[Freeswitch-users] SIPp issues - seems FS doesn't understand ACK message
Anthony Minessale
anthony.minessale at gmail.com
Tue Aug 25 13:39:21 PDT 2009
here's the one i use for making a call waiting x seconds and hanging up
http://www.freeswitch.org/eg/load_test/dft_cap.xml
This requires that the sipp terminate all the calls.
careful with sipp, it's like a roach motel, you can get stuck trying to make
it work and never get it to produce real-life situations.
On Tue, Aug 25, 2009 at 3:23 PM, Bradley Brashier <bjbrashier at gmail.com>wrote:
> Well, you'd have another nickel from over here, then.
> If I can get this working before I'm tasked with something else I'll write
> up something more on the wiki about "Freeswitch and SIPp", but I'm not sure
> I'll get that chance.
>
> BB
>
> On Tue, Aug 25, 2009 at 11:05 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> I wish I had a nickel for every guy struggling with sipp load testing vs
>> real world traffic.
>>
>>
>>
>> On Tue, Aug 25, 2009 at 1:51 AM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>
>>> Hello Takeshi,
>>>
>>> Thanks for your hint... it worked out... so to be precise:
>>>
>>> VIA header of both INVITE and ACK messages MUST be identical (IP:PORT +
>>> branch)... and you are right... it might not be according to SIP
>>> specification. Anyhow, i get FS understand my ACK message.
>>>
>>>
>>> Finally, here is what i used and I'm getting some poor results .. but
>>> this is another topic :)
>>>
>>>
>>> Thanks for your help.
>>> Tihomir.
>>>
>>>
>>> sipp 10.4.4.251 -sf uac_redirect.xml -s 30003016094191500 -trace_err -r 1
>>> -rp 100 -trace_msg -inf test.txt -m 20000 -l 4000
>>>
>>>
>>> <?xml version="1.0" encoding="ISO-8859-1" ?>
>>> <!DOCTYPE scenario SYSTEM "sipp.dtd">
>>>
>>>
>>> <scenario name="Basic Sipstone UAC">
>>> <send retrans="500" start_rtd="1" start_rtd="2">
>>>
>>> <![CDATA[
>>>
>>> INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
>>> Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
>>> Max-Forwards: 70
>>> Contact: <sip:[field1]@[local_ip]>
>>> From: [field1]
>>> <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
>>> To: [service] <sip:[service]@[remote_ip]:[remote_port]>
>>> Call-ID: [call_id]
>>> CSeq: 1 INVITE
>>> 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 response="100"
>>> optional="true" rtd="1">
>>> </recv>
>>>
>>>
>>> <recv response="302" rtd="2">
>>> </recv>
>>>
>>> <send>
>>> <![CDATA[
>>>
>>> ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
>>> Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
>>> From: [field1]
>>> <sip:[field1]@1[local_ip]:[local_port]>;tag=[call_number]
>>> To: [service]
>>> <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
>>> Call-ID: [call_id]
>>> CSeq: 1 ACK
>>> Max-Forwards: 70
>>> 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>
>>>
>>>
>>>
>>> On Tue, Aug 25, 2009 at 3:57 AM, mayamatakeshi <mayamatakeshi at gmail.com>wrote:
>>>
>>>> On Tue, Aug 25, 2009 at 10:52 AM, mayamatakeshi<mayamatakeshi at gmail.com>
>>>> wrote:
>>>> > On Tue, Aug 25, 2009 at 7:31 AM, Tihomir Culjaga<tculjaga at gmail.com>
>>>> wrote:
>>>> >>
>>>> >> sipp_cmd: sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s
>>>> >> 30003016094191500 -trace_err -r 1 -rp 1000 -trace_msg -inf test.txt
>>>> -m 1 -l
>>>> >> 4000
>>>> >> scenario file: uac_redirect.xml
>>>> >> FS dialplan: public.xml
>>>> >> SIP trace: trace.log
>>>> >
>>>> > The Via definition in your SIPp scenario differs between the INVITE
>>>> and the ACK:
>>>> >
>>>> > INVITE:
>>>> > Via: SIP/2.0/[transport] [local_ip];branch=[branch]
>>>> >
>>>> > ACK:
>>>> > Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
>>>> >
>>>> >
>>>> > In the INVITE, you are not adding the [local_port] as you do in the
>>>> ACK.
>>>> > Just adding the [local_port] in the INVITE makes FreeSWITCH accept the
>>>> ACK.
>>>> > So it seems FS is not checking just the ACK's branch against the
>>>> > INVITE's; it seems it is checking the whole Via header.
>>>> > I don't know if this is in accordance to SIP specs.
>>>> > Another thing, about the way you are calling SIPp: do no use "-sn uac"
>>>> > and "-sf uac_redirect.xml" at the same time. The parameter "-sn xxx"
>>>> > means "use the internal (embedded) scenario named xxx". So this
>>>> > conflicts with the other parameter "-sf" which specifies an external
>>>> > profile.
>>>>
>>>> I mean, an external scenario (file).
>>>>
>>>> It seems this doesn't cause any problem (probably because in
>>>> > the sipp startup, -sf overrides -sn), but it is misleading.
>>>> >
>>>> > regards,
>>>> > takeshi
>>>> >
>>>>
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>>>
>>>
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>>>
>>
>>
>> --
>> Anthony Minessale II
>>
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>>
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>> IRC: irc.freenode.net #freeswitch
>>
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>>
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>
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>
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
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Twitter: http://twitter.com/FreeSWITCH_wire
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