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uac_auth_audio.xml
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uac_auth_audio.xml
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<?xml version="1.0" encoding="ISO-8859-1"?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="Basic Sipstone UAC">
<send retrans="500">
<![CDATA[
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=z9hG4bK[call_number]
From: "[field0]" <sip:[field0]@[field1]>;tag=[call_number]
To: <sip:[service]@[field1]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: <sip:[field0]@[local_ip]:[local_port]>
Content-Type: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER
Max-Forwards: 70
User-Agent: SIPp
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: [len]
v=0
o=- 20102 20102 IN IP[local_ip_type] [local_ip]
s=SDP data
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0 8 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendrecv
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="407" auth="true">
</recv>
<send>
<![CDATA[
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
[last_Via:]
[last_From:]
To: <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
[last_Call-ID:]
CSeq: 1 ACK
Content-Length: 0
]]>
</send>
<send retrans="500">
<![CDATA[
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
[last_Via:]
[last_From:]
To: <sip:[service]@[remote_ip]:[remote_port]>
[last_Call-ID:]
CSeq: 2 INVITE
Contact: <sip:[field0]@[local_ip]:[local_port]>
[field2]
Content-Type: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER
Max-Forwards: 70
User-Agent: SIPp
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: [len]
v=0
o=- 20102 20102 IN IP[local_ip_type] [local_ip]
s=SDP data
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0 8 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendrecv
]]>
</send>
<recv response="100" optional="true"></recv>
<recv response="101" optional="true"></recv>
<recv response="180" optional="true"></recv>
<recv response="183" optional="true"></recv>
<recv response="480" optional="true" next="5"></recv>
<recv response="488" optional="true" next="5"></recv>
<recv response="502" optional="true" next="5"></recv>
<recv response="503" optional="true" next="6"></recv>
<recv response="486" optional="true" next="6"></recv>
<recv response="407" optional="true" next="6"></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:[service]@[remote_ip]:[remote_port] SIP/2.0
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
[last_Contact:]
CSeq: 2 ACK
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<pause milliseconds="10000"/>
<nop>
<action>
<exec play_pcap_audio="g711a_2.pcap"/>
</action>
</nop>
<!-- This delay can be customized by the -d command-line option -->
<!-- or by adding a 'milliseconds = "value"' option here. -->
<pause/>
<!-- 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
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
[last_Contact:]
CSeq: 3 BYE
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<recv response="200" crlf="true" next="6"></recv>
<label id="5"/>
<send>
<![CDATA[
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
[last_Contact:]
CSeq: 2 ACK
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<label id="6"/>
<!-- 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>