Try the attached script, this one from SIPp. If this script works then
include your pcap audio file.
On Mon, Jul 18, 2011 at 4:48 PM, Bui Dinh Thang <buidinhthan...@gmail.com>wrote:
> hj all!
> i try to simulate the scenario, but i don't know what problem, phone can
> not send the ACK and BYE to sipp, i don't understand it.
>
> this my scenario!
>
> <?xml version="1.0" encoding="ISO-8859-1" ?>
>
> <!DOCTYPE scenario SYSTEM "sipp.dtd">
> <scenario name="UASBasic">
>
> <recv request="INVITE" crlf="true" rrs="true">
> </recv>
>
> <send>
>
> <![CDATA[
>
> SIP/2.0 100 Trying
>
> [last_To:];tag=[pid]SIPpTag01[call_number]
>
> [last_From:]
>
> [last_Call-ID:]
>
> [last_CSeq:]
>
> Contact: [field0] <sip:[local_ip]:[local_port]>
>
> [last_Via:]
>
> User-Agent: uas
>
> Content-Length: 0
>
> ]]>
>
> </send>
>
> <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]>
>
> [last_record-Router:]
>
> Content-Length: 0
>
> ]]>
>
> </send>
>
> <send>
> <![CDATA[
> SIP/2.0 183 Session Progress
> [last_Via:]
> [last_From:]
> [last_To:]
> [last_Call-ID:]
> [last_CSeq:]
> User-Agent: SIP
> Contact: <sip:[local_ip]:[local_port];transport=[transport]>
> Content-Length: 0
> ]]>
> </send>
>
> <send retrans="500">
>
> <![CDATA[
>
> SIP/2.0 200 OK
>
> [last_Via:]
>
> [last_From:]
>
> [last_To:] ;tag=[call_number]
>
> [last_Call-ID:]
>
> [last_CSeq:]
>
> [last_Record-Route:]
>
> Supported: timer
>
> Contact: <sip:[local_ip]:[local_port];transport=[transport]>
>
> Content-Type: application/sdp
>
> Content-Length: [len]
>
> v=0
>
> o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
>
> s= SIPp - UAS
>
> c=IN IP[media_ip_type] [media_ip]
>
> t=0 0
>
> m=audio [media_port] RTP/AVP 0
>
> a=rtpmap:0 PCMU/8000
>
> a=sendrecv
>
> a=rtpmap:98 telephone-event/8000
>
> ]]>
>
> </send>
> *<recv request="ACK" rtd="true" crlf="true">*
>
> </recv>
>
> <nop>
>
> <action>
>
> <exec play_pcap_audio="pcap/test.pcap"/>
>
> </action>
>
> </nop>
>
> *<recv request="BYE">*
>
> </recv>
>
> <send>
>
> <![CDATA[
>
>
>
> SIP/2.0 200 OK
>
> [last_Via:]
>
> [last_From:]
>
> [last_To:]
>
> [last_Call-ID:]
>
> [last_CSeq:]
>
> Contact: [field0] <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>
>
>
> please help me show the errors my scenario?
> Best Regard!
> thanks
> --
> Thắng
>
>
> ------------------------------------------------------------------------------
> AppSumo Presents a FREE Video for the SourceForge Community by Eric
> Ries, the creator of the Lean Startup Methodology on "Lean Startup
> Secrets Revealed." This video shows you how to validate your ideas,
> optimize your ideas and identify your business strategy.
> http://p.sf.net/sfu/appsumosfdev2dev
> _______________________________________________
> Sipp-users mailing list
> Sipp-users@lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/sipp-users
>
>
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!-- 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 'uas' scenario. -->
<!-- -->
<scenario name="Basic UAS responder">
<!-- 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 request="INVITE" crlf="true">
</recv>
<!-- The '[last_*]' keyword is replaced automatically by the -->
<!-- specified header if it was present in the last message received -->
<!-- (except if it was a retransmission). If the header was not -->
<!-- present or if no message has been received, the '[last_*]' -->
<!-- keyword is discarded, and all bytes until the end of the line -->
<!-- are also discarded. -->
<!-- -->
<!-- If the specified header was present several times in the -->
<!-- message, all occurences are concatenated (CRLF seperated) -->
<!-- to be used in place of the '[last_*]' keyword. -->
<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>
<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: [len]
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>
<recv request="ACK"
optional="true"
rtd="true"
crlf="true">
</recv>
<recv request="BYE">
</recv>
<send>
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<!-- Keep the call open for a while in case the 200 is lost to be -->
<!-- able to retransmit it if we receive the BYE again. -->
<pause milliseconds="4000"/>
<!-- 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>
------------------------------------------------------------------------------
AppSumo Presents a FREE Video for the SourceForge Community by Eric
Ries, the creator of the Lean Startup Methodology on "Lean Startup
Secrets Revealed." This video shows you how to validate your ideas,
optimize your ideas and identify your business strategy.
http://p.sf.net/sfu/appsumosfdev2dev
_______________________________________________
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Sipp-users@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/sipp-users
