Re: [Sipp-users] Phone can not send the ACK and BYE to sipp

[Bui D. T. <bui...@gm...>]

hj all! can you help me!
i don't know  why my softphone can't send any message to my sipp server???
ACK, BYE. i use wireshark to check, but i didn't see any message from phone
send to sipp.
regard
 thanks

2011/7/19 Bui Dinh Thang <bui...@gm...>

> Thanks reply!i will try your suggest
> regard
>
> 2011/7/19 Patrick Wakano <pw...@gm...>
>
>> 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 <
>> bui...@gm...> 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 <gop...@gm...>
>>>
>>>> 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 <
>>>> bui...@gm...> 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
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>>>>> http://p.sf.net/sfu/appsumosfdev2dev
>>>>> _______________________________________________
>>>>> Sipp-users mailing list
>>>>> Sip...@li...
>>>>> 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
>>> Sip...@li...
>>> https://lists.sourceforge.net/lists/listinfo/sipp-users
>>>
>>>
>>
>
>
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