I am running a scenario file(Attached Below) that issues a call, play's at 32
second PCAP file and hangs up. I have compiled the latest stable version of
libpcap from tcpdump.org and the latest stable version of sipp from
sourceforge. I am hoping someone can tell me if the this a performance
limitation or tuning on the OS that needs to be done. It seems like around the
time the system gets to around 400+ in progress calls the sipp goes sideways. I
am hoping it is just my ignorance on tuning the system to work with SIPP that
is causing it.
While attempting to drive load(run command attached below as well) I received
the following error. I am running as root on the system. The OS type is Red
Hat Enterprise Linux Server release 5.1. The system has 2GB of free memory and
2 x Intel Dual processor.
2009-11-04 16:18:19:117 1257373099.117813: Can't create raw socket
(need to run as root?).
Segmentation fault
This is odd since the running status screen reads: 3 open sockets
I have checked ulimits on the system and they are set to unlimited.
core file size (blocks, -c) 0
data seg size (kbytes, -d) unlimited
max nice (-e) 0
file size (blocks, -f) unlimited
pending signals (-i) 32763
max locked memory (kbytes, -l) 32
max memory size (kbytes, -m) unlimited
open files (-n) 1024
pipe size (512 bytes, -p) 8
POSIX message queues (bytes, -q) 819200
max rt priority (-r) 0
stack size (kbytes, -s) unlimited
cpu time (seconds, -t) unlimited
max user processes (-u) 32763
virtual memory (kbytes, -v) unlimited
file locks (-x) unlimited
Command:
sipp X.X.X.X -i X.X.X.X -r 40 -m 144000 -sf xml/sipt35seconepcap.xml -inf
101batch9.csv
In pcap /SIPP/pcap/36_8_secsEVRC0.pcap, npkts 1081
max pkt length 42
base port 8000
XML File
<scenario name="UAC">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<send retrans="750">
<![CDATA[
INVITE sip:[fiel...@[remote_ip]:[remote_port];user=phone;transport=udp
SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: [field2] <sip:[fiel...@[local_ip]:[local_port]>;tag=[call_number]
To: <sip:[fiel...@[remote_ip]:[remote_port];user=phone>
CSeq: 1 INVITE
Expires: 180
Route: <sip:[remote_ip];transport=udp;lr>
Min-SE: 90
Session-Expires: 1800
Supported: replaces, 100rel, timer
Accept: application/sdp, application/ISUP, multipart/mixed
Request-Disposition: fork, sequential
Allow: INVITE, BYE, REGISTER, ACK, OPTIONS, CANCEL, SUBSCRIBE, NOTIFY,
PRACK, INFO, REFER, UPDATE
MIME-Version: 1.0
Content-Length: [len]
Call-ID: [call_id]
P-Asserted-Identity: <sip:[fiel...@[local_ip]:[local_port];user=phone>
Privacy: none
Max-Forwards: 68
Diversion:
sip:[fiel...@[local_ip];reason=user-busy;counter=1;privacy="off<sip:[fiel...@[local_ip];reason=user-busy;counter=1;privacy=%22off>"
;screen="no"
Contact: <sip:[fiel...@[local_ip]:[local_port]>
Content-Type: multipart/mixed; boundary=Telica-boundary
--Telica-boundary
Content-Type: application/sdp
Content-Disposition: session; handling=required
v=0
o=- 3458920926 3458920926 IN IP4 [local_ip]
s=-
c=IN IP4 [local_ip]
t=0 0
m=audio [auto_media_port] RTP/AVP 96 0 101
a=ptime:20
a=rtpmap:96 EVRC0/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=fmtp:96 silencesupp=1 dtxmax=32 dtxmin=12 hangover=1
a=silenceSupp:on - - - -
--Telica-boundary
Content-Type: application/ISUP;version=ansi00
Content-Disposition: signal;handling=required
..`.
..
.......hv7'4
....#b ...=...S..
--Telica-boundary--
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="503" optional="true">
</recv>
<recv response="180" optional="true">
</recv>
<recv response="486" 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" crlf="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:[fiel...@[remote_ip]:[remote_port];user=phone SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
Call-ID: [call_id]
From: [field2] <sip:[fiel...@[local_ip]:[local_port]>;tag=[call_number]
To: <sip:[fiel...@[remote_ip]:[remote_port];user=phone>[peer_tag_param]
CSeq: 1 ACK
Contact: <sip:[fiel...@[local_ip]:[local_port]>
Max-Forwards: 70
Content-Length: [len]
]]>
</send>
<nop>
<action>
<exec play_pcap_audio="/SIPP/pcap/36_8_secsEVRC0.pcap"/>
</action>
</nop>
<!-- was 37500 but trying 36300-->
<pause milliseconds="36300"/>
<!-- The 'crlf' option inserts a blank line in the statistics report. -->
<send retrans="500">
<![CDATA[
BYE sip:[fiel...@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
Call-ID: [call_id]
From: [field2]
<sip:[fiel...@[local_ip]:[local_port];user=phone>;tag=[call_number]
To: <sip:[fiel...@[remote_ip]:[remote_port];user=phone>[peer_tag_param]
CSeq: 2 BYE
Supported: replaces, 100rel, timer
MIME-Version: 1.0
Content-Length: [len]
Reason: Q.850; cause=16;text="Normal call clearing"
Max-Forwards: 70
Content-Type: multipart/mixed; boundary=Telica-boundary
--Telica-boundary
Content-Type: application/ISUP; version=ansi00
Content-Disposition: signal; handling=required
......
--Telica-boundary--
]]>
</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>
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