Hello,
I have an XML scenario which works perfectly for establishing a single phone
call through an IVR system. When attempting to establish two or more
simultaneous calls, the scenario eventually fails with a "dead call msg" error.
Is it possible that I cannot re-use the same media ports in the XML scenario
file? What else am I missing?
The command I use to generate the call is "sipp -i 10.x.x.x -s 4000 10.x.x.x -l
2 -r 2 -rp 2s -m 2 -sf webex.xml"
I have attached the current XML scenario, if someone could take a peek that
would be appreciated.
Rgds
Brett
<?xml version="1.0" encoding="ISO-8859-1" ?>
<scenario name="UAC with media">
<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=[pid]SIPpTag09[call_number]
To: [service] <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=root 1821 1821 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 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="180" optional="true">
</recv>
<recv response="200" rtd="true" crlf="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=[pid]SIPpTag09[call_number]
To: [service] <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: [len]
]]>
</send>
<pause milliseconds="3500"/>
<nop>
<action>
<exec play_dyn_pcap_audio="pcap/dtmf_2833_9.pcap"/>
</action>
</nop>
<pause milliseconds="600"/>
<nop>
<action>
<exec play_dyn_pcap_audio="pcap/dtmf_2833_9.pcap"/>
</action>
</nop>
<pause milliseconds="600"/>
<nop>
<action>
<exec play_dyn_pcap_audio="pcap/dtmf_2833_5.pcap"/>
</action>
</nop>
<pause milliseconds="600"/>
<nop>
<action>
<exec play_dyn_pcap_audio="pcap/dtmf_2833_0.pcap"/>
</action>
</nop>
<pause milliseconds="600"/>
<nop>
<action>
<exec play_dyn_pcap_audio="pcap/dtmf_2833_7.pcap"/>
</action>
</nop>
<pause milliseconds="600"/>
<nop>
<action>
<exec play_dyn_pcap_audio="pcap/dtmf_2833_5.pcap"/>
</action>
</nop>
<pause milliseconds="600"/>
<nop>
<action>
<exec play_dyn_pcap_audio="pcap/dtmf_2833_2.pcap"/>
</action>
</nop>
<pause milliseconds="600"/>
<nop>
<action>
<exec play_dyn_pcap_audio="pcap/dtmf_2833_0.pcap"/>
</action>
</nop>
<pause milliseconds="600"/>
<nop>
<action>
<exec play_dyn_pcap_audio="pcap/dtmf_2833_5.pcap"/>
</action>
</nop>
<pause milliseconds="600"/>
<nop>
<action>
<exec play_dyn_pcap_audio="pcap/dtmf_2833_hash.pcap"/>
</action>
</nop>
<recv request="INVITE">
<action>
<log message="Receive 1st INVITE from CUCM"/>
</action>
</recv>
<send>
<![CDATA[
SIP/2.0 100 Trying
[last_Via:]
[last_From:]
[last_To:]
[last_CSeq:]
[last_Call-ID:]
Content-Length: [len]
]]>
</send>
<send retrans="500">
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[pid]SIPpTag01[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=root 1821 1821 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 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
]]>
</send>
<recv request="ACK" rtd="true" crlf="true">
</recv>
<recv request="INVITE">
<action>
<log message="Receive 2nd INVITE from CUCM"/>
</action>
</recv>
<send>
<![CDATA[
SIP/2.0 100 Trying
[last_Via:]
[last_From:]
[last_To:]
[last_CSeq:]
[last_Call-ID:]
Content-Length: [len]
]]>
</send>
<send retrans="500">
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[pid]SIPpTag01[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=root 1821 1821 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 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
]]>
</send>
<recv request="ACK" rtd="true" crlf="true">
</recv>
<pause milliseconds="8000"/>
<nop>
<action>
<exec play_dyn_pcap_audio="pcap/dtmf_2833_hash.pcap"/>
</action>
</nop>
<pause milliseconds="8000"/>
<nop>
<action>
<exec play_dyn_pcap_audio="pcap/slt.pcap"/>
</action>
</nop>
<pause milliseconds="1000"/>
<nop>
<action>
<exec play_dyn_pcap_audio="pcap/dtmf_2833_hash.pcap"/>
</action>
</nop>
<pause milliseconds="30000"/>
<send>
<![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=[pid]SIPpTag09[call_number]
To: [service] <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: [len]
]]>
</send>
<recv response="200" crlf="true">
</recv>
</scenario>
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