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uas_send_update_after_200ACK.xml
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<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="Basic UAS responder">
<recv request="INVITE" crlf="true">
<action>
<ereg regexp="branch=([[:alnum:]]*)"
search_in="hdr"
header="Via:"
check_it="false"
assign_to="1, 2"/>
</action>
</recv>
<send>
<![CDATA[
SIP/2.0 183 Session Progress
[last_Via:]
[last_From:]
[last_To:];tag=[pid]SIPpTag01[call_number]
[last_Call-ID:]
[last_CSeq:]
Require: 100rel
RSeq: 561
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] 9.6.2.12
s=-
c=IN IP[media_ip_type] 9.6.2.12
t=0 0
m=audio 18200 RTP/AVP 0
]]>
</send>
<recv request="PRACK"
rtd="true"
crlf="true">
</recv>
<send>
<![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]
]]>
</send>
<send>
<![CDATA[
SIP/2.0 200 OK
Via: SIP/2.0/UDP 9.6.2.35:5060;[$1]
[last_From:]
[last_To:]
[last_Call-ID:]
CSeq: 101 INVITE
Require: 100rel
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] 9.6.2.12
s=-
c=IN IP[media_ip_type] 9.6.2.12
t=0 0
m=audio 18200 RTP/AVP 0
]]>
</send>
<recv request="ACK" rtd="true" crlf="true">
<action>
<ereg regexp=".*"
search_in="hdr"
header="From:"
check_it="false"
assign_to="15"/>
<ereg regexp=".*"
search_in="hdr"
header="To:"
check_it="false"
assign_to="14"/>
</action>
</recv>
<pause milliseconds="13000"/>
<send>
<![CDATA[
UPDATE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: [$14]
To: [$15]
[last_Call-ID:]
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 3 UPDATE
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Allow-Events: telephone-event
Expires: 180
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] 9.6.2.12
s=-
c=IN IP[media_ip_type] 9.6.2.12
t=0 0
m=audio 18200 RTP/AVP 0
]]>
</send>
<recv response="200" crlf="true">
</recv>
<recv request="BYE"
rtd="true"
crlf="true">
</recv>
<send>
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
[last_CSeq:]
Content-Length: 0
]]>
</send>
<!-- Keep the call open for a while in case the 200 is lost to be -->
<!-- able to retransmit it if we receive the BYE again. -->
<pause milliseconds="4000"/>
<!-- 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>