sipp测试opensipps

测试参照：http://www.kamailio.org/docs/openser-performance-tests/#tm-tests

1、测试register

SIP entities definition:

UAC 192.168.15.157:random
Proxy - 192.168.5.155:5060

SIP messages flow:
UAC --->REGISTER---> Proxy
UAC <---200 OK<--- Proxy

命令：./sipp 192.168.15.155 -sf uac-reg.xml -inf sip-users-random.txt -r 20000 -m 100000 -trace_rtt -trace_screen -l 100

-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.

-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),

random (RANDOM), or user (USER) 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. Several CSV files can be used simultaneously (syntax:-inf f1.csv -inf f2.csv ...)

-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 * rate_scale,  pressing '-' key to decrease call rate by 1 * rate_scale, pressing '*' key to increase call rate by 10 * rate_scale, pressing '/' key to

decrease call rate by 10 * rate_scale. If the -rp option is used, the call rate is calculated with the period in ms given by the user.

-m：Stop the test and exit when 'calls' calls are processed

-trace_rtt：Allow tracing of all response times in <scenario file name>_<pid>_rtt.csv.

-trce_screen：Dump statistic screens in the <scenario_name>_<pid>_cenaris.log file when quitting SIPp. Useful to get a final status report in background mode (-bg option).

-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).

2、测试呼叫

SIP entities definition:

UAC 192.168.2.100:5060 and 192.168.2.101:5060
Proxy - 192.168.2.102:5060
UAS - 192.168.2.102:5070
SIP messages flow:

UAC --->MESSAGE---> Proxy --->MESSAGE---> UAS
UAC <---200 OK<--- Proxy <---200 OK<--- UAS

命令：./sipp -sf uac_msg.xml -rsa 192.168.15.151:5060 192.168.15.155:5060 -m 200000 -r 10000 -d 1 -l 70

-rsa：Set the remote sending address to host:port for sending the messages.

-d：Controls the length 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 and default unit is milliseconds.

3、TM - Call-oriented tests

SIP entities definition:

UAC 192.168.2.100:5060 and 192.168.2.101:5060
Proxy - 192.168.2.102:5060
UAS - 192.168.2.102:5070
SIP messages flow for one call:

UAC --->INVITE---> Proxy    
UAC <---100 Trying<--- Proxy    
    Proxy --->INVITE---> UAS
UAC <---200 OK<--- Proxy <---200 OK<--- UAS
UAC --->ACK---> Proxy --->ACK---> UAS
UAC --->BYE---> Proxy --->BYE---> UAS
UAC <---200 OK<--- Proxy <---200 OK<--- UAS

命令：./sipp -sf uac_inv.xml -rsa 192.168.15.155:5060 192.168.15.151:5060 -m 200000 -r 7000 -d 1 -l 27