[Freeswitch-users] working sipp scenario for testing kamailio+FS with authorization

[Борисов, Дмитрий / Dmitriy Borisov]

Have anyone subj? I`m look to RFC and see one, but when i`m looking to sip
trace I see another... And it works on SIP-clients, but not on my scenario
(in attach)

<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">

<scenario name="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" start_txn="invite">
    <![CDATA[

INVITE sip:[field3]@[remote_ip] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:[field0]@[field1]:[local_port]>;tag=[pid][call_number]
To: [field3] <sip:[field3]@[field1]>
Call-ID: [call_id]
CSeq: [cseq] INVITE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]

v=0
o=[field0] 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" response_txn="invite"/>

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

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

  <send ack_txn="invite">
    <![CDATA[

ACK sip:[field3]@[remote_ip] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:[field0]@[field1]:[local_port]>;tag=[pid][call_number]
To: [field3] <sip:[field3]@[field1]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: [cseq] ACK
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: [len]

v=0
o=[field0] 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>

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

INVITE sip:[field3]@[remote_ip] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:[field0]@[field1]:[local_port]>;tag=[pid][call_number]
To: [service] <sip:[field0]@[field1]>
Call-ID: [call_id]
CSeq: [cseq] INVITE
Contact: sip:sipp@[local_ip]:[local_port]
[field2]
Max-Forwards: 70
Subject: Performance Test
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" response_txn="invite">
  </recv>

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

  <recv response="183" optional="true" response_txn="invite"/>

  <!-- By adding rrs="true" (Record Route Sets), the route sets         -->
  <!-- are saved and used for following messages sent. Useful to test   -->
  <!-- against stateful SIP proxies/B2BUAs.                             -->
  <recv response="200" rtd="true" response_txn="invite">
  </recv>

  <!-- Packet lost can be simulated in any send/recv message by         -->
  <!-- by adding the 'lost = "10"'. Value can be [1-100] percent.       -->
  <send ack_txn="invite">
    <![CDATA[

ACK sip:[field3]@[remote_ip] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:[field0]@[field1]:[local_port]>;tag=[pid][call_number]
To: [field3] <sip:[field3]@[field1]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: [cseq] ACK
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0

    ]]>
  </send>

  <pause milliseconds="59000"/>

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

BYE sip:[field0]@[remote_ip]:[remote_port] SIP/2.0
[last_Via:]
[last_From:]
[last_To:];tag=[pid][call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0

    ]]>
</send>

<recv response="200" crlf="true" response_txn="bye"/>

<!--
  <recv request="BYE"/>

  <send retrans="500">
    <![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[pid][call_number]
[last_Call-ID:]
[last_CSeq:]
Max-Forwards: 70
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Type: application/sdp
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>

-- 
with best regards,
Dmitriy Borisov
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