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invite-without-auth.xml
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<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!-- You will need to compile SIPp with OpenSSL support 'make ossl' to use this call scenario -->
<!-- Execute this script with SIPp using the following command assuming your UAS is 10.0.0.10 -->
<!-- Replace 10.0.0.10 with your SIP proxy’s address. The command will generate 10 calls (-r) per 10000 -->
<!-- milliseconds (-rp), max 100 concurrent calls (-l) and make a max of 100000 calls (-m) -->
<!-- ./sipp 10.0.0.10 -sf invite-auth-sdp-nomedia.xml -inf user-accounts.csv -m 100000 -l 100 -r 10 -rp 10000 -->
<scenario name="UAC INVITE with Auth and SDP">
<send retrans="500" start_txn="invite">
<![CDATA[
INVITE sip:[field3]@[field1]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:[field0]@[field1]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: sut <sip:[field3]@[field1]:[remote_port]>
Call-ID: [call_id]
CSeq: [cseq] INVITE
Contact: <sip:sipp@[local_ip]:[local_port]>
Max-Forwards: 70
Subject: Performance Test
User-Agent: SIPp Tester UAC
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
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 8
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16
]]>
</send>
<recv response="100" optional="true" response_txn="invite">
</recv>
<recv response="180" optional="true" response_txn="invite">
</recv>
<recv response="183" optional="true" response_txn="invite">
</recv>
<!-- Grab the 200 OK's Contact Header's URI for use in the ACK's Request URI -->
<recv response="200" rtd="true" rrs="true" response_txn="invite">
<action>
<ereg regexp= "sip:[^+>]+" search_in="hdr" header="Contact:" assign_to="1" />
</action>
</recv>
<!-- decrement the ACK's branch by 3 to match previous INVITE's branch -->
<send ack_txn="invite">
<![CDATA[
ACK [$1] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
From: sipp <sip:[field0]@[field1]:[local_port]>;tag=[pid]SIPpTag00[call_number]
[last_To]
[routes]
Call-ID: [call_id]
CSeq: [cseq] ACK
Contact: <sip:sipp@[local_ip]:[local_port]>
Max-Forwards: 70
Subject: Performance Test
User-Agent: SIPp Tester UAC
Content-Length: 0
]]>
</send>
<nop>
<action>
<exec play_pcap_audio="./g711a.pcap"/>
</action>
</nop>
<pause milliseconds="8000"/>
<send retrans="500">
<![CDATA[
BYE [$1] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:[field0]@[field1]:[local_port]>;tag=[pid]SIPpTag00[call_number]
[last_To]
[routes]
Call-ID: [call_id]
CSeq: [cseq] BYE
Contact: <sip:sipp@[local_ip]:[local_port]>
Max-Forwards: 70
Subject: Performance Test
User-Agent: SIPp Tester UAC
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>