Hi,
I'm trying to write a scenario where sipp should act as B side and I want to
send BYE from sipp
A -> invite -> sipp
A <- 100 trying <- sipp
A <- 180 Ringing <- sipp
A <- 200 OK <- sipp
A -> ACK -> sipp
A <- BYE <- sipp
A -> 200 OK -> sipp
However I'm stuck with getting the correct To/From headers ( and requestln)
when creating the BYE request. Anyone that know of an example where the initial
uas side should act as auc and send a request?
Attached are my scenario sofar ( call setup but no BYE from B)
Regards,
// Andreas
<?xml version="1.0" encoding="ISO-8859-1" ?>
<scenario name="B_termination">
<!-- Receive INVITE to trigger call -->
<recv request="INVITE" crlf="true">
</recv>
<!-- Send 100 Trying -->
<send>
<![CDATA[
SIP/2.0 100 Trying
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<!-- Pause X seconds -->
<pause distribution="uniform" min="500" max="1000" />
<!-- Send 180 Ringing -->
<send>
<![CDATA[
SIP/2.0 180 Session Progress
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<!-- Pause Y seconds -->
<pause distribution="uniform" min="2000" max="10000" />
<!-- Send 200 OK (Answer call) -->
<send retrans="500">
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Type: application/sdp
Content-Length: 133
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 0
a=rtpmap:0 PCMU/8000
]]>
</send>
<!-- Receive ACK -->
<recv request="ACK" crlf="true">
</recv>
<!-- Pause Y seconds -->
<pause distribution="uniform" min="5000" max="20000" />
--- SEND BYE HERE ---
<!-- ReceiveBYE -->
<recv response="200" crlf="true">
</recv>
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