Hi,
SIPp -xlite communication working fine
SIPp-SIPp not working ...?
Below is details explanation
Scenario A1-
Node A Caller
Node B Xlite
<send > REGISTER </send>
<recv response="200" crlf="true" />
<send > SUBSCRIBE </send>
<recv response="501" crlf="true" />
<send > INVITE </send>
<recv response="486" optional="true" />
<recv response="180" optional="true"/>
<send> ACK </Send>
This Works Perfectly on Node B xLite is able to receive call. Working Good.
Scenario B2-
Node A Caller
Node B Receiver
<send > REGISTER </send>
<recv response="200" crlf="true" />
<send > SUBSCRIBE </send>
<recv response="501" crlf="true" />
<send > INVITE </send>
<recv response="486" optional="true" />
<recv response="180" optional="true"/>
<send> ACK </Send>
<send > REGISTER </send>
<recv response="200" crlf="true" />
<send > SUBSCRIBE </send>
<recv response="501" crlf="true" />
<recv request="INVITE">
</recv>
<send>
<![CDATA[
SIP/2.0 180 Ringing
[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>
Scenario B2- on Node B
side does not work..
Node B - got error on during Invite receive here is error:
Discarding message which can't be mapped to a known SIPp call: INVITE sip
Any clue or suggestion what could be the reason ?
Thanks in advance
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!-- This program is free software; you can redistribute it and/or -->
<!-- modify it under the terms of the GNU General Public License as -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version. -->
<!-- -->
<!-- This program is distributed in the hope that it will be useful, -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
<!-- GNU General Public License for more details. -->
<!-- -->
<!-- You should have received a copy of the GNU General Public License -->
<!-- along with this program; if not, write to the -->
<!-- Free Software Foundation, Inc., -->
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
<!-- -->
<!-- Sipp default 'uac' scenario. -->
<!-- -->
<scenario name="Basic Sipstone UAC">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<send >
<![CDATA[
REGISTER sip:10.230.53.225 SIP/2.0
Via: SIP/2.0/UDP 10.230.52.50:[local_port];branch=[branch];rport
Max-Forwards: 70
Contact: <sip:te...@10.230.52.50:[local_port];rinstance=df769c75a8bad123>
To: "test2"<sip:te...@10.230.53.225>
From: "test2"<sip:te...@10.230.53.225>;tag=[call_number]
Call-ID: [call_id]
CSeq: [cseq] REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Length: [len]
]]>
</send>
<recv response="200" crlf="true">
</recv>
<send >
<![CDATA[
SUBSCRIBE sip:te...@10.230.53.225 SIP/2.0
Via: SIP/2.0/UDP 10.230.52.50:[local_port];branch=[branch];rport
Max-Forwards: 70
Contact: <sip:te...@10.230.52.50:[local_port]>
To: "test2"<sip:te...@10.230.53.225>
From: "test2"<sip:te...@10.230.53.225>;tag=[call_number]
Call-ID: [call_id]
CSeq: [cseq] SUBSCRIBE
Expires: 300
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Event: message-summary
Content-Length: [len]
]]>
</send>
<recv response="501" crlf="true">
</recv>
<label id="1"/>
<send >
<![CDATA[
INVITE sip:[fiel...@10.230.53.225 SIP/2.0
Via: SIP/2.0/UDP 10.230.52.50:[local_port];branch=[branch];rport
Max-Forwards: 70
Contact: <sip:te...@10.230.52.50:[local_port]>
To: "[field0]"<sip:[fiel...@10.230.53.225>
From: "test2"<sip:te...@10.230.53.225>;tag=[call_number]
Call-ID: [call_id]
CSeq: [cseq] INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Content-Length: [len]
v=0
o=- 9 2 IN IP4 10.230.52.50
s=CounterPath X-Lite 3.0
c=IN IP4 10.230.52.50
t=0 0
m=audio 14554 RTP/AVP 107 0 8 101
a=alt:1 3 : 8M33e0IX 0MCl+l2S 10.230.52.50 14554
a=alt:2 2 : BMV0PwaS OFqCq9q3 192.168.222.1 14554
a=alt:3 1 : RshdMeaM ZFpg93ys 192.168.81.1 14554
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv
]]>
</send>
<recv response="486" optional="true">
</recv>
<recv response="180" optional="true">
<action>
<ereg regexp="\=(.*)"
search_in="hdr"
header="From:"
check_it="true"
assign_to="2"/>
<ereg regexp="\=(.*)"
search_in="hdr"
header="To:"
check_it="true"
assign_to="1"/>
</action>
</recv>
<recv response="100"
optional="true">
</recv>
<!-- By adding rrs="true" (Record Route Sets), the route sets -->
<!-- are saved and used for following messages sent. Useful to test -->
<!-- against stateful SIP proxies/B2BUAs. -->
<recv response="200" rtd="true" >
<action>
<ereg regexp="\=(.*)"
search_in="hdr"
header="From:"
check_it="true"
assign_to="2"/>
<ereg regexp="\=(.*)"
search_in="hdr"
header="To:"
check_it="true"
assign_to="1"/>
</action>
</recv>
<!-- Packet lost can be simulated in any send/recv message by -->
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
<send>
<![CDATA[
ACK sip:2...@10.230.53.225 SIP/2.0
Via:SIP/2.0/UDP 10.230.52.50:[local_port];branch=[branch];rport
Max-Forwards: 70
Contact: <sip:te...@10.230.52.50:[local_port]>
To: "[field0]"<sip:[fiel...@10.230.53.225>;tag[$1]
From: "test2"<sip:te...@10.230.53.225>;tag[$2]
Call-ID: [call_id]
CSeq: [cseq] ACK
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: [len]]
]]>
</send>
<pause milliseconds="20000"/>
<!--
<send >
<![CDATA[
BYE sip:[fiel...@10.230.53.225 SIP/2.0
Via: SIP/2.0/UDP 10.230.52.50:[local_port];branch=[branch];rport
Max-Forwards: 70
Contact: <sip:te...@10.230.52.50:[local_port]>
To: "[field0]"<sip:[fiel...@10.230.53.225>;tag[$1]
From: "test2"<sip:te...@10.230.53.225>;tag[$2]
Call-ID: [call_id]
CSeq: [cseq] BYE
User-Agent: X-Lite release 1104o stamp 56125
Reason: SIP;description="User Hung Up"
Content-Length: [len]
]]>
</send>
-->
<recv response="200">
</recv>
<pause next="1" milliseconds="6000"/>
<!-- 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>
<?xml version="1.0" encoding="iso-8859-1" ?>
<!-- This program is free software; you can redistribute it and/or -->
<!-- modify it under the terms of the GNU General Public License as -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version. -->
<!-- -->
<!-- This program is distributed in the hope that it will be useful, -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
<!-- GNU General Public License for more details. -->
<!-- -->
<!-- You should have received a copy of the GNU General Public License -->
<!-- along with this program; if not, write to the -->
<!-- Free Software Foundation, Inc., -->
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
<!-- -->
<!-- Sipp default 'uac' scenario. -->
<!-- -->
<scenario name="Basic Sipstone UAC">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<send >
<![CDATA[
REGISTER sip:10.230.53.225 SIP/2.0
Via: SIP/2.0/UDP 10.230.53.227:5060;branch=[branch];rport
Max-Forwards: 70
Contact: <sip:4...@10.230.53.227:5060>
To: "420"<sip:4...@10.230.53.225>
From: "420"<sip:4...@10.230.53.225>;tag=[call_number]
Call-ID: [call_id]
CSeq: [cseq] REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Length: [len]
]]>
</send>
<recv response="200" crlf="true">
</recv>
<send >
<![CDATA[
SUBSCRIBE sip:4...@10.230.53.225 SIP/2.0
Via: SIP/2.0/UDP 10.230.53.227:5060;branch=[branch];rport
Max-Forwards: 70
Contact: <sip:4...@10.230.53.227:5060>
To: "420"<sip:4...@10.230.53.225>
From: "420"<sip:4...@10.230.53.225>;tag=[call_number]
Call-ID: [call_id]
CSeq: [cseq] SUBSCRIBE
Expires: 300
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 4204o stamp 56125
Event: message-summary
Content-Length: [len]
]]>
</send>
<recv response="501" crlf="true">
</recv>
<recv request="INVITE">
</recv>
<send>
<![CDATA[
SIP/2.0 180 Ringing
[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>
<!--<recv response="INVITE" >
<action>
<ereg regexp="\=(.*)" search_in="hdr" header="From:" assign_to="1" />
<ereg regexp="\: (.*)" search_in="hdr" header="Call-ID:" assign_to="2" />
<ereg regexp="\=(.*)" search_in="hdr" header="Via:" assign_to="3" />
<ereg regexp="\=(.*)" search_in="hdr" header="Contact:" assign_to="4" />
<log message="From is [last_From]. Custom header is [$1]"/>
<log message="From is [last_Call-ID]. Custom header is [$2]"/>
<log message="From is [last_branch]. Custom header is [$3]"/>
<log message="From is [last_rinstance]. Custom header is [$4]"/>
</action>
</recv>
<send >
<![CDATA[
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.230.53.225:5060;branch[$3]
Contact: <sip:4...@10.230.53.227:64118;rinstance[$4]>
To: <sip:4...@10.230.53.225>;tag=8e213e4e
From: <sip:te...@10.230.53.225>;tag[$1]
Call-ID: [$2]
CSeq: 1 INVITE
User-Agent: X-Lite release 4204o stamp 56125
Content-Length: [len]
]]>
</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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