Hi Paulo,

that's unbelievable. Can you attach your UAC scenario and the pcap you are replaying, and the complete pcap of the result as taken on the Asterisk machine?

As a workaround, I'd suggest to use a dirty trick, allowing you to convert your UAC scenario into an UAS one as seen by SIPp while maintaining its functionality as seen by the Asterisk. To do so, insert the following before the initial

|<send retrans="500">||
||  <![CDATA[||
||    INVITE ...|

of your UAC scenario:

<recv request="INVITE" optional="true" />

<recvCmd/>

Then, create a "trigger" scenario:

|<scenario name="3pcc trigger">||
|

|  <send>||
||    <![CDATA[||
||||      INFO||
||    ]]>||
||  </send>||
|

|  <sendCmd>||
||    <![CDATA[||
||      Call-ID: [call_id]||
||    ]]>||
||  </sendCmd>||

|| <pause milliseconds="whatever_the_duration_of_your_test_call + 10000"/>||
||
</scenario>|

Then, run your modified UAC scenario first, with an additional command line parameter |-3pcc localhost:9001| and without the parameters controlling the call rate and number of calls (|-r|, |-rp|, |-m|). As it is an UAS one now, it will start and do nothing spontaneously, just waiting for external stimuli to come. Next, run the trigger scenario, with the same|-3pcc localhost:9001| command line parameter and the parameters controlling the call rate and number of calls, and with destination address |localhost:xxxx| (where |xxxx| is a UDP port at which nothing is listening at the machine).

What will happen: by adding the |<recv>| to the beginning of your originally UAC scenario, we've made it an UAS one, so the |pcapplay| should start working if the UDP checksum issue really depends on UAS/UAC mode of the scenario. A UAS scenario does not generate call IDs and initiate calls, so we must provide that using the trigger scenario. In order to make SIPp treat the trigger scenario as an UAC one, its first statement must be |<send>|, so we send a bogus SIP message to a blackhole first. Right after, the trigger scenario sends a 3pcc command to the main one. Reception of the 3pcc command triggers the execution of the rest of the main scenario, which uses the call ID value extracted from the received command. The pause in the trigger scenario is necessary because the call in the trigger scenario must not finish before it finishes in the main one.

Hope that helps get you up and running.

Pavel

Dne 29.10.2016 v 16:39 Paulo R. Vieira Jr napsal(a):
Hi,

I made a role`s swap, the machine that was UAS became UAC, and the machine that was UAC became UAS, and the packets comming from UAS arrive correctly to asterisk and asterisk send to UAC, but the packets comming from UAC aways had a bad checksum, so i can only simulate one way audio to the load test. Seens like the problem is in the method/procedure that sends packets from UAC if not the same that sends from UAS.

Thanks.

2016-10-29 8:11 GMT-02:00 sindelka <sinde...@ttc.cz <mailto:sinde...@ttc.cz>>:

    Sorry, I was too quick when reading your initial message because
    99% of "bad UDP checksum" cases are resolved this way.

    So reading it more carefully now, when you've mentioned trying
    different machines, did you try to swap the roles of the two
    machines running the UAC and UAS scenarios? I.e. can you confirm
    that if you run the UAC scenario with its RTP-containing pcap on
    the machine which replays its own RTP-containing pcap properly if
    the replay is spawned from an UAS scenario, the packets come to
    the Asterisk with a bad UDP checksum as well? And, vice versa, do
    the RTP packets sent from an UAS scenario running on the
    previously UAC machine have a correct UDP checksum on arrival to
    the Asterisk machine?

    If both answers are "yes", this would mean that the behaviour
    depends on whether the pcapplay is executed from an UAC or an UAS
    scenario. This is hard to believe, yet it would allow to suggest a
    workaround.

    Pavel


    Dne 28.10.2016 v 18:59 Paulo R. Vieira Jr napsal(a):
    Hi, thanks for response.

    They are all (uas, asterisk and uac) on different machines. The
    packets were captured on asterisk server. The packets are not
    being delivered to asterisk, with "rtp set debug on" no packets
    are seen.
    The server stays 98% 99% idle.
    The firewall is disable, the ports are correct and sip exchange
    messages too. i have no idea what to do...

    Thanks.

    2016-10-28 14:47 GMT-02:00 sindelka <sinde...@ttc.cz
    <mailto:sinde...@ttc.cz>>:

        Hi Paulo,

        on which machine did you run Wireshark/tcpdump? The thing is
        that the
        UDP/TCP/IP checksums are often wrong in the capture if
        capturing on the
        sending machine because the frame is captured before the
        checksum is
        actually calculated. So it may be just a red herring and the
        actual
        problem may be somewhere else.

        Pavel


        
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    resurgence of CLI tooling catch you by surprise? Reconnect with
    the command line and become more productive. Learn the new .NET
    and ASP.NET <http://ASP.NET> CLI. Get your free copy!
    http://sdm.link/telerik
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