On 1/2/08, Charles P Wright <[EMAIL PROTECTED]> wrote:
> Do you have any actions, if not I am not sure why the call would be marked
> as rejected without any other messages in the logs. Can you post your
> scenario XML?
OK. You will find attached a simplified version of the scenario that
reproduces the problem. When the BYE (message 8 in the scenario) is
not received, the call is
marked as failed.
Regards,
K.L.
>
> Charles
>
> "K L" <[EMAIL PROTECTED]> wrote on 01/02/2008 01:32:23 PM:
>
> > On 1/2/08, Charles P Wright <[EMAIL PROTECTED]> wrote:
> > > I don't think SIPp should automatically fail the call. Based on the
> > > statistics file, do you know what the failure reason is listed as?
> >
> > If I start one call, I get those errors reported in the CSV statistics
> file:
> >
> > FailedCall(P): 1
> > FailedCall(C): 1
> > FailedCallRejected(P): 1
> > FailedCallRejected(C): 1
> >
> > All other "failed call" counters are set to zero.
> >
> > >
> > > The error log message is actually a warning, as long as you don't jump
> to
> > > the end, the call should not be terminated. Are there any other
> messages
> > > in the error log file after this one?
> >
> > To make the things clear (sorry if my explanations weren't clear
> > enough), the call is not actually terminated, it continues as
> > expected, but is marked as failed in the statistics.
> >
> > Here is the full content of the error log:
> >
> > "sipp: The following events occured:
> > 2008-01-02 19:50:02:864 1199299802.864035: Call-Id:
> > [EMAIL PROTECTED], receive timeout on message 10, jumping to label 2.
> > 2008-01-02 19:50:32:898 1199299832.898937: Call-Id:
> > [EMAIL PROTECTED], receive timeout on message 10, jumping to label 2.
> > 2008-01-02 19:51:02:927 1199299862.927862: Call-Id:
> > [EMAIL PROTECTED], receive timeout on message 10, jumping to label
> > 2."
> >
> > >
> > > Charles
> > >
> >
> > Regards,
> > K.L.
> >
> > > "K L" <[EMAIL PROTECTED]> wrote on 01/02/2008 11:22:46 AM:
> > >
> > > > On 1/2/08, Charles P Wright <[EMAIL PROTECTED]> wrote:
> > > > > K L,
> > > > >
> > > > > If you jump to the end of the call SIPp marks it as failed
> > > automatically
> > > > > (which I think is probably not the best possible behavior), but
> you
> > > can
> > > > > easily get around it by doing something like:
> > > >
> > > > In my case, I don't jump to the end of the scenario:
> > > >
> > > > - If a BYE is received, I jump to the "next" label (4) to send a 200
> OK.
> > > > - If no BYE is received, I jump to the "ontimeout" label (2), to
> wait
> > > > for another request (non-optional).
> > > >
> > > > So, in both cases, there is an action made, and I do not jump to the
> > > > end of the scenario.
> > > >
> > > > >
> > > > > <recv request="BYE" timeout="25000" ontimeout="2" next="4" />
> > > > >
> > > > > ...
> > > > >
> > > > > <label id="2" />
> > > > > <!-- just so we have a scenario element after label 2 -->
> > > > > <nop />
> > > > > <!-- end of scenario -->
> > > > >
> > > > > Charles
> > > > >
> > > >
> > > > Regards,
> > > > K.L.
> > > >
> > > > > [EMAIL PROTECTED] wrote on 12/31/2007
> 09:51:39
> > > AM:
> > > > >
> > > > > > Hello,
> > > > > >
> > > > > > In a (UAC) scenario, I have the following line, to allow the
> remote
> > > > > > UA to send a
> > > > > > BYE during the call (after the dialog is established):
> > > > > >
> > > > > > <recv request="BYE" optional="true" timeout="25000"
> ontimeout="2"
> > > > > next="4" />
> > > > > >
> > > > > > Whether a BYE is received or not, I'd like the call to be
> considered
> > > > > > as successful.
> > > > > > If no BYE request is received from the remote UA, the call
> proceeds
> > > > > normally.
> > > > > > The problem is that the call is considered as failed by SIPp
> > > ("receive
> > > > > > timeout on message 10, jumping to label 2").
> > > > > >
> > > > > > Since the recv is marked as "optional", it seems strange to me
> that
> > > SIPp
> > > > > > considers the jump to label 2 as an error. Is there an option to
> > > change
> > > > > this
> > > > > > behaviour ?
> > > > > >
> > > > > > Regards,
> > > > > > K.L.
> > > > > >
> > > > > >
> > > > >
> > >
> -------------------------------------------------------------------------
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> > > > >
> > > > >
> > >
> > >
>
>
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="UAC1">
<!-- Call variables -->
<nop>
<action>
<!-- $1: Caller -->
<assignstr assign_to="1" value="[field0]" />
<!-- $4: PAI -->
<assignstr assign_to="4" value="[field1]" />
<!-- $2: Callee -->
<assignstr assign_to="2" value="[service]" />
<!-- $3: Domain -->
<assignstr assign_to="3" value="[field2]" />
</action>
</nop>
<send retrans="500">
<![CDATA[
INVITE sip:[EMAIL PROTECTED]:[remote_port];user=phone SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: "[$1]" <sip:[EMAIL PROTECTED]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: <sip:[EMAIL PROTECTED]>
Call-ID: [call_id]
CSeq: 1 INVITE
P-Asserted-Identity: <sip:[EMAIL PROTECTED]>
Max-Forwards: 68
User-Agent: sipp2
Accept: application/sdp,audio/telephone-event,application/media_control+xml,application/dtmf-relay,message/sipfrag,text/html,text/plain
Expires: 1800
Contact: <sip:[EMAIL PROTECTED]:[local_port];transport=[transport]>
Content-Type: application/sdp
Content-Length: [len]
v=0
o=anonymous 1183378365 1183378365 IN IP[local_ip_type] [local_ip]
s=-
i=test
c=IN IP[media_ip_type] [media_ip]
b=AS:384
t=0 0
m=audio [media_port] RTP/AVP 102 104 9 120 108 18 4 8 0 110
a=rtpmap:102 X-G72x1/16000
a=sendrecv
a=rtpmap:104 X-G72x24/16000
a=rtpmap:9 G722/8000
a=rtpmap:120 X-FT-SCA729/16000
a=rtpmap:108 X-G72xH/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:110 telephone-event/8000
a=fmtp:110 0-15
]]>
</send>
<recv response="100" optional="true" />
<recv response="180" optional="true" />
<recv response="200" optional="false" />
<send>
<![CDATA[
ACK sip:[EMAIL PROTECTED]:[remote_port];user=phone SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: "[$1]" <sip:[EMAIL PROTECTED]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: <sip:[EMAIL PROTECTED]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Max-Forwards: 68
Content-Length: 0
]]>
</send>
<recv request="INFO" optional="false" timeout="8000" ontimeout="3" />
<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-Length: 0
]]>
</send>
<label id="1" />
<recv request="BYE" optional="true" timeout="25000" ontimeout="2" next="4"/>
<label id="2" />
<recv request="INFO" optional="false" timeout="10000" ontimeout="3" />
<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-Length: 0
]]>
</send>
<label id="3" />
<send retrans="500">
<![CDATA[
BYE sip:[EMAIL PROTECTED]:[remote_port];user=phone SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: "[$1]" <sip:[EMAIL PROTECTED]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: <sip:[EMAIL PROTECTED]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 BYE
Contact: <sip:[EMAIL PROTECTED]:[local_port];transport=[transport]>
Max-Forwards: 68
Content-Length: 0
]]>
</send>
<!-- Wait the 200 response to our BYE and stop the call -->
<recv response="200" optional="false" crlf="true">
<action>
<exec int_cmd="stop_call" />
</action>
</recv>
<!-- Send a 200 OK response to a BYE -->
<label id="4" />
<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-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>
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