Hi,
I tried your solution,but faces dead call successful error.
I have executed two calls from server and client,for each invite,different
error should send back..
Error in server side:
[root@elgn0euasa9sip1 Termination]# sipp -inf info -sf 4xx_Error_UAS.xml -p
6001 -trace_msg -m 2
------------------------------ Scenario Screen -------- [1-9]: Change Screen
--
Port Total-time Total-calls Transport
6001 11.54 s 2 UDP
Call limit reached (-m 2), 0.000 s period 0 ms scheduler resolution
0 calls Peak was 2 calls, after 6 s
0 Running, 3 Paused, 0 Woken up
1 dead call msg (discarded)
1 open sockets
Messages Retrans Timeout Unexpected-
Msg
----------> INVITE 2 3 0 0
[ NOP ]
<---------- 408 1 0
[ 5000ms] Pause 2 0
<---------- 491 1 0
------------------------------ Test Terminated -----------------------------
---
UAS.xml :
<scenario name="408 error UAS responder">
<recv request="INVITE" crlf="true">
</recv>
<nop>
<action>
<assignstr assign_to="rsp_code" value="[field0]"/>
<ereg regexp="408" search_in="var" variable="rsp_code"
assign_to="rsp_408"/>
<ereg regexp="491" search_in="var" variable="rsp_code"
assign_to="rsp_491"/>
</action>
</nop>
<send condexec="rsp_408">
<![CDATA[
SIP/2.0 408 [field1]
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
[last_CSeq:]
Content-Length: 0
]]>
</send>
<pause milliseconds="5000">
</pause>
<send condexec="rsp_491">
<![CDATA[
SIP/2.0 491 [field1]
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
[last_CSeq:]
Content-Length: 0
]]>
</send>
<!-- 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>
UAC.xml:
<scenario name="408 Error UAC">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. The Call-ID
-->
<!-- match is what makes it look like a re-invite. In Call-ID for
Invites, -->
<!-- use IP address of this laptop. -->
<send retrans="500">
<![CDATA[
INVITE sip:[field2];phone-context=atlanta.example.com@[remote_ip]:
[remote_port];user=phone SIP/2.0
Content-Type: application/sdp
To: <sip:[field2]@[remote_ip]:[remote_port];user=phone>
Via: SIP/2.0/UDP [local_ip]:[local_port];branch=[branch];x-route-tag="
[field3]";received=[local_ip]
Allow: INVITE, ACK, PRACK, SUBSCRIBE, BYE, CANCEL, NOTIFY, INFO, REFER,
UPDATE
MIME-Version: 1.0
Call-ID: ///[call_id]
From: <sip:[field1]@[local_ip]:[local_port];user=phone>;tag=[call_number]
Max-Forwards: 9
Contact: <sip:[field1]@[local_ip]:[local_port]>
Session-Expires: 604800;refresher=uac
CSeq: [field4] INVITE
Content-Length: [len]
Supported: timer, 100rel
v=0
o=- 270 0 IN IP[local_ip_type] [local_ip]
s=Cisco SDP 0
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0 100
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
a=rtpmap:100 X-NSE/8000
a=fmtp:100 200-202
]]>
</send>
<recv response="408" crlf="true">
</recv>
<pause milliseconds="5000"/>
<recv response="491" 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>
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