Thanks reply!i will try your suggest
regard
2011/7/19 Patrick Wakano <pwak...@gmail.com>
> The first thing you have to fix is the usage of the record-route.
> When you receive a Record-route field you are not supposed to reinsert it
> in the response as you are doing with the field "[last_Record-Route]".
> Basically, you should send all your responses to the servers listed in the
> received Record-Route field. In your SIPp script you can do it, by using
> the rrs=true in the received request and inserting the keyword [routes] in
> the response.
> See if this solves your problem.
>
>
>
>
> On Mon, Jul 18, 2011 at 11:59 AM, Bui Dinh Thang <buidinhthan...@gmail.com
> > wrote:
>
>> Hj! thanks your reply!
>> But the script you sent, I've done but still having the same error, phone
>> not sending ACK and BYE message to the sipp, i don't know what's problem
>> happening????
>> regard
>>
>> 2011/7/18 Gopal krishnan <gopalakrishnan...@gmail.com>
>>
>>> 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
>>>>
>>>>
>>>
>>
>>
>> --
>> 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
>>
>>
>
--
Thắng
------------------------------------------------------------------------------
Magic Quadrant for Content-Aware Data Loss Prevention
Research study explores the data loss prevention market. Includes in-depth
analysis on the changes within the DLP market, and the criteria used to
evaluate the strengths and weaknesses of these DLP solutions.
http://www.accelacomm.com/jaw/sfnl/114/51385063/
_______________________________________________
Sipp-users mailing list
Sipp-users@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/sipp-users
