[SIPForum-discussion] SIP Load testing tool
Stewart Bunce
Stewart.Bunce at telsis.com
Fri Jun 19 09:38:45 UTC 2009
Hi,
You do not appear to be offering the telephone-event type in the media line of the INVITE message i.e. your m = line should read:
m=audio [auto_media_port] RTP/AVP 8 101
Hope this helps
Stew
-----Original Message-----
From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org]On Behalf Of Deb, Nihar Ranjan
Sent: 17 June 2009 16:39
To: discussion at sipforum.org
Subject: Re: [SIPForum-discussion] SIP Load testing tool
Hi All,
I am using following xml codes to send DTMF to my SIP application and I am using the command line command "C:\Program Files\Sipp_3.1>sipp 10.8.2.39 -sf test.xml -i 10.8.2.144 -mi 10.8.2.14
4 -s 3990 -l 1" to start SIPp. From the ethereal trace I can see DTMF is sent to the destination IP address but my SIP application is not detecting the DTMF.Is there anything I am missing please suggest.Normally if I am calling from any phone dtmf is working fine
I am running SIPp in windows platform and using RFC2833 DTMF mode.
<?xml version="1.0" encoding="ISO-8859-1" ?>
<scenario name="UAC with media">
<!--
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=[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[local_ip_type] [local_ip]
t=0 0
m=audio [auto_media_port] RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="180" 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:[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>
<pause milliseconds="8000"/>
<!--
Play an out of band DTMF '1'
-->
<nop>
<action>
<exec play_pcap_audio="pcap/dtmf_2833_1.pcap"/>
</action>
</nop>
<nop>
<action>
<exec play_pcap_audio="pcap/dtmf_2833_pound.pcap"/>
</action>
</nop>
<pause milliseconds="9000"/>
<!--
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=[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" 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>
Regards,
Nihar Deb
-----Original Message-----
From: Allen, Toby Edward Gedis (Toby) [mailto:allen at avaya.com]
Sent: Wednesday, June 17, 2009 6:24 AM
To: Deb, Nihar Ranjan
Subject: RE: [SIPForum-discussion] SIP Load testing tool
Use SIPp
Toby Allen | AVTS - Software Engineer | Avaya | +612 9352 8699| allen at avaya.com
[ Please consider the environment before printing this email ]
_____
From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Deb, Nihar Ranjan
Sent: Wednesday, 17 June 2009 1:57 AM
To: discussion at sipforum.org
Subject: [SIPForum-discussion] SIP Load testing tool
Hi All,
I need to traffic test a SIP based application. For that I need to configure the testing tool so that
1. It can send INVITE packets at a regular interval
2. It can send DTMF at a regular interval to the application as I will configure
3. It will send BYE after completion of one cycle
4. It should be able to run this kinds of more than 80 - 90 cycles same time.
Can anyone please suggest me any SIP load testing tool like this.
Thanks in advance.
Regards,
Nihar Deb
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