You can not select more than 25 topics
Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.
136 lines
3.8 KiB
136 lines
3.8 KiB
<?xml version="1.0" encoding="ISO-8859-1" ?>
|
|
|
|
<scenario name="Sipwise NGCP Benchmark UAC Caller">
|
|
<send start_rtd="1" start_rtd="2">
|
|
<![CDATA[
|
|
|
|
INVITE sip:[field0 file="callee.csv" line=0]@[field3 file="callee.csv" line=0] SIP/2.0
|
|
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
|
|
From: <sip:[field0 file="caller.csv"]@[field2 file="caller.csv"]>;tag=[pid]SIPpTag00[call_number]
|
|
To: <sip:[field0 file="callee.csv" line=0]@[field3 file="callee.csv" line=0]>
|
|
Call-ID: NGCP%[field4 file="callee.csv" line=0]%///[call_id]
|
|
CSeq: [cseq] INVITE
|
|
Contact: <sip:[field0 file="caller.csv"]@[local_ip]:[local_port];transport=[transport]>
|
|
Max-Forwards: 70
|
|
Content-Type: application/sdp
|
|
Content-Length: [len]
|
|
|
|
v=0
|
|
o=root 1100779239 1100779239 IN IP[local_ip_type] [local_ip]
|
|
s=Asterisk PBX 1.8.10.1
|
|
c=IN IP[media_ip_type] [media_ip]
|
|
t=0 0
|
|
m=audio [media_port] RTP/AVP 8 0 101
|
|
a=rtpmap:8 PCMA/8000
|
|
a=rtpmap:0 PCMU/8000
|
|
a=rtpmap:101 telephone-event/8000
|
|
a=fmtp:101 0-16
|
|
a=silenceSupp:off - - - -
|
|
a=ptime:20
|
|
a=sendrecv
|
|
|
|
]]>
|
|
</send>
|
|
|
|
<recv response="100" optional="true"/>
|
|
<recv response="180" optional="true"/>
|
|
<recv response="183" optional="true"/>
|
|
<recv response="200" rrs="true" rtd="3"/>
|
|
|
|
<send>
|
|
<![CDATA[
|
|
|
|
ACK [next_url] SIP/2.0
|
|
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
|
|
From: <sip:[field0 file="caller.csv"]@[field2 file="caller.csv"]>;tag=[pid]SIPpTag00[call_number]
|
|
To: <sip:[field0 file="callee.csv" line=0]@[field3 file="callee.csv" line=0]>[peer_tag_param]
|
|
Call-ID: NGCP%[field4 file="callee.csv" line=0]%///[call_id]
|
|
[routes]
|
|
CSeq: [cseq] ACK
|
|
Contact: <sip:[field0 file="caller.csv"]@[local_ip]:[local_port];transport=[transport]>
|
|
Max-Forwards: 70
|
|
Content-Length: 0
|
|
|
|
]]>
|
|
</send>
|
|
|
|
<nop>
|
|
<action>
|
|
<exec play_pcap_audio="media181sec.pcap"/>
|
|
</action>
|
|
</nop>
|
|
|
|
<recv request="INVITE" crlf="true" rtd="true"/>
|
|
|
|
<send>
|
|
<![CDATA[
|
|
|
|
SIP/2.0 200 OK
|
|
[last_Via:]
|
|
[last_From:]
|
|
[last_To:];tag=[pid]SIPpTag00[call_number]
|
|
[last_Call-ID:]
|
|
[last_CSeq:]
|
|
[last_Record-Route:]
|
|
[last_Route:]
|
|
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
|
|
Content-Type: application/sdp
|
|
Content-Length: [len]
|
|
|
|
v=0
|
|
o=root 1100779239 1100779239 IN IP[local_ip_type] [local_ip]
|
|
s=Asterisk PBX 1.8.10.1
|
|
c=IN IP[media_ip_type] [media_ip]
|
|
t=0 0
|
|
m=audio [media_port] RTP/AVP 8 0 101
|
|
a=rtpmap:8 PCMA/8000
|
|
a=rtpmap:0 PCMU/8000
|
|
a=rtpmap:101 telephone-event/8000
|
|
a=fmtp:101 0-16
|
|
a=silenceSupp:off - - - -
|
|
a=ptime:20
|
|
a=sendrecv
|
|
|
|
]]>
|
|
</send>
|
|
|
|
<recv request="ACK" rtd="true"/>
|
|
|
|
<nop>
|
|
<action>
|
|
<exec play_pcap_audio="media181sec.pcap"/>
|
|
</action>
|
|
</nop>
|
|
|
|
<pause milliseconds="500"/>
|
|
|
|
<send start_rtd="4">
|
|
<![CDATA[
|
|
|
|
BYE [next_url] SIP/2.0
|
|
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
|
|
From: <sip:[field0 file="caller.csv"]@[field2 file="caller.csv"]>;tag=[pid]SIPpTag00[call_number]
|
|
To: <sip:[field0 file="callee.csv" line=0]@[field3 file="callee.csv" line=0]>[peer_tag_param]
|
|
Call-ID: NGCP%[field4 file="callee.csv" line=0]%///[call_id]
|
|
CSeq: [cseq] BYE
|
|
Contact: <sip:[field0 file="caller.csv"]@[local_ip]:[local_port];transport=[transport]>
|
|
[routes]
|
|
Max-Forwards: 70
|
|
Content-Length: 0
|
|
|
|
]]>
|
|
</send>
|
|
|
|
<recv response="100" optional="true">
|
|
</recv>
|
|
|
|
<recv response="200" crlf="true" rtd="4">
|
|
</recv>
|
|
|
|
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200, 500, 1000"/>
|
|
|
|
<!-- <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/> -->
|
|
|
|
</scenario>
|
|
|