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[Sipp-users] Unexpected Retransmissions.........for 200ok and BYE messages.
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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
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