1. IVR 支持多少路
2. 一对一通话, 支持多少路
3. 不同编解码的性能影响.
4. 通话中,录音, 支持多少路.
测试工具: sipp http://sipp.sourceforge.net/
辅助工具: Xlite
SIP rfc: http://www.ietf.org/rfc/rfc3261.txt
RTP for AV http://www.ietf.org/rfc/rfc3551.txt
环境:
CPU: xeon 5110
1.6G
*2 , 1 G MEM 物理机
Asterisk1.4.7
Asterisk 基本操作:
启动: safe_asterisk, 或者asterisk -vvvc
如果是后台启动, 连接监控: astersisk -r
关闭: 在控制栏输入 stop now
Asterisk 配置:
关注两个配置文件(/etc/asterisk):
sip.conf // sip 分机号设置
extensions.conf // dail plan 设置, 控制呼入后是什么动作
sip.conf 添加2000 个分机号, 以便模拟1000 人呼叫(呼叫,应答)
[1000]
type=friend
host=dynamic
context=incoming //和extensions.conf 中对应
canreinvite=no //如果设置为yes, 双方通话信息会直接进行, 而不通过asterisk. 设置成no,表示所有交互都通过Asterisk.
[1001]
type=friend
host=dynamic
context=incoming
canreinvite=no
extensions.conf 这里列举了多种呼叫计划, 包括IVR, 拨号通话, 通话录音等.
[incoming]
;play hello world forever
exten => _XXXX,1,answer()
exten => _XXXX,2,playback(hello-world)
exten => _XXXX,3,goto(OneToOne,_XXXX,1)
;[typetest]
;exten => 1111,1,Wait(2)
;exten => 1111,2,Record(/tmp/asterisk-recording:gsm)
;exten => 1111,3,Hangup
;exten => 1112,1,Wait(2)
;exten => 1112,n,Playback(/tmp/asterisk-recording)
;exten => 1112,n,Hangup
;[typetest2]
;exten => _XXXX,1,answer()
;exten => _XXXX,2,dial(sip/${EXTEN},10,r)
;[typetest3]
;exten => 999,1,answer()
;exten => 999,2,dial(sip/${EXTEN},10,r)
;exten => 999,1,Meetme(1234,i,123456)
;[OneToOne]
;exten => _XXXX,1,answer()
;exten => _XXXX,2,mixmonitor(test${EXTEN}.wav|av(0)V(0))
;exten => _XXXX,3,dial(sip/${EXTEN},10,r)
;exten => _XXXX,4,Hangup
;exten => _XXXX,3,Record(/tmp/asterisk-recording${EXTEN}:gsm)
;[IVR]
;exten => _XXXX,1,answer()
;exten => _XXXX,2,playback(hello-world)
;exten => _XXXX,3,goto(IVR,_XXXX,2)
Sipp 基本操作:
sipp 涉及三个文档(以呼入后,就不停播放IVR 语音):
*.bat 批处理命令,方便调用, 其中 -m 参数表示
sipp -sf a16.xml -inf a16.csv -p 5062 -m 200 -i 172.16.3.199 10.0.1.4:5060 -trace_err
*.xml // 具体的操作, 核心部分, 下篇具体介绍xml 文件
*.csv // xml 里面引用的参数, 以便发起不同呼叫
SEQUENTIAL
2001;1002;
2003;1003;
2200;1200;
xml 文件 . 本文草本出自狼山客人, 保留其相关注释.
<?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 2 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., -->
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
<!-- -->
<!-- Sipp default 'uac' scenario. -->
<!-- -->
<scenario name="Basic Sipstone UAC">
<!--狼山客人：下面这一块表示SIPp发送一个INVITE数据包到SIP server(Wavesplitter的MSP-16)-->
<send>
<![CDATA[
INVITE sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch];rport
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
To: [field1] <sip:[field1]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
Cseq: 1 INVITE
Contact: sip:[field0]@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
t=0 0
c=IN IP[media_ip_type] [media_ip]
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 G726/8000
]]>
</send>
<!--狼山客人：下面这一块表示SIPp在等待SIP server返回一个100的数据包-->
<recv response="100"> optional="true"
</recv>
<!--狼山客人：下面这一块表示SIPp在等待SIP server返回一个200的数据包，如果收到，说明ViVoice公司的VENUS NW800视频电话已经接听了，用户已经提起话筒-->
<recv response="200">
</recv>
<!--狼山客人：下面这一块表示SIPp开始通话-->
<!-- Packet lost can be simulated in any send/recv message by -->
<!-- by adding the 'lost = "10"'. Value can be [1-100] 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:[field0]@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[field1]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:[field0]@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<!--狼山客人：下面这一块表示SIPp开始发送语音RTP stream，在VENUS NW800视频电话上可以听到不知道哪国的老外的声音-->
<!-- Play a pre-recorded PCAP file (RTP stream) -->
<nop>
<action>
<exec play_pcap_audio="pcap/g711a.pcap"/>
</action>
</nop>
<!--狼山客人：暂停10秒钟(10000)，一小时， 3600000， 等待播放语音完毕-->
<pause milliseconds="300000"/>
<!--狼山客人：下面这一块发送BYE信号，这是挂断电话信号-->
<send retrans="500">
<![CDATA[
BYE sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port]
From: sipp <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
To: sip <sip:[field1]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
Cseq: 2 BYE
Contact: sip:[field0]@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<!--狼山客人：下面这一块表示挂断完毕-->
<recv response="200">
</recv>
</scenario>
里面的业务逻辑需要自己确认, 最好的方法是通过抓包(比如ethereal), 分析其工作流. 不同情况下, 返回的信息有所不同, 需要相应调整.
上面是个呼入后,播放IVR 的案例, 比较简单.
下面是register 的例子(要测试通话, 就要先注册分机号)
<?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 2 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., -->
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
<!-- -->
<!-- Sipp default 'branchc' scenario. -->
<!-- -->
<scenario name="Basic Sipstone UAC">
<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];rport
From: [field0] <sip:[field0]@[remote_ip]:[remote_port]>;tag=[call_number]
To: [field0] <sip:[field0]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: [cseq] REGISTER
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=z9hG4bK-d87543-717507386-1--d87543-;rport
Contact: <sip:[field0]@[local_ip]:[local_port]>
Expires: 1200
Max-Forwards: 70
User-Agent: eyeBeam release 3004t stamp 16741
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,NOTIFY,MESSAGE,SUBSCRIBE,INFO
Content-Length: 0
]]>
</send>
<recv response="100"> optional="true"
</recv>
<recv response="200" crlf="true">
</recv>
</recv>
</scenario>
Register 后, 先把被叫号码启动, 进入守护状态, 等到有电话呼叫(INVITE) 后, 震铃接通, 并发送数据. 产生通话效果.
<?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 2 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., -->
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
<!-- -->
<!-- Sipp default 'branchs' scenario. -->
<!-- -->
<scenario name="branch_server">
<!-- Set variable 3 if the ua is of the form ua2... -->
<recv request="INVITE" crlf="true">
</recv>
<!-- send 180 then trying if variable 3 is set -->
<send next="1">
<![CDATA[
SIP/2.0 180 Ringing
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
[last_Via:]
Contact: <sip:[field0]@[local_ip]:[local_port]>
Content-Length: 0
]]>
</send>
<label id="1"/>
<send retrans="500">
<![CDATA[
SIP/2.0 200 OK
[last_From:]
[last_To:];tag=800000[call_number]
[last_Call-ID:]
[last_CSeq:]
[last_Via:]
Contact: <sip:[field0]@[local_ip]:[local_port]>
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,NOTIFY,MESSAGE,SUBSCRIBE,INFO
Content-Type: application/sdp
Content-Length: 136
v=0
o=- 2372001 2372043 IN IP4 [local_ip]
s=eyeBeam
c=IN IP4 [local_ip]
t=0 0
c=IN IP4 [media_ip]
m=audio [media_port] RTP/AVP 0
a=fmtp:101 0-15
a=rtpmap:100 speex/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv
]]>
</send>
<recv request="ACK"
rtd="true"
crlf="true">
</recv>
<label id="2"/>
<nop>
<action>
<exec play_pcap_audio="pcap/g711a.pcap"/>
</action>
</nop>
<pause milliseconds="7000" next="2" />
<!-- Keep the call open for a while in case the 200 is lost to be -->
<!-- able to retransmit it if we receive the BYE again. -->
<!-- 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"/> -->
<ResponseTimeRepartition value="20, 40, 60, 80, 100, 200, 400, 800"/>
<CallLengthRepartition value="20, 100, 200, 1000, 2000, 10000, 20000"/>
</scenario>
被叫准备好以后, 主叫开始呼叫
<?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 2 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., -->
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
<!-- -->
<!-- Sipp 'uac' scenario with pcap (rtp) play -->
<!-- -->
<scenario name="UAC with media">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<send retrans="500">
<![CDATA[
INVITE sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch];rport
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
To: [field1] <sip:[field1]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
Cseq: 1 INVITE
Contact: sip:[field0]@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[local_ip_type] [local_ip]
t=0 0
m=audio [auto_media_port] RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16
]]>
</send>
<recv response="100" optional="true">
</recv>
<!--
<recv response="180" 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="200" rtd="true" crlf="true">
</recv>
<!-- Packet lost can be simulated in any send/recv message by -->
<!-- by adding the 'lost = "10"'. Value can be [1-100] 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:[field0]@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[field1]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:[field0]@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<!-- Play a pre-recorded PCAP file (RTP stream) -->
<label id="1"/>
<nop>
<action>
<exec play_pcap_audio="pcap/g711a.pcap"/>
</action>
</nop>
<pause milliseconds="7000" next="1" />
<!-- Play an out of band DTMF '1' -->
<nop>
<action>
<exec play_pcap_audio="pcap/dtmf_2833_1.pcap"/>
</action>
</nop>
<pause milliseconds="1000"/>
<!-- The 'crlf' option inserts a blank line in the statistics report. -->
<send retrans="500">
<![CDATA[
BYE sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port]
From: sipp <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
To: sip <sip:[field1]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
Cseq: 2 BYE
Contact: sip:[field0]@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<recv response="200" 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"/> -->
<ResponseTimeRepartition value="20, 40, 60, 80, 100, 200, 400, 800"/>
<CallLengthRepartition value="20, 100, 200, 1000, 2000, 10000, 20000"/>
</scenario>
编码对性能影响:
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[local_ip_type] [local_ip]
t=0 0
m=audio [auto_media_port] RTP/AVP 8
a=rtpmap:8 PCMA/8000 //语音发送, 修改PCMA 为G726 等, 不过这个需要asterisk server 支持
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16
通话录音性能:
extensions.conf 中选择通话中录音的方式就可以了. 录音文件保存在: /var/spool/asterisk/monitor
测试结果:
IVR: 500 路
一对一通话: 500 路
录音: 300 路
编解码转换对性能影响较大, 具体和编码方式相关.
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