SIP调试之SIPP测试工具

本文主要是介绍SIP调试之SIPP测试工具，希望对大家解决编程问题提供一定的参考价值，需要的开发者们随着小编来一起学习吧！

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 -h


Usage: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 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:[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]v=0o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]s=-c=IN IP[media_ip_type] [media_ip]t=0 0m=audio [media_port] RTP/AVP 0a=rtpmap:0 PCMU/8000]]></send><recv response="100"optional="true"></recv><recv response="180" optional="true"></recv><recv response="183" 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:[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 retrans="500"><![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]]></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>

uas.xml：

<scenario name="Basic UAS responder"><!-- 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.                             --><!-- Adding ignoresdp="true" here would ignore the SDP data: that     --><!-- can be useful if you want to reject reINVITEs and keep the       --><!-- media stream flowing.                                            --><recv request="INVITE" crlf="true"></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]]></send><send retrans="500"><![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]v=0o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]s=-c=IN IP[media_ip_type] [media_ip]t=0 0m=audio [media_port] RTP/AVP 0a=rtpmap:0 PCMU/8000]]></send><recv request="ACK"optional="true"rtd="true"crlf="true"></recv><recv request="BYE"></recv><send><![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 milliseconds="4000"/><!-- 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>

3.4 示例

3.4.1 运行内置UAS/UAC

# sipp -sn uas

# sipp -sn uac

3.4.2 指定自定义场景文件

sipp  -sf my_uas.xml

3.4.3 指定本地IP/端口

sipp  -sf uas.xml -i 192.168.1.200 5060

3.5 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ctrl+C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 --------------------------------