Hi All,
I am creating a scenario where SIPP will call to an extension and that
extension will get answered manually and trying to play a media file which
is in .wav format
But while executing the scenario,
./sipp 192.168.20.61 -sf Invite.xml -inf Invite.txt -m 1 -i 192.168.20.63
getting this error as below
"Unknown action: action".
192.168.20.61 is the server IP
192.168.20.63 is the local machine ip where SIPP is installed
I am attaching here the XML file.
Please help me to solve this.
--
*Thanks & Regards vinod*
<?xml version="1.0" encoding="ISO-8859-2" ?>
<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">
<![CDATA[
INVITE sip:[field0]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:101@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[field0]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:101@[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="401" auth="true"/>
<send>
<![CDATA[
ACK sip:[field0]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-1]
From: sipp <sip:101@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[field0]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:101@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<send retrans="500">
<![CDATA[
INVITE sip:[field0]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:101@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[field0]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 2 INVITE
Contact: sip:101@[local_ip]:[local_port]
[field2]
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="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"></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:[field0]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
From: sipp <sip:101@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[field0]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 ACK
Contact: sip:101@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<!--
This delay can be customized by the -d command-line option
-->
<!--
or by adding a 'milliseconds = "value"' option here.
-->
<nop>
<action>
<exec rtp_stream="manolo_camp-morning_coffee.wav"/>
<action/>
<nop/>
<!--<exec play_pcap_audio="pcap/dtmf_2833_pound.pcap"/>
-->
<!--
The 'crlf' option inserts a blank line in the statistics report.
-->
<send retrans="500">
<![CDATA[
BYE sip:[field0]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:101@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[field0]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 BYE
Contact: sip:101@[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>
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