Re: [Sipp-users] Sipx with SIPp

[Wen J. <jun...@gm...>]

Hi, Jeff,
 
I am following your instruction in my scenario. When the INVITE from calling
party arrived to called party, the called party always said "ICMP
Destination Unreachable (Port unreachable), and then in the SIPp of called
party, it claimed no INVITE received all the time.
 
Do you have any experience of that ? Are your both instances working as UAC
mode ?
 
Regards,
 
Jun

  _____  

From: Jeff Wright [mailto:JW...@az...] 
Sent: Friday, December 04, 2009 10:42 PM
To: Wen Jun; sip...@li...
Subject: RE: [Sipp-users] Sipx with SIPp


Jun,
 
I have used sipX a lot in testing our SIP-based products here.  Some of our
tests use SIPp as a UA.
 
Here are two scenario files that, together, establish a basic call between
two instances of SIPp (of course, this assumes you have already registered
SIPp on both sides).  These work with sipX as a proxy (I am using sipX
3.8.1-011585).  I run them side by side in tow different terminal windows,
and execute them one right after the other, running the B side first, then
the A side (this way the B side script is ready to accept the INVITE sent
from the A side script).  You will also need to have the CSV file (also
included in this email) in order to populate the information in the [field]
parameters in the scripts.
 
Here are the actual scenario files:
 
<?xml version="1.0" encoding="ISO-8859-1" ?>
<scenario name="successful_call_single_proxy_sideA">
  <send>
    <![CDATA[
      INVITE sip:[field3]@[field4] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      Max-Forwards: 70
      From: <sip:[field0]@[field1]>;tag=from-[call_number]
      To: <sip:[field3]@[field4]>
      Call-ID: [call_id]
      CSeq: [cseq] INVITE
      Contact: sip:[field0]@[local_ip]:[local_port]
      User-Agent: SIPp/Linux
      Subject: Test 3.2.1.1.7
      Content-Type: application/sdp
      Content-Length: [len]
      v=0
      o=sipp1 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">
  </recv>
  
  <recv response="180" optional="true">
  </recv>  
  
  <recv response="200">
  </recv>
  <send>
    <![CDATA[
      ACK sip:[field3]@[field4] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      [last_From:]
      [last_To:]
      [last_Call-ID:]
      CSeq: [cseq] ACK
      Contact: sip:[field0]@[local_ip]:[local_port]
      User-Agent: SIPp/Linux
      Content-Length: 0
      
    ]]>
  </send>
  
  <pause milliseconds="1000"/>
  
  <send>
    <![CDATA[
      BYE sip:[field3]@[field4] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      [last_From:]
      [last_To:]
      [last_Call-ID:]
      CSeq: [cseq] BYE
      Contact: sip:[field0]@[local_ip]:[local_port]
      User-Agent: SIPp/Linux
      Max-Forwards: 70
      Content-Length: 0
    ]]>
  </send>
  <recv response="200">
  </recv>
    
</scenario>
 
<?xml version="1.0" encoding="ISO-8859-1" ?>
<scenario name="successful_call_single_proxy_sideB">
  <recv request="INVITE">
  </recv>
  <send>
    <![CDATA[
      SIP/2.0 180 UA2 Ringing
      [last_Via:]
      [last_From:]
      [last_To:];tag=to-[call_number]
      [last_Call-ID:]
      [last_CSeq:]
      Contact: sip:[field3]@[local_ip]:[local_port]
      User-Agent: SIPp/Linux
      Content-Length: 0
    ]]>
  </send>
  <pause milliseconds="500"/>
  <send>
    <![CDATA[
      SIP/2.0 200 OK UA2 Answered
      [last_Via:]
      [last_From:]
      [last_To:];tag=to-[call_number]
      [last_Call-ID:]
      [last_CSeq:]
      Contact: sip:[field3]@[local_ip]:[local_port]
      User-Agent: SIPp/Linux
      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 request="ACK">
  </recv>
  <recv request="BYE">
  </recv>
  <send>
    <![CDATA[
      SIP/2.0 200 OK UA2 Goodbye
      [last_Via:]
      [last_From:]
      [last_To:]
      [last_Call-ID:]
      [last_CSeq:]
      Contact: sip:[field3]@[local_ip]:[local_port]
      User-Agent: SIPp/Linux
      Content-Length: 0

    ]]>
  </send>  
</scenario>
 
Here is the CSV file.  You will have to modify the individual fields to
match your domain, and the SIPp usernames and passwords as set up in sipX.
The domain (e.g. test.azteknetworks.net) has to be the same as what you have
it set up for in sipX.
 
SEQUENTIAL
sipp1;test.azteknetworks.net;[authentication username=sipp1
password=sipp1];sipp2;test.azteknetworks.net;[authentication username=sipp2
password=sipp2]
 
Best of luck!
 
Jeffrey Wright

System Test Engineering Manager

Aztek Networks, Inc.

  _____  

From: Wen Jun [mailto:jun...@gm...]
Sent: Fri 12/4/2009 2:18 AM
To: sip...@li...
Subject: [Sipp-users] Sipx with SIPp


Hi, it might be a frequently asked question about how to test SIPX with
SIPp. Does any have successful experience about that ? Appreciated if you
can share some hints to me .
 
I've well done the registration from SIPp UAC to SIPX but the call from SIPp
UAC to SIPX was struck yet.
 
Regards,
 
Jun