[SIPForum-discussion] Load call test for sip audio calls with DTMF

[Manish Anand]

Hi All,

Here is my invite and  prack request..


INVITE sip:14312011234 at 160.144.33.14:45090;transport=TCP SIP/2.0
To: <sip:+14312012222 at ims.telus.com>
Content-Type: application/sdp
Via: SIP/2.0/TCP 160.144.33.14:25090
;branch=z9hG4bK-23293-1-0;received=101.184.40.192
Call-ID: 1-23293 at 160.144.33.14
Max-Forwards: 70
From: <sip:+14312011111 at mt.com>;tag=1
Contact: <sip:160.144.33.14:25090;transport=TCP>
Wlss-Popped-Route: <sip:callchoke at 101.184.40.193:35060;lr;transport=TCP>
CSeq: 1 INVITE
Content-Length: 138
Supported: 100rel
Require: 100rel
P-Asserted-Identity: <sip:14312012222 at ims.com>
P-Asserted-Identity: <tel:+14312012222>
v=0
o=user1 53655765 2353687637 IN IP4 169.144.33.14
s=-
c=IN IP4
169.144.33.14
t=0 0
m=audio 6001 RTP/AVP 0
a=rtpmap:0 PCMU/8000


[PRACK sip:callchoke at 101.184.40.193:35060 SIP/2.0
To: <sip:+14312012222 at im.com>;tag=23230SIPpTag011
Via: SIP/2.0/TCP 169.144.33.14:25090
;branch=z9hG4bK-10999-1-3;received=10.184.40.192
CSeq: 2 PRACK
Content-Length: 0
Subject: Performance Test
Route: <sip:callchoke at 10.184.40.193:35060;lr;transport=TCP>
Call-ID: 1-23293 at 169.144.33.14
Max-Forwards: 70
RAck: 2 1 INVITE
From: <sip:+143120111111 at mt.com>;tag=1
Contact: <sip:169.144.33.14:25090;transport=TCP>

Thanks,

Manish Anand



On Tue, Mar 10, 2015 at 11:05 AM, Mukul Jain <er.mukuljain at gmail.com> wrote:

> Are you creating prack using wlss patch?
> On 09-Mar-2015 11:02 pm, "Manish Anand" <manishanand.t at gmail.com> wrote:
>
>> Hi All,
>>
>>
>> I am facing one issue, when i am sending prack message through sipp tool,
>> In Sipp message log message is printing correctly like
>>
>> PRACK sip:127.0.0.1:35060;transport=UDP SIP/2.0
>> Via: SIP/2.0/UDP 127.0.0.1:25060;branch=z9hG4bK-14036-1-3
>> CSeq: 2 PRACK
>> To: <sip:+4032011234 at im.com>;tag=1
>> P-Early-Media: sendonly
>> Allow: INVITE,ACK,PRACK,BYE
>> Call-ID: 1-14036 at 127.0.0.1
>> From: <sip:+4032014321 at mt.com>;tag=1
>> Max-Forwards: 70
>> RAck: 1 INVITE
>> Route: <sip:127.0.0.1:5060;lr>
>> Contact: <sip:127.0.0.1:25060>
>> Content-Length:     0
>>
>> but in sip servlet in am getting error in doPrack method.
>>
>> *java.lang.IllegalArgumentException: cseq number must be positive*
>>
>>
>> Please suggest where i am doing mistake.
>>
>>
>>
>> On Sat, Mar 7, 2015 at 1:01 PM, Saurabh Shah <
>> saurabh.shah at matrixcomsec.com> wrote:
>>
>>> Use SIPP.
>>>
>>> On Fri, Feb 20, 2015 at 3:38 PM, Antonio Manchon Romero <
>>> antoniomanchonromero at yahoo.es> wrote:
>>>
>>>> Hi everybody,
>>>>
>>>> I´m looking for a free tool to test a new environment. It would be for
>>>> audio sip calls only. Codecs: G711, G729 and dtmf capabilities would be
>>>> required.
>>>>
>>>> This software would be able to launch and maintain automatically
>>>> several calls. With this software I would like to launch calls
>>>> progressively to the new environment to test system delays and performance.
>>>>
>>>> After setup, software would have to send two DMTF sequences in
>>>> communication with an IVR. So it would have to send first sequence of
>>>> digits and after a known delay the second one.
>>>>
>>>> Thanks for your help,
>>>>
>>>> Antonio
>>>>
>>>>
>>>>
>>>> _______________________________________________
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>>>> Post to the list at discussion at sipforum.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Regards,
>>> Saurabh Shah
>>>
>>> _______________________________________________
>>> This is the SIP Forum discussion mailing list
>>> TO UNSUBSCRIBE, or edit your delivery options, please visit
>>> http://sipforum.org/mailman/listinfo/discussion
>>> Post to the list at discussion at sipforum.org
>>>
>>>
>>
>>
>> --
>> Thanks and Regards
>>
>> Manish Anand
>>
>> _______________________________________________
>> This is the SIP Forum discussion mailing list
>> TO UNSUBSCRIBE, or edit your delivery options, please visit
>> http://sipforum.org/mailman/listinfo/discussion
>> Post to the list at discussion at sipforum.org
>>
>>


-- 
Thanks and Regards

Manish Anand
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