
the SWF URL, the time it was fetched, the modification timestamp of the SWF
file, its size, and its hash. By default, the cached info will be used
for 30 days before re-checking.
Specify how many days to use the cached SWF info before re-checking. Use
0 to always check the SWF URL. Note that if the check shows that the
SWF file has the same modification timestamp as before, it will not be
An example character string suitable for use with
"rtmp://flashserver:1935/ondemand/thefile swfUrl=http://flashserver/player.swf swfVfy=1"
The value of
is used as the location for the
Cache of SWF Verification information