南山做网站公司在哪里,做网站现在可以挣钱吗,店铺头像logo免费生成,湖南平台网站建设公司SIPP是针对SIP协议的一个性能测试的命令行工具#xff0c;可以动态显示测试的统计信息#xff08;如呼叫速率、延时、消息统计等#xff09;。用户可以通过XML场景配置文件#xff0c;自定义模拟各种UAC/UAS测试场景的信令交互流程#xff0c;可以被用来测试IP话机、SIP代… SIPP是针对SIP协议的一个性能测试的命令行工具可以动态显示测试的统计信息如呼叫速率、延时、消息统计等。用户可以通过XML场景配置文件自定义模拟各种UAC/UAS测试场景的信令交互流程可以被用来测试IP话机、SIP代理、SIP PBX等SIP设备。
一、参考资料
官方网站https://sipp.sourceforge.net/ 参考文档https://sipp.sourceforge.net/doc/reference.pdf
二、安装
linux平台
sudo apt install sip-tester注也可以下载源码编译安装
三、使用
3.1 查看用法
# sipp -hUsage:sipp remote_host[:remote_port] [options]Example:Run SIPp with embedded server (uas) scenario:./sipp -sn uasOn the same host, run SIPp with embedded client (uac) scenario:./sipp -sn uac 127.0.0.1Available options:*** Scenario file options:-sd : Dumps a default scenario (embedded in the SIPp executable)-sf : Loads an alternate XML scenario file. To learn more about XML scenariosyntax, use the -sd option to dump embedded scenarios.They contain all the necessary help.
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..................................3.2 常用参数说明
-v
作用显示版本信息
-bg
作用后台模式运行
-sd
作用输出SIPP内嵌的默认场景
sipp -sd uac/uas/uac_pcap部分参数说明
uac: uac场景uas: uas场景uac_pcap: uac带媒体场景
将场景内容重定向到文件
sipp -sd uas uas.xml-sf
作用加载指定的场景文件
sipp -sf uas.xml-sn uac/uas
作用使用默认的内置场景文件
# sipp -sn uas# sipp -sn uac-t u1/un/t1/tn
作用设置传输方式 部分参数说明 u1: 使用一个UDPun: 一个呼叫用一个UDPt1: 使用一个TCPtn: 一个呼叫使用一个TCP-i
作用设置本地的IP地址如contact,via,from
sip -i 192.168.1.100-p
作用设置本地的端口
sip -p 5060-bind_local
作用绑定本地IP地址
-ci
作用本地控制IP
-cp
作用本地控制PORT
3.3 场景文件 可以从在线参考文档https://sipp.sourceforge.net/doc/reference.pdf) 中的链接获取一些内置的测试场景的xml模板手动改变其中的一些参数或流程形成目标场景文件。
uac.xml:
scenario nameBasic Sipstone UAC!-- In client mode (sipp placing calls), the Call-ID MUST be --!-- generated by sipp. To do so, use [call_id] keyword. --send retrans500![CDATA[INVITE sip:[service][remote_ip]:[remote_port] SIP/2.0Via: SIP/2.0/[transport] [local_ip]:[local_port];branch[branch]From: sipp sip:sipp[local_ip]:[local_port];tag[pid]SIPpTag00[call_number]To: [service] sip:[service][remote_ip]:[remote_port]Call-ID: [call_id]CSeq: 1 INVITEContact: sip:sipp[local_ip]:[local_port]Max-Forwards: 70Subject: Performance TestContent-Type: application/sdpContent-Length: [len]v0ouser1 53655765 2353687637 IN IP[local_ip_type] [local_ip]s-cIN IP[media_ip_type] [media_ip]t0 0maudio [media_port] RTP/AVP 0artpmap:0 PCMU/8000]]/sendrecv response100optionaltrue/recvrecv response180 optionaltrue/recvrecv response183 optionaltrue/recv!-- By adding rrstrue (Record Route Sets), the route sets --!-- are saved and used for following messages sent. Useful to test --!-- against stateful SIP proxies/B2BUAs. --recv response200 rtdtrue/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:[service][remote_ip]:[remote_port] SIP/2.0Via: SIP/2.0/[transport] [local_ip]:[local_port];branch[branch]From: sipp sip:sipp[local_ip]:[local_port];tag[pid]SIPpTag00[call_number]To: [service] sip:[service][remote_ip]:[remote_port][peer_tag_param]Call-ID: [call_id]CSeq: 1 ACKContact: sip:sipp[local_ip]:[local_port]Max-Forwards: 70Subject: Performance TestContent-Length: 0]]/send!-- 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 retrans500![CDATA[BYE sip:[service][remote_ip]:[remote_port] SIP/2.0Via: SIP/2.0/[transport] [local_ip]:[local_port];branch[branch]From: sipp sip:sipp[local_ip]:[local_port];tag[pid]SIPpTag00[call_number]To: [service] sip:[service][remote_ip]:[remote_port][peer_tag_param]Call-ID: [call_id]CSeq: 2 BYEContact: sip:sipp[local_ip]:[local_port]Max-Forwards: 70Subject: Performance TestContent-Length: 0]]/sendrecv response200 crlftrue/recv!-- definition of the response time repartition table (unit is ms) --ResponseTimeRepartition value10, 20, 30, 40, 50, 100, 150, 200/!-- definition of the call length repartition table (unit is ms) --CallLengthRepartition value10, 50, 100, 500, 1000, 5000, 10000//scenariouas.xml
scenario nameBasic UAS responder!-- By adding rrstrue (Record Route Sets), the route sets --!-- are saved and used for following messages sent. Useful to test --!-- against stateful SIP proxies/B2BUAs. --!-- Adding ignoresdptrue here would ignore the SDP data: that --!-- can be useful if you want to reject reINVITEs and keep the --!-- media stream flowing. --recv requestINVITE crlftrue/recv!-- The [last_*] keyword is replaced automatically by the --!-- specified header if it was present in the last message received --!-- (except if it was a retransmission). If the header was not --!-- present or if no message has been received, the [last_*] --!-- keyword is discarded, and all bytes until the end of the line --!-- are also discarded. --!-- --!-- If the specified header was present several times in the --!-- message, all occurrences are concatenated (CRLF separated) --!-- to be used in place of the [last_*] keyword. --send![CDATA[SIP/2.0 180 Ringing[last_Via:][last_From:][last_To:];tag[pid]SIPpTag01[call_number][last_Call-ID:][last_CSeq:]Contact: sip:[local_ip]:[local_port];transport[transport]Content-Length: 0]]/sendsend retrans500![CDATA[SIP/2.0 200 OK[last_Via:][last_From:][last_To:];tag[pid]SIPpTag01[call_number][last_Call-ID:][last_CSeq:]Contact: sip:[local_ip]:[local_port];transport[transport]Content-Type: application/sdpContent-Length: [len]v0ouser1 53655765 2353687637 IN IP[local_ip_type] [local_ip]s-cIN IP[media_ip_type] [media_ip]t0 0maudio [media_port] RTP/AVP 0artpmap:0 PCMU/8000]]/sendrecv requestACKoptionaltruertdtruecrlftrue/recvrecv requestBYE/recvsend![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!-- 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. --timewait milliseconds4000/!-- definition of the response time repartition table (unit is ms) --ResponseTimeRepartition value10, 20, 30, 40, 50, 100, 150, 200/!-- definition of the call length repartition table (unit is ms) --CallLengthRepartition value10, 50, 100, 500, 1000, 5000, 10000//scenario3.4 示例
3.4.1 运行内置UAS/UAC
# sipp -sn uas# sipp -sn uac3.4.2 指定自定义场景文件
sipp -sf my_uas.xml3.4.3 指定本地IP/端口
sipp -sf uas.xml -i 192.168.1.200 50603.5 测试
192.168.1.16终端呼叫192.168.1.200
测试结果
# sipp -sn uas 192.168.1.200
Resolving remote host 192.168.1.200... Done.
------------------------------ Scenario Screen -------- [1-9]: Change Screen --Port Total-time Total-calls Transport5060 30.29 s 1 UDP0 new calls during 0.172 s period 1 ms scheduler resolution 0 calls Peak was 1 calls, after 10 s0 Running, 3 Paused, 0 Woken up0 dead call msg (discarded) 3 open sockets 0/0/0 UDP errors (send/recv/cong)0 Total RTP pckts sent 0.000 last period RTP rate (kB/s)Messages Retrans Timeout Unexpected-Msg---------- INVITE 1 0 0 0 ---------- 180 1 0 ---------- 200 1 0 0 ---------- ACK E-RTD1 1 0 0 0 ---------- BYE 1 0 0 0 ---------- 200 1 0 [ 4000ms] Pause 1 0 点击ctrlC结束sipp打印结果
------------------------------ Test Terminated --------------------------------
----------------------------- Statistics Screen ------- [1-9]: Change Screen --Start Time | 2024-03-15 15:01:44.429203 1710486104.429203 Last Reset Time | 2024-03-15 15:02:14.553107 1710486134.553107 Current Time | 2024-03-15 15:02:14.731231 1710486134.731231
------------------------------------------------------------------------------Counter Name | Periodic value | Cumulative value
------------------------------------------------------------------------------Elapsed Time | 00:00:00:178000 | 00:00:00:178000 Call Rate | 0.000 cps | 0.033 cps
------------------------------------------------------------------------------Incoming calls created | 0 | 1 Outgoing calls created | 0 | 0 Total Calls created | | 1 Current Calls | 0 |
------------------------------------------------------------------------------Successful call | 0 | 1 Failed call | 0 | 0
------------------------------------------------------------------------------Response Time 1 | 00:00:00:000000 | 00:00:00:000000 Call Length | 00:00:00:000000 | 00:00:00:000000
------------------------------ Test Terminated --------------------------------网络抓包
四、关键说明
一般用于性能、压力、稳定性测试可以模拟一些异常或客户现场的信令交互流程用于复现分析问题场景文件编写人员既要熟悉SIP协议又要熟悉SIPP的相关参数应用因此实际上很少使用SIPP作为日常测试手段
