Re: [Sipp-users] UAC with auth (Github sipp-samples/uac-auth.xml): SiPp ignores 401

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Thanks Pavel for replying.

Thanks to your advice, I restarted all over, using another Asterisk client
instance as a guide to a canonical REGISTER dialog.
When I got a successful REGISTER dialog between 2 Asterisk instances, I
stopped Asterisk on client host and used SIPp instead.
I progressed step by step to the point I saw my Asterisk server replying
with a 200OK to the REGISTER !

I'm very happy with this result but the issue is that this REGISTER dialog
is repeated many times while I would like to play it only once.
How can I do that ?

SIPp is invoked with:
sipp 192.168.64.46:5062 -sf /home/foobar/my-uac-auth.xml -ap passsipp -s
sipp -i 192.168.64.45 -p 5062

My /home/foobar/my-uac-auth.xml
file is now:

<scenario name="Basic Sipstone UAC">
  <!-- In client mode (sipp placing calls), the Call-ID MUST be         -->
  <!-- generated by sipp. To do so, use [call_id] keyword.
-->

  <send retrans="500">
    <![CDATA[

      REGISTER sip:[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: <sip:[service]@[remote_ip]>;tag=[pid]SIPpTag00[call_number]
      To: <sip:[service]@[remote_ip]>
      Call-ID: [call_id]
      CSeq: 1 REGISTER
      Contact: <sip:s@[local_ip]:[local_port]>
      Max-Forwards: 70
      Subject: Performance Test
      Content-Type: application/sdp
      Content-Length: [len]
    ]]>
  </send>

  <recv response="200"
        optional="true"
        next="auth_done">
  </recv>

  <recv response="401"
        auth="true">
  </recv>

  <send retrans="500">
    <![CDATA[

      REGISTER sip:[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: <sip:[service]@[remote_ip]>;tag=[pid]SIPpTag00[call_number]
      To: <sip:[service]@[remote_ip]>
      Call-ID: [call_id]
      CSeq: 1 REGISTER
      Contact: <sip:s@[local_ip]:[local_port]>
      Max-Forwards: 70
      [authentication username=sipp password=passsipp]
      Subject: Performance Test
      Content-Type: application/sdp
      Content-Length: [len]
    ]]>
  </send>

  <recv response="200">
  </recv>

  <label id="auth_done" />

  <pause milliseconds="5000"/>

</scenario>

Here is a trace of corresponding REGISTER dialog (captured on Asterisk and
received on SIPp host):

<--- SIP read from UDP:192.168.64.45:5062 --->
REGISTER sip:192.168.64.46:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.64.45:5062;branch=z9hG4bK-19512-22-0
From: <sip:sipp@192.168.64.46>;tag=19512SIPpTag0022
To: <sip:sipp@192.168.64.46>
Call-ID: 22-19512@192.168.64.45
CSeq: 1 REGISTER
Contact: <sip:s@192.168.64.45:5062>
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 192.168.64.45:5062 (no NAT)
Sending to 192.168.64.45:5062 (no NAT)

<--- Transmitting (no NAT) to 192.168.64.45:5062 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.64.45:5062
;branch=z9hG4bK-19512-22-0;received=192.168.64.45
From: <sip:sipp@192.168.64.46>;tag=19512SIPpTag0022
To: <sip:sipp@192.168.64.46>;tag=as41ad7250
Call-ID: 22-19512@192.168.64.45
CSeq: 1 REGISTER
Server: Asterisk PBX 13.23.1~dfsg-1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="canonical", nonce="3386ce74"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '22-19512@192.168.64.45' in 32000 ms
(Method: REGISTER)

<--- SIP read from UDP:192.168.64.45:5062 --->
REGISTER sip:192.168.64.46:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.64.45:5062;branch=z9hG4bK-19512-22-3
From: <sip:sipp@192.168.64.46>;tag=19512SIPpTag0022
To: <sip:sipp@192.168.64.46>
Call-ID: 22-19512@192.168.64.45
CSeq: 1 REGISTER
Contact: <sip:s@192.168.64.45:5062>
Max-Forwards: 70
Authorization: Digest username="sipp",realm="canonical",uri="sip:
192.168.64.46:5062
",nonce="3386ce74",response="b0388098db7f3fd69dbd4c8a030d8d28",algorithm=MD5
Subject: Performance Test
Content-Type: application/sdp
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.64.45:5062 (no NAT)

<--- Transmitting (no NAT) to 192.168.64.45:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.64.45:5062
;branch=z9hG4bK-19512-22-3;received=192.168.64.45
From: <sip:sipp@192.168.64.46>;tag=19512SIPpTag0022
To: <sip:sipp@192.168.64.46>;tag=as41ad7250
Call-ID: 22-19512@192.168.64.45
CSeq: 1 REGISTER
Server: Asterisk PBX 13.23.1~dfsg-1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 120
Contact: <sip:s@192.168.64.45:5062>;expires=120
Date: Thu, 25 Oct 2018 16:08:24 GMT
Content-Length: 0

Any clue ?

Le jeu. 25 oct. 2018 à 13:26, Šindelka Pavel <sin...@tt...> a écrit :

> Hi Olivier,
>
> looking at your command line with -m=1 and at the scenario, I suppose
> that the repeated REGISTER requests without the authentication header you
> can see are retransmissions of the initial one, implying that the sipp
> machine/process does not receive the responses from the Asterisk.
>
> This can have a number of reasons:
>
>    - wrong population of the headers in the messages generated from the
>    scenario (should not be the case as you've specified -i on the command
>    line)
>    - routing issue (unlikely unless you've intentionally split
>    192.168.64.0/24 into several subnets or misconfigured the network mask
>    on either machine unintentionally)
>    - Asterisk configuration issue (not permitting incoming registrations
>    from this address/subnet)
>    - firewall issue on either machine
>
> So SIPp logs, Asterisk logs, and tcpdump/Wireshark are your best friends.
> See whether the REGISTER arrives to the Asterisk, what is its contents, and
> whether the Asterisk responds at all and where it sends the responses.
>
> Pavel
>
> Dne 25.10.2018 v 11:17 Olivier napsal(a):
>
> Hello,
>
> I'm quite new to SIPp.
> I've just discovered [1].
> I'm testing this uac-auth.xml file with the bellow command against an
> Asterisk instance:
>
> sipp -sf uac-auth.xml 192.168.64.250 -au 7005 -ap 7005 -s 7005 -i
> 192.168.64.45 -m 1
>
> I see that Asterisk challenges incoming REGISTER with a WWW-Authenticate
> but SIPp does not reply with any new REGISTER with an Authorization header.
> Instead, it keeps sending first REGISTER.
>
> 1. Am I correct to expect, with referenced uac-auth.xml, SIPp to send a
> REGISTER with an Authorization header ?
>
> 2. If negative, what should be changed to in uac-auth.xml to implement
> this ? If positive, is it correct to expect [authentication] lines in a
> REGISTER to be replaced with an Authorization built with data coming from
> matching 401 reply (nonce, realm, ...) ?
>
> Best regards
>
> [1] https://github.com/rkday/sipp-samples/blob/master/uac-auth.xml
>
>
>
>
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