Hello, I'm doing a stress test to a SIP proxy server to analyse the performance of several overload control algorithms I've implemented.
The network's topology I'm using is the following: on the same host, a Centrino Duo with 2 GB Ram, I'm running uas and uac, while on the second one (Pentium Three with 512 MB Ram), linked through a Gb crossover cable, I'm running my modified proxy server. I have attached SIPp xml scenario files and bash scripts used to start tests, where you can see the options I've set to run SIPp instances. I'm using Openser version 1.4 for my proxy, modified with the introduction of a queue for coming Invite messages, that are managed according to the overload control mechanism. The problem I'm running into during my tests is that SIPp uac can't keep generating the target call-rate in situations of overload. I mean that it can easily generate high call-rate when the proxy doesn't suffer from overload problems, but when openser comes to the overload condition, for very low rates too (incoming rate at which overload verifies depends on the tested algorithm), the real call-rate becomes very unsteady and meanly significantly lower than the nominal call-rate value I've set. The cpu usage oh the host running SIPp doesn't exceed 50% and the set limit for simultaneously calls (200000) is much higher than necessary. Is there anything wrong in what I've done or what else? Thanks in advance. Regards, Giuseppe
sipp_client.sh
Description: Bourne shell script
sipp_server.sh
Description: Bourne shell script
<?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" start_rtd="true">
<![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=[call_number]
To: sut <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="180" optional="true">
</recv>
<recv response="503" optional="true">
<action>
<exec int_cmd="stop_call"/>
</action>
</recv>
<recv response="200" rtd="true">
</recv>
<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=[call_number]
To: sut <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>
<send retrans="500" crlf="true">
<![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=[call_number]
To: sut <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" >
</recv>
<!--
definition of the response time repartition table (unit is ms)
-->
<ResponseTimeRepartition value="10, 50, 100, 150, 200, 500, 1000"/>
</scenario>
<?xml version="1.0" encoding="ISO-8859-1" ?> <!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="Basic UAS responder">
<recv request="INVITE" crlf="true">
</recv>
<send>
<![CDATA[
SIP/2.0 180 Ringing
[last_Via:]
[last_From:]
[last_To:];tag=[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=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
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 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>
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
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