On Fri, Mar 4, 2011 at 3:00 AM, Gopalakrishnan A.N <sai...@gmail.com> wrote:
> while executing the attached script the result is success and there is no
> failed call, infact i am able to generate sip log in the server end. But
> while checking the wireshark log the flow seems to be different when
> compared to normal scenario. Attached is the pcap file.
>
> why is the flow seems to be different just want to understand. The results
> are success no error in the sipp.
>
>
You are not telling us what is the difference you see between the scenario
file and the packet capture.
And there is not a single complete call in the pcap file (all calls are in
"IN CALL" or "CALL SETUP" state instead of "COMPLETED").
It is better to run sipp with "-m 1" and get the capture of only one call
from start to end. Then you can be more clear saying: "Look at packet
number 3. Why we have such packet? It is not expected according to the
scenario but the call was successful anyway" or something like that.
>
> On Thu, Mar 3, 2011 at 8:46 PM, Gopalakrishnan A.N <sai...@gmail.com>wrote:
>
>> Hi,
>>
>> I am able to execute the INVITE - 100 - 180 - 200 - ACK - Bye... its
>> working fine... let me do some more R&D....Thanks for all your assistance.
>>
>>
>> On Thu, Mar 3, 2011 at 6:41 PM, Gopalakrishnan A.N <sai...@gmail.com>wrote:
>>
>>> ok thank you
>>>
>>>
>>> On Thu, Mar 3, 2011 at 6:16 AM, mayamatakeshi
>>> <mayamatake...@gmail.com>wrote:
>>>
>>>>
>>>>
>>>> On Thu, Mar 3, 2011 at 9:45 AM, mayamatakeshi
>>>> <mayamatake...@gmail.com>wrote:
>>>>
>>>>>
>>>>>
>>>>> On Thu, Mar 3, 2011 at 12:57 AM, Gopalakrishnan A.N
>>>>> <sai...@gmail.com>wrote:
>>>>>
>>>>>> What is the difference between writing script like this for INVITE
>>>>>>
>>>>>> <recv request="INVITE" rrs="true" crlf="true" />
>>>>>>
>>>>>> and like this
>>>>>>
>>>>>> INVITE sip:*[service]*@*[remote_ip]*:*[remote_port]* SIP/2.0
>>>>>> Via: SIP/2.0/*[transport]* *[local_ip]*:*[local_port]*
>>>>>> From: sipp <sip:sipp@*[local_ip]*:*[local_port]*>;tag=*[call_number]*
>>>>>> To: sut <sip:*[service]*@*[remote_ip]*:*[remote_port]*>
>>>>>> Call-ID: *[call_id]*
>>>>>> Cseq: 1 INVITE
>>>>>> Contact: sip:sipp@*[local_ip]*:*[local_port]*
>>>>>> Max-Forwards: 70
>>>>>> Subject: Performance Test
>>>>>> Content-Type: application/sdp
>>>>>> Content-Length: *[len]*
>>>>>>
>>>>>> v=0
>>>>>> o=user1 53655765 2353687637 IN IP*[local_ip_type]* *[local_ip]*
>>>>>> s=-
>>>>>> t=0 0
>>>>>> c=IN IP*[media_ip_type]* *[media_ip]*
>>>>>> m=audio *[media_port]* RTP/AVP 0
>>>>>> a=rtpmap:0 PCMU/8000
>>>>>>
>>>>>> Is it same or different?
>>>>>>
>>>>>
>>>>> They are not the same thing. The first is a xml tag used to instruct
>>>>> sipp to wait for a message (recv means receive)
>>>>> The second is actually the contents of a message that you are
>>>>> instructing sipp to send (so actually, it goes inside a xml tag <send/>).
>>>>> Read about xml here http://en.wikipedia.org/wiki/XML
>>>>> A sipp scenario is basically a sequence of <send/. and <recv/>
>>>>> commands. So you send a message like USIN using <send/> and wait for
>>>>> responses like 100/401/407/200 using <recv/>, then you send another
>>>>> message
>>>>> using <send/> and wait for other responses using <recv/> and so on.
>>>>>
>>>>
>>>> There were some typos:
>>>>
>>>>
>>>> They are not the same thing. The first is a xml tag used to instruct
>>>> sipp to wait for a message (recv means receive)
>>>> The second is actually the contents of a message that you are
>>>> instructing sipp to send (so actually, it goes inside a xml tag <send/>).
>>>> Read about xml here http://en.wikipedia.org/wiki/XML
>>>> A sipp scenario is basically a sequence of <send/> and <recv/> commands.
>>>> So you send a message like INVITE using <send/> and wait for responses like
>>>> 100/401/407/200 using <recv/>, then you send another message using <send/>
>>>> and wait for other responses using <recv/> and so on.
>>>>
>>>>
>>>
>>>
>>> --
>>> Thank you with regards,
>>> Gopalakrishnan A.N.
>>> VoIP call - sip:sai...@gtalk2voip.com
>>>
>>>
>>>
>>
>>
>> --
>> Thank you with regards,
>> Gopalakrishnan A.N.
>> VoIP call - sip:sai...@gtalk2voip.com
>>
>>
>>
>
>
> --
> Thank you with regards,
> Gopalakrishnan A.N.
> VoIP call - sip:sai...@gtalk2voip.com
>
>
>
------------------------------------------------------------------------------
What You Don't Know About Data Connectivity CAN Hurt You
This paper provides an overview of data connectivity, details
its effect on application quality, and explores various alternative
solutions. http://p.sf.net/sfu/progress-d2d
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