sipp指令学习

常用指令：

sipp -i 192.168.10.189 -sf reg.xml -inf reg.csv 192.168.10.188:442 -r 100
sipp -i 192.168.10.189 -sf invite1.xml -inf invite.csv 192.168.10.188:442 -r 100
sipp -i 192.168.10.189 -sf reg.xml -inf reg.csv 192.168.10.188:442 -r 10000 -max_recv_loops 10000 -max_sched_loops 10000

 

188为目标机，189为客户机

 

 

指令说明：

 

sipp remote_host[:remote_port] [options]

 

  Available options:

   -v               : Display version and copyright information.   ；显示版本号与版权信息

   -aa              : Enable automatic 200 OK answer for INFO, UPDATE and ；对 INFO ,UPDATE，NOTIFY 回复 200OK

                      NOTIFY messages.

   -auth_uri        : Force the value of the URI for authentication.   ；验证 URI

                      By default, the URI is composed of    ；默认情况下 URI 由

                      remote_ip:remote_port.     ；remote_ip:remote_port 组成

   -base_cseq       : Start value of [cseq] for each call.    ；互相送 cseq 起始值

   -bg              : Launch SIPp in background mode.    ；运行 sipp 在后台模式

   -bind_local      : Bind socket to local IP address, i.e. the local IP   ；绑定本地IP与端口，本地IP地址会使用

                      address is used as the source IP address.  If SIPp runs ；源Ip 地址。如果sipp运行在UAS模式

                      in server mode it will only listen on the local IP  ；它仅会监听本地端口

                      address instead of all IP addresses.

   -buff_size       : Set the send and receive buffer size.    ；设定发送与接收的缓冲区大小

   -calldebug_file  : Set the name of the call debug file.    ；设定debug文件名

   -calldebug_overwrite: Overwrite the call debug file (default true).   ；重写呼叫 debug 文件名，

   -cid_str         : Call ID string (default %u-%p@%s).  %u=call_number,  ；呼叫 字符串 （默认是%u-%p@%s）

                      %s=ip_address, %p=process_number, %%=% (in any order). ；%u=call_number %s=ip_address %p=process_number %%=%(in any order)

   -ci              : Set the local control IP address    ；设定本地管理IP地址

   -cp              : Set the local control port number. Default is 8888.  ；设定本地管理 端口 默认 8888

   -d               : Controls the length of calls. More precisely, this  ；控制呼叫时间

                      controls the duration of 'pause' instructions in the  ；两次呼叫的中间时间

                      scenario, if they do not have a 'milliseconds' section.  ；如果没有设定 毫秒级的间隔

                      Default value is 0 and default unit is milliseconds.  ；默认值为 0，单位是 毫秒

   -deadcall_wait   : How long the Call-ID and final status of calls should be  ；设定bug提醒之间延时

                      kept to improve message and error logs (default unit is ；单位为 ms

                      ms).

   -default_behaviors: Set the default behaviors that SIPp will use.  Possbile  ；设定Sipp的默认行为 ：

                      values are:

                      - all     Use all default behaviors    ；使用默认行为

                      - none    Use no default behaviors    ；不使用默认行为

                      - bye     Send byes for aborted calls   ；默认送 bye

                      - abortunexp      Abort calls on unexpected messages ；终止 意外的呼叫

                      - pingreply       Reply to ping requests   ；回复Ping 请求

                      If a behavior is prefaced with a -, then it is turned  ；添加一个 - ，则代表关闭此功能

                      off.  Example: all,-bye     ；如 ，all，-bye

                    

   -error_file      : Set the name of the error log file.    ；设定error log 文件名

   -error_overwrite : Overwrite the error log file (default true).   ；重写 error log 文件

   -f               : Set the statistics report frequency on screen. Default is  ；设定在屏幕上显示的统计报告

                      1 and default unit is seconds.    ；默认是 1， 时间为秒

   -fd              : Set the statistics dump log report frequency. Default is  ；设定转储日志的时间与单位

                      60 and default unit is seconds.    ；默认是 60 s

   -i               : Set the local IP address for 'Contact:','Via:', and  ；设定本地IP地址，用在 ‘Contact：’‘Via：’‘From：’

                      'From:' headers. Default is primary host IP address.  ；头域， 默认是主机Ip 地址

                    

   -inf             : Inject values from an external CSV file during calls into  ；载入 CSV 场景文件

                      the scenarios.

                      First line of this file say whether the data is to be  ；SEQUENTIAL

                      read in sequence (SEQUENTIAL), random (RANDOM), or user ；RANDOM

                      (USER) order.      ；USER

                      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.

                      Several CSV files can be used simultaneously (syntax:

                      -inf f1.csv -inf f2.csv ...)

   -infindex        : file field       ；创建索引的引用段

                      Create an index of file using field.  For example -inf  ；范例

                      users.csv -infindex users.csv 0 creates an index on the  ；

                      first key.

   -ip_field        : Set which field from the injection file contains the IP  ；设定引用的文件中

                      address from which the client will send its messages.  ；用来 送出IP 字段关键字

                      If this option is omitted and the '-t ui' option is  ；如果这个设置被忽略并且 ‘-t ui’设置

                      present, then field 0 is assumed.    ；则假设 字段 为 0

                      Use this option together with '-t ui'    ；使用时 一般会 带有 ‘-t ui’

   -l               : Set the maximum number of simultaneous calls. Once this ；设定并发呼叫数

                      limit is reached, traffic is decreased until the number  ；

                      of open calls goes down. Default:    ；

                        (3 * call_duration (s) * rate).    ；默认 ：(3 * call_duration (s) * rate).

   -log_file        : Set the name of the log actions log file.

   -log_overwrite   : Overwrite the log actions log file (default true).

   -lost            : Set the number of packets to lose by default (scenario  ；设定没有场景文件如何拨打号码

                      specifications override this value).    ；场景文件会覆盖此设置

   -rtcheck         : Select the retransmisison detection method: full  ；设定 检查 RTP 语音流 full / loose

                      (default) or loose.      ；默认为 full

   -m               : Stop the test and exit when 'calls' calls are processed  ；停止呼叫的次数

   -mi              : Set the local media IP address (default: local primary  ；设定本地 media IP address

                      host IP address)

   -master          : 3pcc extended mode: indicates the master number 

   -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_sched_loops : Set the maximum number of calsl run per event loop.

                      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_log_size    : What is the limit for error and message log file sizes.

   -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).

   -message_file    : Set the name of the message log file.

   -message_overwrite: Overwrite the message log file (default true).

   -mp              : Set the local RTP echo port number. Default is 6000.   ；设定本地RTP端口，默认6000

   -nd              : No Default. Disable all default behavior of SIPp which   ；使用 没有默认行为的 SIPp

                      are the following:

                      - On UDP retransmission timeout, abort the call by   ；RTP语音超时

                        sending a BYE or a CANCEL     ；自动发送 BYE or CANCEL

                      - On receive timeout with no ontimeout attribute, abort  ；接收超时无超时属性，

                        the call by sending a BYE or a CANCEL    ；发送 BYE or CANCEL

                      - On unexpected BYE send a 200 OK and close the call  ；回200 OK 给 BYE消息

                      - On unexpected CANCEL send a 200 OK and close the call  ；回200 OK 给 CANCEL消息

                      - On unexpected PING send a 200 OK and continue the call  ；回200 OK 给 PING 消息并继续通话

                      - On any other unexpected message, abort the call by  ；其他的意外消息 ，发送BYE or CANCEL

                        sending a BYE or a CANCEL     ；回应并结束通话

                    

   -nr              : Disable retransmission in UDP mode.     ；禁止使用 UDP 重发模式

   -nostdin         : Disable stdin.       ；no stdin

                     

   -p               : Set the local port number.  Default is a random free port  ；设定本地端口

                      chosen by the system.      ；默认使用一个空闲的端口

   -pause_msg_ign   : Ignore the messages received during a pause defined in  ；忽略在暂停时场景文件对

                      the scenario        ；消息体的返回

   -periodic_rtd    : Reset response time partition counters each logging   ；重置每个分区响应计时器

                      interval.        ；在每个日志记录间隔

   -plugin          : Load a plugin.       ；加载插件。。插件？

   -r               : Set the call rate (in calls per seconds).  This value can   ；设置默认的 拨打时间 （呼叫/秒）

                      bechanged during test by pressing '+','_','*' or '/'.

                      Default is 10.       ；默认是10

                      pressing '+' key to increase call rate by 1 *    ；+ 加快1 个呼叫

                      rate_scale,

                      pressing '-' key to decrease call rate by 1 *    ；- 降低1 个呼叫

                      rate_scale,

                      pressing '*' key to increase call rate by 10 *    ；* 加快10 个呼叫

                      rate_scale,

                      pressing '/' key to decrease call rate by 10 *    ；/ 降低10 个呼叫

                      rate_scale.

                      If the -rp option is used, the call rate is calculated   ；如果加上 -rp这个参数

                      with the period in ms given by the user.    ；将会计算出每秒用户调用率

   -rp              : Specify the rate period for the call rate.  Default is 1   ；指定的 通话周期换用调用率

                      second and default unit is milliseconds.  This allows   ；默认是 1s 单位是 ms

                      you to have n calls every m milliseconds (by using -r n  ；这个选项将会让你设置 n个呼叫 /每毫秒

                      -rp m).

                      Example: -r 7 -rp 2000 ==> 7 calls every 2 seconds.   ；-r 7 -rp 2000 每2000ms 7个呼叫

                               -r 10 -rp 5s => 10 calls every 5 seconds.   ；-r 10 -rp 5s 每5s 10个呼叫

   -rate_scale      : Control the units for the '+', '-', '*', and '/' keys.

   -rate_increase   : Specify the rate increase every -fd units (default is

                      seconds).  This allows you to increase the load for each

                      independent logging period.

                      Example: -rate_increase 10 -fd 10s

                        ==> 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 -rate_max 100

                        ==> increase calls by 10 until 100 cps is hit.

   -no_rate_quit    : If -rate_increase is set, do not quit after the rate

                      reaches -rate_max.

   -recv_timeout    : Global receive timeout. Default unit is milliseconds. If

                      the expected message is not received, the call times out

                      and is aborted.

   -send_timeout    : Global send timeout. Default unit is milliseconds. If a

                      message is not sent (due to congestion), the call times

                      out and is aborted.

   -sleep           : How long to sleep for at startup. Default unit is

                      seconds.

   -reconnect_close : Should calls be closed on reconnect?

   -reconnect_sleep : How long (in milliseconds) to sleep between the close and

                      reconnect?

   -ringbuffer_files: How many error/message files should be kept after

                      rotation?

   -ringbuffer_size : How large should error/message files be before they get

                      rotated?

   -rsa             : Set the remote sending address to host:port for sending ；？？设定远程发送 地址与端口

                      the messages.

   -rtp_echo        : Enable RTP echo. RTP/UDP packets received on port defined ；开启 RTP 回升。 将接受的 RTP 包

                      by -mp are echoed to their sender.    ；回传到发送者

                      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  ；设定 resquest URI 的username 部分

                      'service'.       ；默认是  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.

   -shortmessage_file: Set the name of the short message log file.

   -shortmessage_overwrite: Overwrite the short message log file (default true).

   -oocsf           : Load out-of-call scenario.

   -oocsn           : Load out-of-call scenario.

   -skip_rlimit     : Do not perform rlimit tuning of file descriptor limits.

                      Default: false.

   -slave           : 3pcc extended mode: indicates the slave number

   -slave_cfg       : 3pcc extended mode: indicates the file where the master

                      and slave addresses are stored

   -sn              : Use a default scenario (embedded in the sipp executable). ；使用默认的场景文件

                      If this option is omitted, the Standard SipStone UAC  ；如果没设定场景文件又没设置这个参数

                      scenario is loaded.      ；将会默认使用 UAC

                      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:    ； 设定传输模式

                      - u1: UDP with one socket (default),   ；u1 UDP 一个端口传输 默认

                      - un: UDP with one socket per call,   ；un UDP 每个通话每个端口

                      - ui: UDP with one socket per IP address The IP ；ui UDP 每一个定义的IP 地址

                        addresses must be defined in the injection file. ；每个端口

                      - 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).  This plugin is not provided with sipp.

                     

   -timeout         : Global timeout. Default unit is seconds.  If this option

                      is set, SIPp quits after nb units (-timeout 20s quits

                      after 20 seconds).

   -timeout_error   : SIPp fails if the global timeout is reached is set

                      (-timeout option required).

   -timer_resol     : Set the timer resolution. Default unit is 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.

   -T2              : Global T2-timer in milli seconds

   -sendbuffer_warn : Produce warnings instead of errors on SendBuffer

                      failures.

   -trace_msg       : Displays sent and received SIP messages in <scenario file

                      name>_<pid>_messages.log

   -trace_shortmsg  : Displays sent and received SIP messages as CSV in

                      <scenario file name>_<pid>_shortmessages.log

   -trace_screen    : Dump statistic screens in the

                      <scenario_name>_<pid>_screens.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_calldebug : Dumps debugging information about aborted calls to

                      <scenario_name>_<pid>_calldebug.log file.

   -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_counts    : Dumps individual message counts in a CSV file.

   -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.

   -users           : Instead of starting calls at a fixed rate, begin 'users'

                      calls at startup, and keep the number of calls constant.

   -watchdog_interval: Set gap between watchdog timer firings.  Default is 400.  ；设定看门狗计时器 默认400

   -watchdog_reset  : If the watchdog timer has not fired in more than this

                      time period, then reset the max triggers counters.

                      Default is 10 minutes.

   -watchdog_minor_threshold: If it has been longer than this period between watchdog

                      executions count a minor trip.  Default is 500.

   -watchdog_major_threshold: If it has been longer than this period between watchdog

                      executions count a major trip.  Default is 3000.

   -watchdog_major_maxtriggers: How many times the major watchdog timer can be tripped

                      before the test is terminated.  Default is 10.

   -watchdog_minor_maxtriggers: How many times the minor watchdog timer can be tripped

                      before the test is terminated.  Default is 120.

   -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  ； 启动 3pcc 工具

                      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             : keyword value

                      Set the generic parameter named "keyword" to "value".

   -set             : variable value

                      Set the global variable parameter named "variable" to

                      "value".

   -dynamicStart    : variable value

                      Set the start offset of dynamic_id varaiable

   -dynamicMax      : variable value

                      Set the maximum of dynamic_id variable   

   -dynamicStep     : variable value

                      Set the increment of dynamic_id variable

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

   -2: Fatal error binding a socket

 

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