Oops... I've tried to run your scenario and realized that I haven't compiled sipp at my new Windows laptop under cygwin yet (and the one saved from the old one asks for old libraries not available any more by peaceful means). So it won't be fast I'm afraid as I never did that before, last time a colleague did that for me who is not a colleague any more.

P.

Dne 26.03.2020 v 1:25 sshark wsk napsal(a):
Hi Alberto,

Thank you, honestly I could not find any difference from the line you have used. I tried it with below and my sipp does not go the label after timeout <recv response="200" rtd="true" rrs="true" timeout="2000" ontimeout="BYE_AND_REGISTER">

I agree to your comment about sending CANCEL instead of BYE

I even compiled SIPp yesterday and tried, still same. Running on Ubuntu 18.04, hope it's not a reason :)
SIPp v3.6.0-23-g4dad7f2-SCTP-PCAP-RTPSTREAM

@Pavel Sindelka <mailto:sindelk...@gmail.com> - Were you able to test my scenario file and see if it worked or not...

Thanks for all the help
sshark


On Tue, Mar 24, 2020 at 11:29 PM Alberto Valente <alberto-r-vale...@telecom.pt <mailto:alberto-r-vale...@telecom.pt>> wrote:

    I had the idea of having a similar scenario some time ago. I found
    time to look for it and tested it in version 3.6.0 and it works
    (it just doesn’t unregisters the calling subscriber).

    I run it with  sudo sipp -sf ims/regInvCancel.xml -i
    source-ip-address -inf csv/csvfile.csv remote-address:remote-port -m 1

    The csv file has a line (may have more) like:

    +351123456789;[authentication username=+351123456789
    password=somepassword];somefqdn.com
    <http://somefqdn.com>;3600;+351987654321;+351299999999;

    +351123456789 – Subscriber to register and originate call

    [authentication username=+351123456789 password=somepassword –
    Authentication user and password

    somefqdn.com <http://somefqdn.com> – subscriber realm

    3600: time to use for register duration

    ;+351987654321 . Number to call

    Just one note. If call is not answered send BYE is not the right
    option.  A CANCEL request shall be used instead.

    Best regards;

    <http://altice.pt/>

    *Alberto Valente*
    DEO/EFV - Core Network and Service platforms Engineering

    +351 963288480 / +351 215000290
    alberto.r.vale...@telecom.pt <mailto:alberto.r.vale...@telecom.pt>
    Rua Tomás Ribeiro Nº2
    1050-229 LISBOA
    *altice.pt* <http://altice.pt>

    facebook <https://www.facebook.com/alticeportugal>linkedin
    <https://www.linkedin.com/company/altice-portugal/>twitter
    <http://twitter.com/altice_portugal>sapovideos
    <http://videos.sapo.pt/alticeportugal>youtube
    <http://www.youtube.com/alticeportugal>

    AVISO DE CONFIDENCIALIDADE
    Esta mensagem e quaisquer ficheiros anexos a ela contêm informação
    confidencial, propriedade da Altice Portugal e/ou das demais
    sociedades que com ela se encontrem em relação de domínio,
    Fundação Altice Portugal e ACS, destinando-se ao uso exclusivo do
    destinatário. Se não for o destinatário pretendido, não deve usar,
    distribuir, imprimir ou copiar este e-mail. Se recebeu esta
    mensagem por engano, por favor informe o emissor e elimine-a
    imediatamente.
    Obrigado

    *From:*Šindelka Pavel [mailto:sinde...@ttc.cz
    <mailto:sinde...@ttc.cz>]
    *Sent:* terça-feira, 24 de março de 2020 07:24
    *To:* sshark wsk <sshark...@gmail.com <mailto:sshark...@gmail.com>>
    *Cc:* sipp-users@lists.sourceforge.net
    <mailto:sipp-users@lists.sourceforge.net>
    *Subject:* Re: [Sipp-users] ontimeout label does not work - sipp 3.6

    Please post the complete scenario. I'll run it using an older SIPp
    version on which my own scenarios with |ontimeout| work properly.

    P.

    Dne 24.03.2020 v 0:46 sshark wsk napsal(a):

        Hi Pavel,

        Thanks for your email

        1. I am trying this flag for the first time. I have not tried
        it with earlier SIPp versions. I just provided what version I
        am testing on as I saw some threads showing that older
        versions do not support this functionality.

        2. I tried with optional recv lines removed, still same. It
        timeout but it is treated as failed rather than jumping to
        de-register sequence.

        3. I have entire scenario in one file (reg, invite/answer,
        dereg) and using the same call_id. I can see the registration
        & call setup progresses until 180

        BR,

        sshark

        On Tue, Mar 24, 2020 at 2:37 AM Šindelka Pavel
        <sinde...@ttc.cz <mailto:sinde...@ttc.cz>> wrote:

            Hi Sshark,

            1. you've indicated a particular version of SIPp; have you
            done that because your scenario did work in previous
            versions and with this one it stopped, or did you just
            want to provide as much information as possible but you
            actually haven't tried this particular scenario with any
            previous SIPp version?

            2. I remember I had some trouble to make ontimeout work in
            one of the older releases (and I haven't changed the
            scenarios since, so don't read it as "and it worked
            flawlessly in the never ones"), but I cannot remember the
            details (except that even with the |ontimeout| handling
            defined next to the |timeout| specification, a call where
            a timeout has elapsed is always counted as a failed one
            which is sometimes not desirable, but you're not that far
            yet). So can you try without the |<recv response="xxx"
            optional="true">| between the |<send>| and _<recv
            response="200" ontimeout="do_something" ... > ? It's not a
            solution, it is a diagnostic step, but I'm afraid this
            type of setup (optional messages and one mandatory one
            with a timeout specified) may not have been addressed in
            the SIPp code.

            3. I hope you don't actually send |REGISTER| and |INVITE|
            with the same |Call-ID| value, as a normal SIP stack would
            fail on that. This one is not related to |ontimeout|, it's
            related to how SIP itself works.

            Pavel

            Dne 23.03.2020 v 13:43 sshark wsk napsal(a):

                Hi Alberto,

                Thank you for your suggestion. I have tried with below
                flag. However still the it does not work

                <recv response="200" crlf="true" timeout="3000"
                ontimeout="3">

                Is it possible to do some debug mode/prints to see why
                it does not work

                BR,

                Sshark

                On Mon, Mar 23, 2020 at 8:01 PM Alberto Valente
                <alberto-r-vale...@telecom.pt
                <mailto:alberto-r-vale...@telecom.pt>> wrote:

                    I think you must use something like

                    <recv response="200" crlf="true" timeout="xxxxxx"
                    ontimeout="BYE_AND_DEREGISTER">

                    Where xxxxxx stands for the number of milliseconds
                    you want to wait for the 200 OK Mresponse

                    Best regards

                    <http://altice.pt/>

                    *Alberto Valente*
                    DEO/EFV - Core Network and Service platforms
                    Engineering

                    +351 963288480 / +351 215000290
                    alberto.r.vale...@telecom.pt
                    <mailto:alberto.r.vale...@telecom.pt>
                    Rua Tomás Ribeiro Nº2
                    1050-229 LISBOA
                    *altice.pt* <http://altice.pt>

                    facebook
                    <https://www.facebook.com/alticeportugal>linkedin
                    <https://www.linkedin.com/company/altice-portugal/>twitter
                    <http://twitter.com/altice_portugal>sapovideos
                    <http://videos.sapo.pt/alticeportugal>youtube
                    <http://www.youtube.com/alticeportugal>

                    AVISO DE CONFIDENCIALIDADE
                    Esta mensagem e quaisquer ficheiros anexos a ela
                    contêm informação confidencial, propriedade da
                    Altice Portugal e/ou das demais sociedades que com
                    ela se encontrem em relação de domínio, Fundação
                    Altice Portugal e ACS, destinando-se ao uso
                    exclusivo do destinatário. Se não for o
                    destinatário pretendido, não deve usar,
                    distribuir, imprimir ou copiar este e-mail. Se
                    recebeu esta mensagem por engano, por favor
                    informe o emissor e elimine-a imediatamente.
                    Obrigado

                    *From:*sshark wsk [mailto:sshark...@gmail.com
                    <mailto:sshark...@gmail.com>]
                    *Sent:* sábado, 21 de março de 2020 04:17
                    *To:* sipp-users@lists.sourceforge.net
                    <mailto:sipp-users@lists.sourceforge.net>
                    *Subject:* [Sipp-users] ontimeout label does not
                    work - sipp 3.6

                    Dear Users,

                    I have sipp version - SIPp
                    v3.6-dev-112-g8c0e358-TLS-PCAP-RTPSTREAM

                    I have prepared a UAC scenario where

                    1. User sends SIP registration

                    2. Gets 401/200 OK and registers

                    3. Sends INVITE

                    3. Waits for 100/180/200

                    4. Sends ACK

                    5. Pauses for 4s and sends BYE

                    6. De-register the user

                    In step 3, I would like to timeout if User B does
                    not answer (not receiving 200 OK within a certain
                    time) with below

                    <recv response="200" ontimeout="BYE_AND_DEREGISTER">

                     Unfortunately above does not work

                    I run SIPp with below flags

                    ./sipp 192.168.1.1 -bind_local -i 10.10.10.1 -mp
                    50000 -t u1 -p 5060 -recv_timeout 3200 -sf
                    register_and_call.xml -inf USER_A.csv -m 1 -trace_msg

                    Hope you can assist me

                    Thank You

                    sshark




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