Difference between revisions of "SIPp command line parameters"
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Revision as of 02:01, 4 February 2007
Usage
sipp remote_host[:remote_port] [options]
Available options
- -v : Display version and copyright information.
- -aa : Enable automatic 200 OK answer for INFO and NOTIFY messages.
- -auth_uri : Force the value of the URI for authentication. By default, the URI is composed of remote_ip:remote_port.
- -base_cseq : Start value of [cseq] for each call.
- -bg : Launch SIPp in background mode.
- -bind_local : Bind socket to local IP address, i.e. the local IP address is used as the source IP address. If SIPp runs in server mode it will only listen on the local IP address instead of all IP addresses.
- -buff_size : Set the send and receive buffer size.
- -cid_str : Call ID string (default %u-%p@%s). %u=call_number, %s=ip_address, %p=process_number, %%=% (in any order).
- -d : Controls the length (in milliseconds) of calls. More precisely, this controls the duration of 'pause' instructions in the scenario, if they do not have a 'milliseconds' section. Default value is 0.
- -f : Set the statistics report frequency on screen (in seconds). Default is 1.
- -fd : Set the statistics dump log report frequency (in seconds). Default is 60.
- -i : Set the local IP address for 'Contact:','Via:', and 'From:' headers. Default is primary host IP address.
- -inf : Inject values from an external CSV file during calls into the scenarios.
First line of this file say whether the data is to be read in sequence (SEQUENTIAL) or random (RANDOM) order. Each line corresponds to one call and has one or more ';' delimited data fields. Those fields can be referred as [field0], [field1], ... in the xml scenario file.
- -ip_field : Set which field from the injection file contains the IP address from which the client will send its messages. If this option is omitted and the '-t ui' option is present, then field 0 is assumed. Use this option together with '-t ui'
- -l : Set the maximum number of simultaneous calls. Once this limit is reached, traffic is decreased until the number of open calls goes down. Default: (3 * call_duration (s) * rate).
- -m : Stop the test and exit when 'calls' calls are processed
- -mi : Set the local media IP address
- -max_recv_loops : Set the maximum number of messages received read per cycle. Increase this value for high traffic level. The default value is 1000.
- -max_reconnect : Set the the maximum number of reconnection.
- -max_retrans : Maximum number of UDP retransmissions before call ends on timeout. Default is 5 for INVITE transactions and 7 for others.
- -max_invite_retrans: Maximum number of UDP retransmissions for invite transactions before call ends on timeout.
- -max_non_invite_retrans: Maximum number of UDP retransmissions for non-invite transactions before call ends on timeout.
- -max_socket : Set the max number of sockets to open simultaneously. This option is significant if you use one socket per call. Once this limit is reached, traffic is distributed over the sockets already opened. Default value is 50000
- -mb : Set the RTP echo buffer size (default: 2048).
- -mp : Set the local RTP echo port number. Default is 6000.
- -nd : No Default. Disable all default behavior of SIPp which are the following:
- On UDP retransmission timeout, abort the call by sending a BYE or a CANCEL
- On receive timeout with no ontimeout attribute, abort the call by sending a BYE or a CANCEL
- On unexpected BYE send a 200 OK and close the call
- On unexpected CANCEL send a 200 OK and close the call
- On unexpected PING send a 200 OK and continue the call
- On any other unexpected message, abort the call by sending a BYE or a CANCEL
- -nr : Disable retransmission in UDP mode.
- -p : Set the local port number. Default is a random free port chosen by the system.
- -pause_msg_ign : Ignore the messages received during a pause defined in the scenario
-r : Set the call rate (in calls per seconds). This value can bechanged during test by pressing '+','_','*' or '/'. Default is 10.
- -pressing '+' key to increase call rate by 1,
- -pressing '-' key to decrease call rate by 1,
- -pressing '*' key to increase call rate by 10,
- -pressing '/' key to decrease call rate by 10.
- -If the -rp option is used, the call rate is calculated with the period in ms given by the user.
-rp : Specify the rate period in milliseconds for the call rate. Default is 1 second. This allows you to have n calls every m milliseconds (by using -r n -rp m). Example: -r 7 -rp 2000 ==> 7 calls every 2 seconds.
-rate_increase : Specify the rate increase every -fd seconds. This allows you to increase the load for each independent logging period. Example: -rate_increase 10 -fd 10 ==> increase calls by 10 every 10 seconds.
-rate_max : If -rate_increase is set, then quit after the rate reaches this value. Example: -rate_increase 10 -max_rate 100 ==> increase calls by 10 until 100 cps is hit.
-recv_timeout : Global receive timeout in milliseconds. If the expected message is not received, the call times out and is aborted.
-reconnect_close : Should calls be closed on reconnect?
-reconnect_sleep : How long to sleep between the close and reconnect?
-rsa : Set the remote sending address to host:port for sending the messages.
-rtp_echo : Enable RTP echo. RTP/UDP packets received on port defined by -mp are echoed to their sender. RTP/UDP packets coming on this port + 2 are also echoed to their sender (used for sound and video echo).
-rtt_freq : freq is mandatory. Dump response times every freq calls in the log file defined by -trace_rtt. Default value is 200.
-s : Set the username part of the resquest URI. Default is 'service'.
-sd : Dumps a default scenario (embeded in the sipp executable)
-sf : Loads an alternate xml scenario file. To learn more about XML scenario syntax, use the -sd option to dump embedded scenarios. They contain all the necessary help.
-sn : Use a default scenario (embedded in the sipp executable). If this option is omitted, the Standard SipStone UAC scenario is loaded. Available values in this version:
- 'uac' : Standard SipStone UAC (default). - 'uas' : Simple UAS responder. - 'regexp' : Standard SipStone UAC - with regexp and variables. - 'branchc' : Branching and conditional branching in scenarios - client. - 'branchs' : Branching and conditional branching in scenarios - server.
Default 3pcc scenarios (see -3pcc option):
- '3pcc-C-A' : Controller A side (must be started after all other 3pcc scenarios) - '3pcc-C-B' : Controller B side. - '3pcc-A' : A side. - '3pcc-B' : B side.
-stat_delimiter : Set the delimiter for the statistics file
-stf : Set the file name to use to dump statistics
-t : Set the transport mode: - u1: UDP with one socket (default), - un: UDP with one socket per call, - ui: UDP with one socket per IP address The IP addresses must be defined in the injection file. - t1: TCP with one socket, - tn: TCP with one socket per call, - l1: TLS with one socket, - ln: TLS with one socket per call, - c1: u1 + compression (only if compression plugin loaded), - cn: un + compression (only if compression plugin loaded).
-timeout : Global timeout in seconds. If this option is set, SIPp quits after nb seconds.
-timer_resol : Set the timer resolution in milliseconds. This option has an impact on timers precision.Small values allow more precise scheduling but impacts CPU usage.If the compression is on, the value is set to 50ms. The default value is 10ms.
-trace_msg : Displays sent and received SIP messages in <scenario file name>_<pid>_messages.log
-trace_screen : Dump statistic screens in the <scenario_name>_<pid>_ s.log file when quitting SIPp. Useful to get a final status report in background mode (-bg option).
-trace_err : Trace all unexpected messages in <scenario file name>_<pid>_errors.log.
-trace_timeout : Displays call ids for calls with timeouts in <scenario file name>_<pid>_timeout.log
-trace_stat : Dumps all statistics in <scenario_name>_<pid>.csv file. Use the '-h stat' option for a detailed description of the statistics file content.
-trace_rtt : Allow tracing of all response times in <scenario file name>_<pid>_rtt.csv.
-trace_logs : Allow tracing of <log> actions in <scenario file name>_<pid>_logs.log.
-up_nb : Set the number of updates of the internal clock during the reading of received messages. Default value is 1.
-ap : Set the password for authentication challenges. Default is 'password
-tls_cert : Set the name for TLS Certificate file. Default is 'cacert.pem
-tls_key : Set the name for TLS Private Key file. Default is 'cakey.pem'
-tls_crl : Set the name for Certificate Revocation List file. If not specified, X509 CRL is not activated.
-3pcc : Launch the tool in 3pcc mode ("Third Party call control"). The passed ip address is depending on the 3PCC role. - When the first twin command is 'sendCmd' then this is the address of the remote twin socket. SIPp will try to connect to this address:port to send the twin command (This instance must be started after all other 3PCC scenarii). Example: 3PCC-C-A scenario. - When the first twin command is 'recvCmd' then this is the address of the local twin socket. SIPp will open this address:port to listen for twin command. Example: 3PCC-C-B scenario.
-tdmmap : Generate and handle a table of TDM circuits. A circuit must be available for the call to be placed. Format: -tdmmap {0-3}{99}{5-8}{1-31}
-key : key value Set the generic parameter named "key" to "value".
Signal handling:
SIPp can be controlled using posix signals. The following signals are handled: USR1: Similar to press 'q' keyboard key. It triggers a soft exit of SIPp. No more new calls are placed and all ongoing calls are finished before SIPp exits. Example: kill -SIGUSR1 732 USR2: Triggers a dump of all statistics screens in <scenario_name>_<pid>_screens.log file. Especially useful in background mode to know what the current status is. Example: kill -SIGUSR2 732
Exit code:
Upon exit (on fatal error or when the number of asked calls (-m option) is reached, sipp exits with one of the following exit code: 0: All calls were successful 1: At least one call failed 97: exit on internal command. Calls may have been processed 99: Normal exit without calls processed -1: Fatal error
Example:
Run sipp with embedded server (uas) scenario: ./sipp -sn uas On the same host, run sipp with embedded client (uac) scenario ./sipp -sn uac 127.0.0.1