安装依赖包
yum install make gcc gcc-c++ ncurses ncurses.x86_64 ncurses-devel ncurses-devel.x86_64 openssl libnet libpcap libpcap-devel libpcap.x86_64 libpcap-devel.x86_64 gsl gsl-devel

[general]

        context=default

        allowoverlap=no

        allowguest=no

        udpbindaddr=0.0.0.0

        tcpenable=no

        tcpbindaddr=0.0.0.0

        srvlookup=yes

        language=cn

        callevents=yes

        limitonpeers=no

        jbenable=yes

[sipp]

        host=192.168.106.141

        type=friend

        disallow=all

        allow=g729

        allow=g723

        allow=ulaw

        allow=alaw

        context=sipp

[general]

        static=yes

        writeprotect=no

        clearglobalvars=no

[globals]

[sipp]

        exten => ,,Answer

        exten => ,,SetMusicOnHold(default)

        exten => ,,WaitMusicOnHold()

        exten => ,,Hangup

<?xml version="1.0" encoding="ISO-8859-1" ?>

<!DOCTYPE scenario SYSTEM "sipp.dtd">

<!-- This program is free software; you can redistribute it and/or      -->

<!-- modify it under the terms of the GNU General Public License as     -->

<!-- published by the Free Software Foundation; either version  of the -->

<!-- License, or (at your option) any later version.                    -->

<!--                                                                    -->

<!-- This program is distributed in the hope that it will be useful,    -->

<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of     -->

<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the      -->

<!-- GNU General Public License for more details.                       -->

<!--                                                                    -->

<!-- You should have received a copy of the GNU General Public License  -->

<!-- along with this program; if not, write to the                      -->

<!-- Free Software Foundation, Inc.,                                    -->

<!--  Temple Place, Suite , Boston, MA  - USA             -->

<!--                                                                    -->

<!--                 Sipp 'uac' scenario with pcap (rtp) play           -->

<!--                                                                    -->

<scenario name="Basic Sipstone UAC">

  <!-- In client mode (sipp placing calls), the Call-ID MUST be         -->

  <!-- generated by sipp. To do so, use [call_id] keyword.                -->

  <send retrans="">

      <![CDATA[

      INVITE sip:[field1]@[remote_ip]:[remote_port] SIP/2.0

      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]

      From: sipp<sip:sipp@[local_ip]:[local_port]>;tag=[call_number]

      To: sut<sip:[field1]@[remote_ip]:[remote_port]>

      Call-ID: [call_id]

      CSeq:  INVITE

      Contact: sip:sipp@[local_ip]:[local_port]

      Max-Forwards:

      Subject: Performance Test

      Content-Type: application/sdp

      Content-Length: [len]

      v=

      o=user1   IN IP[local_ip_type] [local_ip]

      s=-

      c=IN IP[local_ip_type] [local_ip]

      t=

      m=audio [media_port] RTP/AVP

      a=rtpmap: G729

    ]]>

  </send>

  <recv response=""

         optional="true">

    </recv>

  <!-- By adding rrs="true" (Record Route Sets), the route sets         -->

  <!-- are saved and used for following messages sent. Useful to test   -->

  <!-- against stateful SIP proxies/B2BUAs.                             -->

  <recv response="" rtd="true">

  </recv>

  <!-- Packet lost can be simulated in any send/recv message by         -->

  <!-- by adding the 'lost = "10"'. Value can be [-] percent.       -->

  <send>

      <![CDATA[

      ACK sip:[field1]@[remote_ip]:[remote_port] SIP/2.0

      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]

      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]

      To: sut <sip:[field1]@[remote_ip]:[remote_port]>[peer_tag_param]

      Call-ID: [call_id]

      CSeq:  ACK

      Contact: sip:sipp@[local_ip]:[local_port]

      Max-Forwards:

      Subject: Performance Test

      Content-Length:

    ]]>

  </send>

  <!-- Play a pre-recorded PCAP file (RTP stream)                       -->

  <nop>

      <action>

            <exec play_pcap_audio="pcap/g711a.pcap"/>

                </action>

                  </nop>

  <!-- Pause  seconds, which is approximately the duration of the      -->

  <!-- PCAP file                                                        -->

  <pause milliseconds=""/>

  <!--This delay can be customized by the -d command-line option      -->

  <!--or by adding a'milliseconds = "value" option here              -->

  <pause/>

  <!-- The 'crlf' option inserts a blank line in the statistics report. -->

  <send retrans="">

      <![CDATA[

      BYE sip:[field1]@[remote_ip]:[remote_port] SIP/2.0

      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]

      From: sipp <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]

      To: sut <sip:[field1]@[remote_ip]:[remote_port]>[peer_tag_param]

      Call-ID: [call_id]

      CSeq:  BYE

      Contact: sip:sipp@[local_ip]:[local_port]

      Max-Forwards:

      Subject: Performance Test

      Content-Length:

      ]]>

   </send>

  <recv response="" crlf="true">

    </recv>

  <!-- definition of the response time repartition table (unit is ms)   -->

  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>

  <!-- definition of the call length repartition table (unit is ms)     -->

  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>

</scenario>

SEQUENTIAL

;;[authentication username= password=]

./sipp -sf uac-media.xml -inf .csv -p  -i 192.168.106.141 -m  192.168.106.170 -l 

Last Error: Aborting call on UDP retransmission timeout for Call-ID '281...

------------------------------ Scenario Screen -------- [-]: Change Screen --

  Call-rate(length)   Port   Total-time  Total-calls  Remote-host

  10.0( ms)/.000s        313.57 s           192.168.106.170:(UDP)

   new calls during 0.952 s period       ms scheduler resolution

   calls (limit )                Peak was  calls, after  s

   Running,  Paused,  Woken up

   dead call msg (discarded)          out-of-call msg (discarded)

   open sockets                        

                                 Messages  Retrans   Timeout   Unexpected-Msg

      INVITE ---------->

          <----------

          <----------  E-RTD1

         ACK ---------->

       Pause [    :]

       Pause [      0ms]

         BYE ---------->

          <----------                                            

------------------------------ Test Terminated --------------------------------

SIP/sipp-00000cb7    @sipp:          Up      WaitMusicOnHold()

SIP/sipp-00000a18    @sipp:          Up      WaitMusicOnHold()

SIP/sipp-00000cb4    @sipp:          Up      WaitMusicOnHold()

SIP/sipp-00000a3e    @sipp:          Up      WaitMusicOnHold()

SIP/sipp-00000cb5    @sipp:          Up      WaitMusicOnHold()

SIP/sipp-00000a3d    @sipp:          Up      WaitMusicOnHold()

SIP/sipp-00000ab4    @sipp:          Up      WaitMusicOnHold()

SIP/sipp-00000cb2    @sipp:          Up      WaitMusicOnHold()

SIP/sipp-00000ab5    @sipp:          Up      WaitMusicOnHold()

SIP/sipp-00000cb3    @sipp:          Up      WaitMusicOnHold()

SIP/sipp-00000a3f    @sipp:          Up      WaitMusicOnHold()

SIP/sipp-00000ab6    @sipp:          Up      WaitMusicOnHold()

SIP/sipp-00000cb0    @sipp:          Up      WaitMusicOnHold()

SIP/sipp-00000a3a    @sipp:          Up      WaitMusicOnHold()

SIP/sipp-00000ab7    @sipp:          Up      WaitMusicOnHold()

SIP/sipp-00000cb1    @sipp:          Up      WaitMusicOnHold()

SIP/sipp-00000ab0    @sipp:          Up      WaitMusicOnHold()

 active channels

 active calls

 calls processed

[root@test asterisk]# asterisk -rx"core show channels"