测试环境
asterisk 192.168.106.170
版本astrisk1.8
sipp 192.168.106.141
sipp版本3.3
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
asterisk配置
sip.conf
[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
extension.conf
[general] static=yes writeprotect=no clearglobalvars=no [globals] [sipp] exten => 7777,1,Answer exten => 7777,2,SetMusicOnHold(default) exten => 7777,3,WaitMusicOnHold(2000) exten => 7777,4,Hangup
sipp客户端配置
uac-media.xml
<?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="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="500"> <![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: 1 INVITE Contact: sip:sipp@[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 [media_port] RTP/AVP 0 a=rtpmap:18 G729 ]]> </send> <recv response="100" 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"> </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:sipp@[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:sipp@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Length: 0 ]]> </send> <!-- Play a pre-recorded PCAP file (RTP stream) --> <nop> <action> <exec play_pcap_audio="pcap/g711a.pcap"/> </action> </nop> <!-- Pause 800 seconds, which is approximately the duration of the --> <!-- PCAP file --> <pause milliseconds="800000"/> <!--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="500"> <![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: 2 BYE Contact: sip:sipp@[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"/> </scenario>
44.csv
SEQUENTIAL 4444;7777;[authentication username=4444 password=8888]
之后执行命令
./sipp -sf uac-media.xml -inf 44.csv -p 5060 -i 192.168.106.141 -m 100000 192.168.106.170 -l 2000
显示
Last Error: Aborting call on UDP retransmission timeout for Call-ID '281... ------------------------------ Scenario Screen -------- [1-9]: Change Screen -- Call-rate(length) Port Total-time Total-calls Remote-host 10.0(0 ms)/1.000s 5060 313.57 s 3135 192.168.106.170:5060(UDP) 9 new calls during 0.952 s period 0 ms scheduler resolution 1322 calls (limit 2000) Peak was 1332 calls, after 300 s 1 Running, 1659 Paused, 62 Woken up 1702 dead call msg (discarded) 1444 out-of-call msg (discarded) 3 open sockets Messages Retrans Timeout Unexpected-Msg INVITE ----------> 3135 10823 1680 100 <---------- 1140 0 0 0 200 <---------- E-RTD1 1140 1803 0 0 ACK ----------> 1140 1803 Pause [ 13:20] 1140 133 Pause [ 0ms] 0 0 BYE ----------> 0 0 0 200 <---------- 0 0 0 0 ------------------------------ Test Terminated --------------------------------
asterisk服务器显示
SIP/sipp-00000cb7 7777@sipp:3 Up WaitMusicOnHold(2000) SIP/sipp-00000a18 7777@sipp:3 Up WaitMusicOnHold(2000) SIP/sipp-00000cb4 7777@sipp:3 Up WaitMusicOnHold(2000) SIP/sipp-00000a3e 7777@sipp:3 Up WaitMusicOnHold(2000) SIP/sipp-00000cb5 7777@sipp:3 Up WaitMusicOnHold(2000) SIP/sipp-00000a3d 7777@sipp:3 Up WaitMusicOnHold(2000) SIP/sipp-00000ab4 7777@sipp:3 Up WaitMusicOnHold(2000) SIP/sipp-00000cb2 7777@sipp:3 Up WaitMusicOnHold(2000) SIP/sipp-00000ab5 7777@sipp:3 Up WaitMusicOnHold(2000) SIP/sipp-00000cb3 7777@sipp:3 Up WaitMusicOnHold(2000) SIP/sipp-00000a3f 7777@sipp:3 Up WaitMusicOnHold(2000) SIP/sipp-00000ab6 7777@sipp:3 Up WaitMusicOnHold(2000) SIP/sipp-00000cb0 7777@sipp:3 Up WaitMusicOnHold(2000) SIP/sipp-00000a3a 7777@sipp:3 Up WaitMusicOnHold(2000) SIP/sipp-00000ab7 7777@sipp:3 Up WaitMusicOnHold(2000) SIP/sipp-00000cb1 7777@sipp:3 Up WaitMusicOnHold(2000) SIP/sipp-00000ab0 7777@sipp:3 Up WaitMusicOnHold(2000) 1000 active channels 1000 active calls 3783 calls processed [root@test asterisk]# asterisk -rx"core show channels"
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Can't create thread to send RTP packets，翻译过来就是“无法创建线程发送RTP数据包”
大概知道是需要设置ulimit ，但设置好像没有生效。在龙舟兄弟的指导与提醒下，索性直接ulimit -s unlimited，设置为不限制了。
如此之后，rtp就不受限制了。
xml实例文件