127 lines
4.8 KiB
Text
127 lines
4.8 KiB
Text
# This is a StreamingLoadTool config file
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#player user agent name
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player QTS
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# Use the "clienttype" directive to specify whether StreamingLoadTool should make
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# RTSP / UDP connections or RTSP / HTTP connections or . Say "http" for
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# the latter, "udp" for the former. Say "reliableudp" for reliable UDP.
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# Say "tcp" for straight interleaved RTSP / RTP
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clienttype udp
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# If doing RTSP / HTTP, set droppost to "yes" if you would like StreamingLoadTool
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# to drop the POST half of each RTSP / HTTP connection after sending the
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# PLAY. "yes" best emulates the "real" client behavior.
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droppost yes
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# Set this to the # of concurrent clients you would like StreamingLoadTool to maintain(default = 1)
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# When not running forever, this is also the number of clients to load before exiting
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concurrentclients 1
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# Specify a connection port for each connection(default 554)
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port 554
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# Specify a proxy IP address in dotted-decimal form. If 0, StreamingLoadTool
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# will not use a proxy to connect
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proxyip 0
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# Client window (size of UDP socket buffers). Default = 32768
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# For reliable UDP, this affects packet loss and the server's retransmission algorithm
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clientwindow 32768
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# StreamingLoadTool should send a TEARDOWN after streaming for this # of seconds
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movielength 60
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# If runforever is set to "no", StreamingLoadTool will quit after finishing the
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# list of URLs provided below. If "yes" StreamingLoadTool will loop forever. (Default = no)
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runforever no
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# Each instance of StreamingLoadTool must have a unique httpcookie value. This
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# can also be specified on the command-line (see usage by doing StreamingLoadTool -v) (Default = 1)
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httpcookie 1
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# Set to "yes" if you would like StreamingLoadTool to generate a connection log
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shouldlog yes
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# Append junk data after each DESCRIBE request
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appendjunk no
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# Location to place the connection log
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logpath streamingloadtool.log
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# Interval in milliseconds between attempts to read media data. For acking
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# udp clients, this is also the interval between acks. (Default = 50 milliseconds)
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readinterval 10
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# how late should packets be allowed to be sent? Value is in seconds.
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# 0 = no late tolerance will be specified at all
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latetolerance 0
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# The "Packet-Range" header is used as an alternative to the standard "Range"
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# header on a Play request for specifying a range of packets. Leave
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# this line blank to issue a standard play, specify a packet range header
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# here to send that instead of standard Range header.
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packetplayheader
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# The overbuffer window size is the number of K bytes the server can send
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# ahead of time. This applies to all transports except "udp".
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overbufferwindowsize 5192
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# Set this to be the value of the x-RTP-Meta-Info header sent to the server.
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# If it is empty, no x-RTP-Meta-Info header will be sent. Otherwise, specify
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# the fields you would like to receive
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# rtpmetainfo tt;ft;pn;fd;md;sq
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# Set this to be the speed you want the streams at (1 = normal speed, 2 = 2x normal speed, etc; float)
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speed 1
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# Enable this to have StreamingLoadTool randomly thumb around
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randomthumb no
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# Set sendoptions "yes" to send an OPTIONS request before executing each DESCRIBE request
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sendoptions no
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# Set requestrandomdata "yes" to send an OPTIONS request with a random data request header before executing each DESCRIBE request
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requestrandomdata no
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# How many random bytes the server should send in the body of the OPTIONS response
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randomdatasize 0
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# How often should the client send RTCP messages in milliseconds. (Default = 5000; ACK's in reliableudp are sent ASAP)
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rtcpinterval 1000
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# List of rtsp URLs for StreamingLoadTool to execute
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url rtsp://foo.bar.com/sample.mov
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# List of users:passwords for authentication (format: user:password)
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user user0:pass0
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#The advertised bandwidth in bps (default: not sent)
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bandwidth 31235
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#The player buffer space per stream in bytes. Default of 0 is unlimited
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buffer 15000
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#target delay in milliseconds. Default 3000. Use 0 to turn it off
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#This is the desired amount of playback time to keep in the buffer.
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delay 10000
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#Start play time delay; expressed as a fraction of the target delay. This is how much data there should be in the buffer before
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#the media starts playing. Use 0.0 to start playing immediately, and 1.0 to start playing when the target delay has been met. Default = 5.0
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startdelay 0.5
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#Turn on 3GPP features and headers? 3GPP should be not be enabled on reliableudp transport; default = no
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enable3GPP yes
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# The advertised guarenteed bit rate(GBW), maximum bit rate(MBW), and maximum transfer delay(MTD) for the wireless link
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# Units are in kilobits per seconds and milliseconds.
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# Default = 0 = no specified value
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#GBW 32
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#MBW 128
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#MTD 2000
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# Enable a forced PlayoutDelay value for 3GPP RTCP packets (The value defined for the playoutDelay preference will be sent)
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enableForcePlayoutDelay no
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# PlayoutDelay value to use if enableForcePlayoutDelay is enabled
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playoutDelay 65535
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