一、安装sipp
1、下载地址：
https://github-production-release-asset-2e65be.s3.amazonaws.com/13161657/99df6100-9216-11e9-9439-d9c9f5284379?X-Amz-Algorithm=AWS4-HMAC-SHA256&X-Amz-Credential=AKIAIWNJYAX4CSVEH53A%2F20200904%2Fus-east-1%2Fs3%2Faws4_request&X-Amz-Date=20200904T024024Z&X-Amz-Expires=300&X-Amz-Signature=df9c9aeba4b219e5a3c6ea4cd0cbb76785900829a5294ac6d72f5c7ba4409f9b&X-Amz-SignedHeaders=host&actor_id=30207023&key_id=0&repo_id=13161657&response-content-disposition=attachment%3B%20filename%3Dsipp-3.6.0.tar.gz&response-content-type=application%2Foctet-stream
2、安装依赖包
yum install ncurse*
yum install openssl*
yum install lksctp*
yum install libpcap*
3、安装sipp3.6
tar -xvzf sipp-xxx.tar.gz
cd sipp
./configure --with-sctp --with-pcap --with-openssl
make
4、验证安装
sipp -v
二、环境设置
1、修改系统openfile限制
（1）vim /etc/security/limits.conf，添加：
* soft nofile 32768
* hard nofile 65535
（2）vim /etc/pam.d/login，添加：
session required /lib/security/pam_limits.so
（3）命令行输入：
ulimit -s unlimited
ulimit -a
2、修改freeswitch配置
（1）cd /etc/freeswitch/autoload_configs，编辑vim switch.conf.xml
# 修改
<param name="max-sessions" value="100000"/>
<param name="sessions-per-second" value="10000"/>
（2）修改拨号正则，cd /etc/freeswitch/dialplan，修改某个profile:
<?xml version="1.0" encoding="utf-8"?>
<include>
<extension name="Balance_load">
<condition field="destination_number" expression="^([2-6][0-9][0-9][0-9])$">
<action application="export" data="dialed_extension=$1"/>
<action application="set" data="sip_h_History-Info=${sip_history_info}"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="set" data="callId=${uuid}"/>
<action application="export" data="execute_on_answer=record_session /home/Records/${caller_id_number}.wav" />
<action application="bridge" data="{absolute_codec_string=pcma,callId=${uuid}}sofia/webphonetest/$1@${local_ip_v4}:56148" />
</condition>
</extension>
</include>
其中，@172.200.115.13:56148是被叫IP和端口
（3）添加default配置文件。
cd /etc/freeswitch/directory/default
# 3000 5999为自己需要的用户
for i in `seq 2000 5999`; do sed -e "s/1000/$i/g" 1000.xml > $i.xml ; done
（4）添加白名单，无需鉴权
cd /etc/freeswitch/autoload_configs
vim acl.conf.xml # 进入编辑模式修改
<list name="domains" default="deny">
<!-- domain= is special it scans the domain from the directory to build the ACL -->
<node type="allow" domain="$${domain}"/>
<!-- use cidr= if you wish to allow ip ranges to this domains acl. -->
<node type="allow" cidr="192.168.200.0/24"/>
<!--新增此行. -->
<node type="allow" cidr="10.10.81.0/24"/>
</list>
三、配置文件
1、uac.csv
SEQUENTIAL
2000;2050;[authentication username=2000 password=1234]
2001;2051;[authentication username=2001 password=1234]
2002;2052;[authentication username=2002 password=1234]
2003;2053;[authentication username=2003 password=1234]
2004;2054;[authentication username=2004 password=1234]
2005;2055;[authentication username=2000 password=1234]
2006;2056;[authentication username=2001 password=1234]
2007;2057;[authentication username=2002 password=1234]
2008;2058;[authentication username=2003 password=1234]
2009;2059;[authentication username=2004 password=1234]
2010;2060;[authentication username=2000 password=1234]
2011;2061;[authentication username=2001 password=1234]
2012;2062;[authentication username=2002 password=1234]
2013;2063;[authentication username=2003 password=1234]
2014;2064;[authentication username=2004 password=1234]
2015;2065;[authentication username=2000 password=1234]
2016;2066;[authentication username=2001 password=1234]
2017;2067;[authentication username=2002 password=1234]
2018;2068;[authentication username=2003 password=1234]
2019;2069;[authentication username=2004 password=1234]
2020;2070;[authentication username=2000 password=1234]
2021;2071;[authentication username=2001 password=1234]
2022;2072;[authentication username=2002 password=1234]
2023;2073;[authentication username=2003 password=1234]
2024;2074;[authentication username=2004 password=1234]
2025;2075;[authentication username=2000 password=1234]
2026;2076;[authentication username=2001 password=1234]
2027;2077;[authentication username=2002 password=1234]
2028;2078;[authentication username=2003 password=1234]
2029;2079;[authentication username=2004 password=1234]
2030;2080;[authentication username=2000 password=1234]
2031;2081;[authentication username=2001 password=1234]
2032;2082;[authentication username=2002 password=1234]
2033;2083;[authentication username=2003 password=1234]
2034;2084;[authentication username=2004 password=1234]
2035;2085;[authentication username=2000 password=1234]
2036;2086;[authentication username=2001 password=1234]
2037;2087;[authentication username=2002 password=1234]
2038;2088;[authentication username=2003 password=1234]
2039;2089;[authentication username=2004 password=1234]
2040;2090;[authentication username=2000 password=1234]
2041;2091;[authentication username=2001 password=1234]
2042;2092;[authentication username=2002 password=1234]
2043;2093;[authentication username=2003 password=1234]
2044;2094;[authentication username=2004 password=1234]
2045;2095;[authentication username=2000 password=1234]
2046;2096;[authentication username=2001 password=1234]
2047;2097;[authentication username=2002 password=1234]
2048;2098;[authentication username=2003 password=1234]
2049;2099;[authentication username=2004 password=1234]
2、uas.csv
SEQUENTIAL
2050;;[authentication username=2050 password=1234]
2051;;[authentication username=2051 password=1234]
2052;;[authentication username=2052 password=1234]
2053;;[authentication username=2053 password=1234]
2054;;[authentication username=2054 password=1234]
2055;;[authentication username=2050 password=1234]
2056;;[authentication username=2051 password=1234]
2057;;[authentication username=2052 password=1234]
2058;;[authentication username=2053 password=1234]
2059;;[authentication username=2054 password=1234]
2060;;[authentication username=2050 password=1234]
2061;;[authentication username=2051 password=1234]
2062;;[authentication username=2052 password=1234]
2063;;[authentication username=2053 password=1234]
2064;;[authentication username=2054 password=1234]
2065;;[authentication username=2050 password=1234]
2066;;[authentication username=2051 password=1234]
2067;;[authentication username=2052 password=1234]
2068;;[authentication username=2053 password=1234]
2069;;[authentication username=2054 password=1234]
2070;;[authentication username=2050 password=1234]
2071;;[authentication username=2051 password=1234]
2072;;[authentication username=2052 password=1234]
2073;;[authentication username=2053 password=1234]
2074;;[authentication username=2054 password=1234]
2075;;[authentication username=2050 password=1234]
2076;;[authentication username=2051 password=1234]
2077;;[authentication username=2052 password=1234]
2078;;[authentication username=2053 password=1234]
2079;;[authentication username=2054 password=1234]
2080;;[authentication username=2050 password=1234]
2081;;[authentication username=2051 password=1234]
2082;;[authentication username=2052 password=1234]
2083;;[authentication username=2053 password=1234]
2084;;[authentication username=2054 password=1234]
2085;;[authentication username=2050 password=1234]
2086;;[authentication username=2051 password=1234]
2087;;[authentication username=2052 password=1234]
2088;;[authentication username=2053 password=1234]
2089;;[authentication username=2054 password=1234]
2090;;[authentication username=2050 password=1234]
2091;;[authentication username=2051 password=1234]
2092;;[authentication username=2052 password=1234]
2093;;[authentication username=2053 password=1234]
2094;;[authentication username=2054 password=1234]
2095;;[authentication username=2050 password=1234]
2096;;[authentication username=2051 password=1234]
2097;;[authentication username=2052 password=1234]
2098;;[authentication username=2053 password=1234]
2099;;[authentication username=2054 password=1234]
3、regclient_set_c_port.xml
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="regclient">
<Global variables="c_port" /> <nop hide="true">
<action>
<assignstr assign_to="EXP" value="3600" />
</action>
</nop> <send>
<![CDATA[
REGISTER sip:[remote_ip] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: <sip:[field0]@[remote_ip]>;tag=acknnkkg.[call_number]
To: <sip:[field0]@[remote_ip]>
Call-ID: [call_id]
CSeq: 1 REGISTER
Contact: <sip:[field0]@[local_ip]:[$c_port]>
Max-Forwards: 70
Subject: Reg Performance Test
user-agent: SIPp client
Expires: [$EXP]
Content-Length: 0
]]>
</send> <recv response="401" optional="true" auth="true" next="auth" >
</recv> <recv response="403" optional="true" next="END">
</recv> <recv response="404" optional="true" next="END">
</recv> <recv response="200" next="END" timeout="5000">
</recv> <label id="auth" />
<send retrans="500">
<![CDATA[
REGISTER sip:[field0]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: [field0] <sip:[field0]@[remote_ip]:[remote_port]>;tag=[call_number]
To: [field0] <sip:[field0]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 2 REGISTER
Contact: sip:[field0]@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Reg Performance Test
user-agent: SIPp client
Expires: [$EXP]
[field2]
Content-Length: 0 ]]>
</send> <recv response="200" next="END" timeout="5000">
</recv> <label id="END"/>
<nop hide="true">
</nop> <!--<Reference variables="microseconds,seconds" />--> <!-- Definition of the response time repartition table (unit is ms) -->
<ResponseTimeRepartition value="50, 200"/> <!-- Definition of the call length repartition table (unit is ms) -->
<CallLengthRepartition value="500, 5000"/> </scenario>
4、caller_with_auth.xml
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd"> <scenario name="caller_with_auth">
<!--执行命令样例：sipp -sf caller_with_auth.xml xx.x.x.xx:5060 -p 5066 -inf caller.csv -m 1 -d 10000 -oocsn ooc_default-->
<!--发送INVITE消息，设定重传定时器为1000ms，同时启动定时器invite-->
<send retrans="1000" start_rtd="invite">
<![CDATA[
INVITE sip:[field1]@[remote_ip] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: "[field0]" <sip:[field0]@[remote_ip]>;tag=[call_number]zhg8
To: "[field1]"<sip:[field1]@[remote_ip]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: <sip:[field0]@[local_ip]:[local_port]>
User-Agent: SIPp client mode version [sipp_version]
Allow: INVITE,PRACK,ACK,UPDATE,CANCEL,BYE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: [len] v=0
o=SIPp [pid][call_number] 8[pid][call_number]8 IN IP[local_ip_type] [local_ip]
s=SIPp Normal Call Test
t=0 0
m=audio [media_port] RTP/AVP 0 8
c=IN IP[media_ip_type] [media_ip]
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv
]]>
</send> <recv response="100" optional="true">
</recv> <!-- <recv response="401" auth="true"> -->
<!-- </recv> --> <!-- 部分呼叫鉴权可能为407 -->
<!-- <recv response="407" option="true" auth="true">
</recv> <send>
<![CDATA[
ACK sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
From: "[field0]" <sip:[field0]@[remote_ip]>;tag=[call_number]zhg8
To: "[field1]"<sip:[field1]@[remote_ip]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: <sip:[field0]@[local_ip]:[local_port]>
Max-Forwards: 70
Subject: normal call scenario
user-agent: SIPp client mode version [sipp_version]
Content-Length: 0
]]>
</send> <send retrans="1000" start_rtd="invite">
<![CDATA[
INVITE sip:[field1]@[remote_ip] SIP/2.0
[last_Via:]
From: "[field0]" <sip:[field0]@[remote_ip]>;tag=[call_number]zhg8
To: "[field1]"<sip:[field1]@[remote_ip]>
Call-ID: [call_id]
CSeq: 2 INVITE
[field2]
Contact: <sip:[field0]@[local_ip]:[local_port]>
User-Agent: SIPp client mode version [sipp_version]
Allow: INVITE,PRACK,ACK,UPDATE,CANCEL,BYE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: [len] v=0
o=SIPp [pid][call_number] 8[pid][call_number]8 IN IP[local_ip_type] [local_ip]
s=SIPp Normal Call Test
t=0 0
m=audio [media_port] RTP/AVP 0 8
c=IN IP[media_ip_type] [media_ip]
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv ]]>
</send> <!--1xx响应均为可选接收消息，且接收到临时响应后，即可停止invite定时器的计时-->
<!--收到4xx/5xx错误响应后，直接进入呼叫失败-->
<!-- <recv response="100" optional="true" rtd="invite">
</recv> <recv response="183" optional="true" rtd="invite" next="normal">
</recv> <recv response="403" optional="true" rtd="invite" next="err_ack">
</recv> <recv response="407" optional="true" rtd="invite" next="err_ack">
</recv> <recv response="415" optional="true" rtd="invite" next="err_ack">
</recv> <recv response="480" optional="true" rtd="invite" next="err_ack">
</recv> <recv response="486" optional="true" rtd="invite" next="err_ack">
</recv> <recv response="500" optional="true" rtd="invite" next="err_ack">
</recv> <recv response="503" optional="true" rtd="invite" next="err_ack">
</recv> -->
-->
<recv response="180" optional="true" rtd="invite" next="normal">
</recv> <label id="normal"/>
<recv response="200" rtd="invite">
<action>
<ereg regexp="m=audio ([0-9]*)"
search_in="msg"
check_it="true"
assign_to="junk,callee_media_port" />
</action>
</recv> <nop hide="true"> </nop> <send>
<![CDATA[
ACK sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: "[field0]" <sip:[field0]@[remote_ip]>;tag=[call_number]zhg8
To: "[field1]"<sip:[field1]@[remote_ip]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 ACK
Contact: <sip:[field0]@[local_ip]:[local_port]>
Max-Forwards: 70
Subject: normal call scenario
user-agent: SIPp client mode version [sipp_version]
Content-Length: 0
]]>
</send> <!--使用rtp_stream循环播放PCMA音频
<nop hide="true">
<action>
<exec rtp_stream="pcap/g711a.pcap,-1,0"/>
</action>
</nop>
-->
<!--使用rtp_stream循环播放PCMU音频
<nop hide="true">
<action>
<exec rtp_stream="pcap/g711u.pcap,-1,0"/>
</action>
</nop>
--> <!--使用play_pcap单次播放PCMA音频-->
<nop hide="true">
<action>
<exec play_pcap_audio="pcap/g711a.pcap"/>
</action>
</nop>
<!--使用play_pcap单次播放PCMU音频
<nop hide="true">
<action>
<exec play_pcap_audio="pcap/g711u.pcap"/>
</action>
</nop>
--> <!--媒体流传输完毕后，暂停发送BYE结束呼叫，在执行命令中增加参数-d 指定暂停时间：如-d 10000暂停10秒-->
<pause /> <send start_rtd="bye">
<![CDATA[
BYE sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: "[field0]" <sip:[field0]@[remote_ip]>;tag=[call_number]zhg8
To: "[field1]"<sip:[field1]@[remote_ip]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 3 BYE
Max-Forwards: 70
Subject: normal call scenario
Content-Length: 0
]]>
</send> <recv response="200" rtd="bye" next="END">
</recv> <!--异常结束，复用err_ack流程-->
<label id="err_ack"/> <send>
<![CDATA[
ACK sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
[last_Via:]
From: "[field0]" <sip:[field0]@[remote_ip]>;tag=[call_number]zhg8
To: "[field1]"<sip:[field1]@[remote_ip]>[peer_tag_param]
[last_Call-ID:]
CSeq: 2 ACK
Max-Forwards: 70
Subject: normal call scenario
user-agent: SIPp client mode version [sipp_version]
Content-Length: 0
]]>
</send> <!--正常结束-->
<label id="END"/>
<nop hide="true">
</nop> <!--如果存在定义了但未被使用的变量，可以在下面语句的双引号中增加，避免运行时报错-->
<Reference variables="junk,callee_media_port" /> <!--definition of the response time repartition table (unit is ms) -->
<ResponseTimeRepartition value="50, 200,1000,2000,4000,10000"/> <!--definition of the call length repartition table (unit is ms) -->
<CallLengthRepartition value="500, 1000, 10000"/> </scenario>
5、callee_with_bye.xml
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd"> <scenario name="callee_with_bye">
<!--用于模拟局内被叫侧用户的正常业务流程
媒体类型：PCMU
呼叫挂机：主叫方（60秒超时后主动发BYE拆话）--> <!--执行命令样例：sipp -sf callee_with_bye.xml -p 5068--> <!--定义全局状态机，如果收到OPTIONS消息，则跳转至options标签处-->
<recv request="OPTIONS" optional="global" next="options">
</recv> <recv request="INVITE">
<!--参数caller_num、callee_num和caller_tag用于主叫未挂机，BYE接收超时主动发BYE的流程-->
<action>
<ereg regexp="sip:(.*)@(.*)>;tag=(.*)"
search_in="hdr"
header="From: "
check_it="true"
assign_to="junk,caller_num,domain,caller_tag" >
</ereg>
<ereg regexp="sip:(.*)@.*>"
search_in="hdr"
header="To: "
check_it="true"
assign_to="junk,callee_num" >
</ereg>
</action>
</recv> <!--增加间隔20ms，避免偶现系统不发送100响应的问题-->
<pause hide="true" milliseconds="20"/> <send>
<![CDATA[
SIP/2.0 100 Trying
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send> <!--增加间隔20ms，避免偶现系统不发送180响应的问题-->
<pause hide="true" milliseconds="20"/> <send>
<![CDATA[
SIP/2.0 180 Ringing
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send> <!--设置发送200后等待ACK的重传周期为1秒，如果1秒内收不到ACK则进行200的重传-->
<send retrans="1000" start_rtd="ack">
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact:<sip:[local_ip]:[local_port];transport=[transport]>
Content-Type: application/sdp
Content-Length: [len] v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
]]>
</send> <!--设置等待ACK的超时定时器为30秒，如果30秒内收不到ACK则呼叫超时失败而结束-->
<recv request="ACK" rtd="ack" timeout="30000" /> <!--使用rtp_stream循环播放PCMA音频
<nop hide="true">
<action>
<exec rtp_stream="pcap/g711a.pcap,-1,0"/>
</action>
</nop>
-->
<!--使用rtp_stream循环播放PCMU音频
<nop hide="true">
<action>
<exec rtp_stream="pcap/g711u.pcap,-1,0"/>
</action>
</nop>
--> <!--使用play_pcap单次播放PCMA音频-->
<nop hide="true">
<action>
<exec play_pcap_audio="pcap/g711a.pcap"/>
</action>
</nop>
<!--使用play_pcap单次播放PCMU音频
<nop hide="true">
<action>
<exec play_pcap_audio="pcap/g711u.pcap"/>
</action>
</nop>
--> <recv request="BYE" timeout="60000" ontimeout="send_bye"/>
<send next="END">
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send> <label id="options"/>
<send next="END" >
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_Call-ID:]
[last_From:]
[last_To:];tag=telpo-options[call_number]
[last_CSeq:]
[last_Contact:]
user-agent: SIPP version [sipp_version]
subject: reg performance
link-status: I am alive
Content-Length: 0 ]]>
</send> <!--主叫未挂机，BYE接收超时,被叫主动发BYE-->
<label id="send_bye"/>
<send start_rtd="bye">
<![CDATA[
BYE sip:[$caller_num]@[local_ip]:[local_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: "[$caller_num]" <sip:[$caller_num]@[local_ip]>;tag=[call_number]
To: "[$callee_num]"<sip:[$callee_num]@[local_ip]>;tag=[$caller_tag]
Call-ID: [call_id]
CSeq: 2 BYE
Max-Forwards: 70
Subject: normal call scenario
Content-Length: 0
]]>
</send> <recv response="200" rtd="bye">
</recv> <label id="END"/> <Reference variables="junk,domain" /> <!-- definition of the response time repartition table (unit is ms)-->
<ResponseTimeRepartition value="50, 200"/> <!-- definition of the call length repartition table (unit is ms)-->
<CallLengthRepartition value="500, 1000, 10000"/> </scenario>
四、命令
sipp -sf regclient_set_c_port.xml 172.29.50.60:5050 -i 172.29.50.60 -p 26000 -inf uac.csv -r 5 -rp 1000 -l 5 -m 100 sipp -sf regclient_set_c_port.xml 172.29.50.60:5050 -i 172.29.50.60 -p 56148 -inf uas.csv -r 5 -rp 1000 -l 5 -m 100 sipp -sf callee_with_bye.xml -i 172.29.50.60 -p 56148 -trace_err sipp -sf caller_with_auth.xml 172.29.50.60:5050 -i 172.29.50.60 -p 26000 -inf uac.csv -r 10 -rp 1000 -l 30 -m 100 -d 60000 -oocsn ooc_default -trace_err -aa # -r：并发数量
#-rp:并发的时间，单位ms,例如：-r 800 -rp 1000，就是每秒800并发
#-l:设置同时呼叫的最大数目；一旦达到此值，流量将被限制直到打的通话数下降；默认值3*call_duration(s)*rate
#-m:通话总数，当设置的通话数完成时，停止测试并退出；
#-d：自定义的通话时长，单位ms
#-aa:针对INFO, UPDATE 和 NOTIFY消息，进行200 OK自动回复应答；
