From: Terrence R. \(trouse\) <tr...@ci...> - 2005-04-21 22:47:01
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Hello, I am using SIPp for Load/perf testing.... I am getting unexpected retransmissions on 200ok and BYE.....I have AUTH on and Record-Route via THREE proxies. Thanks In advance. [root@stim2 sipp]# ./sipp_fix 75.208.1.183 -sf uas_rr-on-no-bye-dssfp.xml -inf 919212.csv -ap 919212 -rsa 75.208.1.183:5060 -i 101.152.1.1 <?xml version="1.0" encoding="ISO-8859-1" ?> <!-- --> <!-- UAS - Basic Call to Proxy Server without Record Route --> <!-- The -d option may be used to pause between 180 and 200 --> <!-- --> <scenario name="Basic UAS responder"> <recv request="INVITE" rrs="true"> <action> <ereg regexp=".*" search_in="hdr" check_it="true" header="From:" assign_to="1"/> <ereg regexp=".*" search_in="hdr" check_it="true" header="To:" assign_to="2"/> <ereg regexp="919212...." search_in="hdr" check_it="true" header="To:" assign_to="6"/> </actions> </recv> <send> <![CDATA[ SIP/2.0 100 Trying [last_Via:] [last_From:] [last_To:];tag=[call_number] [last_Record-Route:] [last_Call-ID:] [last_CSeq:] Contact: <sip:[$6]@[local_ip]:[local_port];transport=[transport]> Content-Length: 0 ]]> </send> <send> <![CDATA[ SIP/2.0 180 Ringing [last_Via:] [last_From:] [last_To:];tag=[call_number] [last_Record-Route:] [last_Call-ID:] [last_CSeq:] Contact: <sip:[$6]@[local_ip]:[local_port];transport=[transport]> Content-Length: 0 ]]> </send> <pause/> <send retrans="500"> <![CDATA[ SIP/2.0 200 Ok [last_Via:] [last_From:] [last_To:] [last_Record-Route:] [last_Call-ID:] [last_CSeq:] Contact: <sip:[$6]@[local_ip]:[local_port]> Content-Type: application/sdp Content-Length: 255 v=0 o=Cisco-SIPUA 10872 26568 IN IP4 [local_ip] s=SIP Call c=IN IP4 [media_ip] t=0 0 m=audio [media_port] RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv ]]> </send> <recv request="ACK" optional="true"> </recv> <recv request="BYE"> </recv> <send retrans="500"> <![CDATA[ SIP/2.0 200 Ok [last_Via:] [last_From:] [last_To:] [last_Record-Route:] [last_Route:] [last_Call-ID:] [last_CSeq:] Contact: <sip:[$6]@[local_ip]:[local_port]> Content-Type: application/sdp Content-Length: 0 ]]> </send> <pause milliseconds="4000"/> <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> [root@stim1 sipp]# ./sipp_fix 75.208.1.183 -ap 703796 -sf uac_rr_on-sendbye-sfp.xml -inf 703-919.csv -i 101.151.1.1 -l 15000 <?xml version="1.0" encoding="ISO-8859-1" ?> <!-- --> <!-- UAC - Basic Call to Proxy Server without Record Route --> <!-- --> <scenario name="Basic Sipstone UAC"> <send retrans="500"> <![CDATA[ INVITE sip:[field2]@dssfp.pop1.com:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port] From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[field0] To: [field2] <sip:[field2]@dssfp.pop1.com:[remote_port]> Call-ID: [call_id] CSeq: 1 INVITE Contact: <sip:[field0]@[local_ip]:[local_port]> Max-Forwards: 5 Subject: Performance Test Content-Type: application/sdp Content-Length: 255 v=0 o=Cisco-SIPUA 10872 26568 IN IP4 [local_ip] s=SIP Call c=IN IP4 [media_ip] t=0 0 m=audio [media_port] RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ]]> </send> <recv response="100" optional="true"> </recv> <recv response="407" auth="true"> <action> <ereg regexp=".*" search_in="hdr" check_it="true" header="From:" assign_to="1"/> <ereg regexp=".*" search_in="hdr" check_it="true" header="To:" assign_to="2"/> </actions> </recv> <send> <![CDATA[ ACK sip:[field2]@dssfp.pop1.com SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port] To: [$2] From: [$1] Call-ID: [call_id] CSeq: 1 ACK Content-Length: 0 ]]> </send> <send retrans="500"> <![CDATA[ INVITE sip:[field2]@dssfp.pop1.com:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port] From: [field0] <sip:[field0]@dssfp.pop1.com:[local_port]>;tag=[field0] To: [field2] <sip:[field2]@dssfp.pop1.com:[remote_port]> Call-ID: [call_id] CSeq: 2 INVITE Contact: <sip:[field0]@[local_ip]:[local_port]> [field1] Max-Forwards: 25 Subject: Performance Test Content-Type: application/sdp Content-Length: 255 v=0 o=Cisco-SIPUA 10872 26568 IN IP4 [local_ip] s=SIP Call c=IN IP4 [media_ip] t=0 0 m=audio [media_port] RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ]]> </send> <recv response="100" optional="true"> </recv> <recv response="180" optional="true"> </recv> <recv response="200" rrs="true"> <action> <ereg regexp=".*" search_in="hdr" check_it="true" header="From:" assign_to="3"/> <ereg regexp=".*" search_in="hdr" check_it="true" header="To:" assign_to="4"/> </action> </recv> <send> <![CDATA[ ACK sip:[field2]@dssfp.pop1.com SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port] To: [$4] From: [$3] Call-ID: [call_id] CSeq: 2 ACK Content-Length: 0 ]]> </send> <pause/> <send> <![CDATA[ BYE sip:[field2]@dssfp.pop1.com;lr SIP/2.0 Via: SIP/2.0/UDP [local_ip]:[local_port] From: [$3] To: [$4] [last_Call-ID:] Cseq: 10 BYE Contact: <sip:[field0]@[local_ip]:[local_port];transport=[transport]> Max-Forwards: 25 Content-Length: 0 [routes] ]]> </send> <recv response="407" auth="true"> </recv> <send retrans="500"> <![CDATA[ BYE sip:[field2]@dssfp.pop1.com;lr SIP/2.0 Via: SIP/2.0/UDP [local_ip]:[local_port] From: [$3] To: [$4] [last_Call-ID:] Cseq: 11 BYE Contact: <sip:[field0]@[local_ip]:[local_port];transport=[transport]> Max-Forwards: 25 Content-Length: 0 [routes] [field1] ]]> </send> <pause milliseconds="1000"/> <!-- 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> ------------------------------ Scenario Screen -------- [1-4]: Change Screen -- Call-rate(length) Port Total-time Total-calls Remote-host 10.0(0 ms)/1.000s 5060 40.89 s 408 75.208.1.183:5060(UDP) 8 new calls during 0.855 s period 2 ms scheduler resolution 16 concurrent calls (limit 15000) Peak was 23 calls, after 37 s 194 out-of-call msg (discarded) 1 open sockets Messages Retrans Timeout Unexpected-Msg INVITE ----------> 408 0 0 100 <---------- 404 0 0 407 <---------- 407 0 0 ACK ----------> 407 0 INVITE ----------> 407 0 0 100 <---------- 407 0 7 180 <---------- 333 0 0 200 <---------- 394 0 0 ACK ----------> 394 0 [ 0 ms] BYE ----------> 394 0 407 <---------- 385 0 0 BYE ----------> 385 376 0 [ 1000 ms] ------------------------------ Test Terminated -------------------------------- ----------------------------- Statistics Screen ------- [1-4]: Change Screen -- Start Time | 2005-04-21 18:31:23 Last Reset Time | 2005-04-21 18:32:03 Current Time | 2005-04-21 18:32:04 -------------------------+---------------------------+---------------------- ---- Counter Name | Periodic value | Cumulative value -------------------------+---------------------------+---------------------- ---- Elapsed Time | 00:00:00:855 | 00:00:40:913 Call Rate | 9.357 cps | 9.972 cps -------------------------+---------------------------+---------------------- ---- Incoming call created | 0 | 0 OutGoing call created | 8 | 408 Total Call created | | 408 Current Call | 16 | -------------------------+---------------------------+---------------------- ---- Successful call | 7 | 371 Failed call | 5 | 21 -------------------------+---------------------------+---------------------- ---- Response Time | 00:00:00:000 | 00:00:00:000 Call Length | 00:00:00:771 | 00:00:01:206 ------------------------------ Test Terminated -------------------------------- ------------------------------ Scenario Screen -------- [1-4]: Change Screen -- Port Total-time Total-calls Transport 5060 42.69 s 1621 UDP 49 new calls during 0.678 s period 2 ms scheduler resolution 63 concurrent calls Peak was 66 calls, after 42 s 1 open sockets Messages Retrans Timeout Unexpected-Msg ----------> INVITE 395 0 0 <---------- 100 395 0 <---------- 180 395 0 [ 0 ms] <---------- 200 395 93 0 ----------> ACK 380 0 0 ----------> BYE 384 1058 0 <---------- 200 368 2111 0 [ 4000 ms] ------------------------------ Test Terminated -------------------------------- ----------------------------- Statistics Screen ------- [1-4]: Change Screen -- Start Time | 2005-04-21 18:31:20 Last Reset Time | 2005-04-21 18:32:02 Current Time | 2005-04-21 18:32:03 -------------------------+---------------------------+---------------------- ---- Counter Name | Periodic value | Cumulative value -------------------------+---------------------------+---------------------- ---- Elapsed Time | 00:00:00:676 | 00:00:42:714 Call Rate | 72.485 cps | 37.950 cps -------------------------+---------------------------+---------------------- ---- Incoming call created | 49 | 1621 OutGoing call created | 0 | 0 Total Call created | | 1621 Current Call | 63 | -------------------------+---------------------------+---------------------- ---- Successful call | 7 | 332 Failed call | 40 | 1226 -------------------------+---------------------------+---------------------- ---- Response Time | 00:00:00:000 | 00:00:00:000 Call Length | 00:00:00:629 | 00:00:00:902 ------------------------------ Test Terminated -------------------------------- Terrence Rouse Technical Marketing Engineer, Voice Systems Marketing |