我们先学习 uac 的xml,输入命令./sipp -sd uac 得到如下内容(部分注释被删除了):
<?xml version="1.0" encoding="ISO-8859-1" ?> <!DOCTYPE scenario SYSTEM "sipp.dtd"> <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/2.0 Via: 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 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[media_ip_type] [media_ip] t=0 0 m=audio [media_port] RTP/AVP 0 a=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 adding the 'lost = "10"'. Value can be [1-100] percent. --> <send> <![CDATA[ ACK sip:[service]@[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=[pid]SIPpTag00[call_number] To: [service] <sip:[service]@[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> <!-- 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/2.0 Via: 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 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>