Hi David,
a single pass through a scenario (a "call") has to use the same Call-ID,
that's the principle of SIPp operation. I have written a more detailed
explanation in this response to a loosely related question
<https://sourceforge.net/p/sipp/mailman/message/34707334/>.
For your case, a quick workaround would be to use a feature of SIPp
described in the manual:
Note: [call_id] can be pre-pended with an arbitrary string using '///'.
Example: Call-ID: ABCDEFGHIJ///[call_id] - it will still be recognized
by SIPp as part of the same call.
<https://sourceforge.net/p/sipp/mailman/message/34707334/>
So you would still use the Call-ID value extracted from the incoming
REGISTER also for the dialog initiated using the INVITE, but for all
messages sent in the context of that dialog, you would prepend the
Call-ID value with e.g. a "my///" string. This should be enough to make
the UUT interpret such Call-ID value as a distinct one, while SIPp would
recognize the messages received with this modified call-id value as
belonging to the same pass through the scenario.
Pavel
Dne 14.7.2016 v 18:30 Drell, David napsal(a):
hi all,
my scenario is sipp acting as registrar and proxy which accepts the
registration then invites the unit I am testing.
uas-uut sipp
---------- ------
--> REGISTER -->
<-- 200 OK <--
<-- INVITE <-----
--> 100 -->
--> 180 -->
--> 200 -->
sipp is using the same call ID from the received register in the
outbound invite - which is wrong and causes my uas code to reject the
invite.
how can I make sipp increment the call ID when it starts the call?
thank you.
////////////////////////////////// scenario below
//////////////////////////////////////
<scenario name="Gs registers with sipp as CUCM">
<recv request="REGISTER" ></recv>
<send>
<![CDATA[
SIP/2.0 200 OK
Via: SIP/2.0/TCP [local_ip];branch=[branch]
From: "2367" <sip:2367@[local_ip]>;tag=[call_number]
To: <sip:2367@[local_ip]>;tag=585257356
Date: Wed, 13 Jul 2016 14:32:59 GMT
[last_Call-ID:]
Server: Cisco-CUCM10.5
CSeq: 1 REGISTER
Expires: 120
Contact: <sip:2367@[local_ip];transport=tcp>;methods="INVITE, ACK,
BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE,
REFER"
Supported: X-cisco-sis-7.1.1
Content-Length: 0
]]>
</send>
<send>
<![CDATA[
INVITE sip:2367@[local_ip];transport=tcp SIP/2.0
Via: SIP/2.0/TCP [local_ip];branch=[branch]
From: "GS Austin Dev Codec 8" <sip:2368@[local_ip]>;tag=[call_number]
To: <sip:2367@[local_ip]>
Date: Wed, 13 Jul 2016 14:33:24 GMT
Call-ID: [call_id]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM10.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Send-Info: conference, x-cisco-conference
Alert-Info: <file://Bellcore-dr1/>
Contact:
<sip:2368@[local_ip];transport=tcp>;video;audio;proxy=replace;+sip.instance="<urn:uuid:8500b350-5ef3-592d-bd58-77ebbd405217>"
Remote-Party-ID: "GS Austin Dev Codec 8"
<sip:2368@[local_ip];x-cisco-callback-number=2368>;party=calling;screen=yes;privacy=off
Max-Forwards: 69
Content-Length: 0
]]>
</send>
<recv response="100" optional="true"></recv>
</scenario>
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