Re: [Sipp-users] Troubleshooting a basic call

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Thank you for the help!  This is indeed my base case copy+pasted, just calling into voicemail and then either voicemail hangs up or I hang up.

I added the modifications you suggested but I had to add </recv> after each recv tag to have the program not throw an error of: (The label 'auth_challenge_received' was not defined (index 2, next attribute)).  However now the scenario only lists an invite and 100 message, not the full scenario that I was expecting.

Interestingly the INVITE lists a bad UDP checksum.  That might help explain why I'm not getting a response back from anything.

12:35:17.894298 IP (tos 0x0, ttl 64, id 62100, offset 0, flags [DF], proto UDP (17), length 891)
    localhost.localdomain.sip > 10.20.128.149.sip: [bad udp cksum 0x128e -> 0xb0db!] SIP, length: 863
        INVITE sip:17120@10.20.128.149;user=phone;transport=tcp SIP/2.0
        Via: SIP/2.0/UDP 129.82.3.26;branch=z9hG4bK-17371-1-0
        From: "Robert Remsik" <sip:17...@ot...>;tag=1
        To: <sip:17120@10.20.128.149;user=phone>
        CSeq: 1 INVITE
        Call-ID: 1-17371@129.82.3.26
        Contact: <sip:17120@129.82.3.26;transport=UDP>
        Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
        User-Agent: PolycomVVX-VVX_410-UA/4.1.8.0628
        Accept-Language: en
        Supported: 100rel,replaces
        Allow-Events: conference,talk,hold
        Max-Forwards: 70
        Content-Type: application/sdp
        Content-Length:   245

        v=0
        o=- 1470166919 1470166919 IN IP4 129.82.3.26
        s=Polycom IP Phone
        c=IN IP4 129.82.3.26
        t=0 0
        a=sendrecv
        m=audio 6000 RTP/AVP 9 0 8 101
        a=rtpmap:9 G722/8000
        a=rtpmap:0 PCMU/8000
        a=rtpmap:8 PCMA/8000
        a=rtpmap:101 telephone-event/8000

                                 Messages  Retrans   Timeout   Unexpected-Msg
      INVITE ---------->         1         0
         100 <----------         0         0         0         0

  <recv response="100" optional="true"/>
  </recv>

  <recv response="401" optional="true" auth="true" next="auth_challenge_received"/>
  </recv>

  <recv response="407" auth="true"/>
  </recv>

  <label id="auth_challenge-received"/>
  </recv>

Robert Remsik

ACNS

Desk Phone: 970 491 7120

Rob...@co...

________________________________
From: sindelka <sin...@tt...>
Sent: Tuesday, August 2, 2016 9:12 PM
To: sip...@li...
Subject: Re: [Sipp-users] Troubleshooting a basic call

Hi Robert,

if the scenario you've provided is a verbatim copy of the one you
actually use, not an edited version for illustration purposes, the
trouble is that you blindly fire a series of requests without waiting
for any response from the server. Normally, the authentication challenge
comes in a 401 or 407 response to the first INVITE, and the generation
of the authentication response as requested by the [authentication]
keyword depends on the information from the authentication challenge.

So you must insert some optional and mandatory <recv> statements between
the first INVITE and first ACK - namely,

<recv response="100" optional="true"/>

<recv response="401" optional="true" auth="true"
next="auth_challenge_received"/>

<recv response="407" auth="true"/>

<label id="auth_challenge-received"/>

This ensures that regardless whether the sipx device sends the 100 or
not, and regardless whether it uses 401 or 407 to send the
authentication challenge, you'll handle it properly.

Also, remove the auth="true" AVPs from all the <send> blocks, they are
useless there.

On top of that, please <recv> optionally a 100, a 180, a 183 and then
compulsorily a 200 between sending the second INVITE (with the
authentication response) and sending the second ACK, otherwise you may
run into some other issues.

And the last point, if you specify user and password inside the
scenario, SIPp does not use the -au and -ap specified on the command line.

Pavel

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