Re: [Sipp-users] REGISTER and recv INVITE in same scenario file

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Thank you for the explanation.  That confirms why it works for me when I
split the two portions of the scenario and first run one instance of
sipp and then another.

My problem with that solution is that while I am using sipp for both
client and server, my device under test is a sip proxy in the middle.
That sip proxy will only forward invites to the same ip address/port
that the registration came in on (physically, not just what's specified
in the protocol message/scenario file).  No problem if I force the port
of sipp or leave it as default 5065.  The problem is I want to generate
thousands of registrations/answerers for different phone numbers.  That
means a unique port for each. =20

Rather than start up 1000 sipps each with a different port, I thought I
would use the -m 1000 and -t un option so that each registration would
go out from a different port.  But in order for the invite to be
received on the same port, I can't run another instance of sipp to
receive the invite.  I have to use the same instance and therefore
combine the register and recv invite into one scenario file. =20

Sounds like I'm kind of stuck because the incoming INVITE call_id won't
match the call_id that was used to send the register.  (The "inviter"
does not receive the registration - the DUT handles that.  So the
"inviter" does not know what call_id was used for the regisrtation).

As the doc says:=20

[call_id]  	-  	A call_id identifies a call and is generated by
SIPp for each new call. In client mode, it is mandatory to use the value
generated by SIPp in the "Call-ID" header. Otherwise, SIPp will not
recognise the answer to the message sent as being part of an existing
call.

Maybe I can hack the code for myself and force it to be in Server mode
even though the first action is a send.  Not sure that would even work,
but since I am using unique ports (via -t un) maybe sipp will match the
instance of the call based on the port.  Doubt it.

Any other ideas (I guess just start 1000 sipp processes - thank goodness
its linux and not windows :-)

-----Original Message-----
From: Artur Silveira da Cunha
[mailto:Art...@al...]=20
Sent: Wednesday, July 19, 2006 11:54 AM
To: Edward Russell
Cc: sip...@li...
Subject: Re: [Sipp-users] REGISTER and recv INVITE in same scenario file

Hi,

Your problem comes certainly from the following:

- sipp when it creates each scenario, takes the generated call_id and
stores it to match all incoming replies to affect them to the running
scenario.

In your sample:
 - the Register is send and "sipp" stores the generated call_id
 - in the incoming 200, sipp must match the generated call_id, to affect
it to the scenario,
 - in the incoming Invite, you have a different "call_id", then sipp
can't knows for which scenario it belongs.

That is for me the reason why sipp can't affect the incoming Invites to
your Register.

For the same cases I use the following solution:

- one sipp working in port 5065 that issues Register and waits for the
200 OK, in the xml scenario you indicates another port like 5066.

<?xml version=3D"1.0" encoding=3D"ISO-8859-1" ?> <scenario
name=3D"register_answer"> <send retrans=3D"1000">  <![CDATA[
  REGISTER sip:hal9004.whatever.com SIP/2.0
  Via: SIP/2.0/[transport] =
[local_ip]:[local_port];rport;branch=3D[branch]
  From:
<sip:[field0]@hal9004.whatever.com:[local_port]>;tag=3D[call_number]
  To: <sip:[field0]@hal9004.whatever.com:[local_port]>
  Contact: <sip:[field0]@[local_ip]:5066>
  Call-ID: [call_id]
  CSeq: [cseq] REGISTER
  Expires: 1800
  Max-Forwards: 70
  Content-Length: 0
 ]]>
</send>
=20
<nop><action><log message=3D"sent REGISTER."/></action></nop>
=20
<recv response=3D"200" crlf=3D"true" rrs=3D"true">  <action><log
message=3D"received OK."/></action> </recv> </scenario>

- another sipp listening in port 5066, with its xml scenario for the
Invite:

 <?xml version=3D"1.0" encoding=3D"ISO-8859-1" ?> <scenario
name=3D"register_answer"> <recv request=3D"INVITE" crlf=3D"true" =
rrs=3D"true">
<action><log message=3D"received INVITE."/></action> </recv>
=20
<send>
 <![CDATA[
  SIP/2.0 180 Ringing
  [last_Via:]
  [last_From:]
  [last_To:];tag=3D[call_number]
  [last_Call-ID:]
  [last_CSeq:]
  [last_Record-Route:]
  Contact: <sip:9782060836@[local_ip]:[local_port]>
  Content-Length: [len]
 ]]>
</send>
<nop><action><log message=3D"sent RINGING."/></action></nop> <send>
<![CDATA[
  SIP/2.0 200 OK
  [last_Via:]
  [last_From:]
  [last_To:];tag=3D[call_number]
  [last_Call-ID:]
  [last_CSeq:]
  [last_Record-Route:]
  Contact: <sip:9782060836@[local_ip]:[local_port]>
  Content-Type: application/sdp
  Content-Length: [len]
=20
  v=3D0
  o=3D9782060836 [pid] [pid][cseq] IN IP4 [local_ip]
  s=3DQA Test
  c=3DIN IP4 [local_ip]
  t=3D0 0
  m=3Daudio [media_port] RTP/AVP 8
  a=3Drtpmap:8 pcmu/8000
  a=3Drtpmap:101 telephone-event/8000
  a=3Dfmtp:101 0-11,16
 ]]>
</send>
<nop><action><log message=3D"sent OK."/></action></nop> <recv
request=3D"ACK" crlf=3D"true" rrs=3D"true">  <action><log =
message=3D"received
ACK."/></action> </recv> <nop><action><log message=3D"CALL
INPROGRESS."/></action></nop> <recv request=3D"BYE" crlf=3D"true"
rrs=3D"true">  <action><log message=3D"received BYE."/></action> </recv>
<send>  <![CDATA[
  SIP/2.0 200 OK
  [last_Via:]
  [last_From:]
  [last_To:];tag=3D[call_number]
  [last_Call-ID:]
  [last_CSeq:]
  [last_Record-Route:]
  Contact: <sip:9782060836@[local_ip]:[local_port]>
  Content-Length: [len]
 ]]>
</send>
<nop><action><log message=3D"sent OK."/></action></nop> =
<nop><action><log
message=3D"CALL OVER."/></action></nop> </scenario>

I hope that this helps you.

Artur

Edward Russell wrote:
> I want to start up sipp with a scenario file which first sends a=20
> register, receives the 200 acknowledgement, and then waits to receive=20
> an INVITE.
> =20
> This does not appear to work.   While tcpdump shows the INVITE was=20
> received on the system, sipp does not display any evidence that it=20
> received it.  It just sits waiting for the current call to end
> ("current call" being the register I guess).   Message counts=20
> (discarded or otherwise) do not go up after the 200.  I am starting=20
> sipp with -m 1 because I just want to register and wait for 1 call.
> Later I'll put in a branch loop to register then wait for multiple
calls.
> =20
> However if I split the scenario into two files, and first run sipp=20
> with the send register/recv 200 portion and the -m 1 flag, and then=20
> run sipp with the receive INVITE/send RINGING portion with NO -m flag,

> all works fine.
> =20
> Looking at the code, I think it has something to do with server mode=20
> and client mode.  Because I am SENDING a register first, sipp puts=20
> itself into client mode and since I have the -m 1 flag it thinks the=20
> REGISTER was the one call and won't go on to process the INVITE?  If I

> leave off -m or put -m 2, multiple registrations are sent which is not

> what I want either.
> =20
> Every example I see has Register followed by SENDING an Invite, but I=20
> want to RECEIVE an INVITE after sending the register.
> =20
> Any thoughts?  If I solve this, will I really be able to branch and=20
> loop on the recv INVITE portion?  Would I use the -m flag then?
> =20
> Scenario I am using follows:
> =20
> <?xml version=3D"1.0" encoding=3D"ISO-8859-1" ?> <scenario=20
> name=3D"register_answer"> <send retrans=3D"1000">  <![CDATA[
>   REGISTER sip:hal9004.whatever.com SIP/2.0
>   Via: SIP/2.0/[transport]
[local_ip]:[local_port];rport;branch=3D[branch]
>   From:
<sip:[field0]@hal9004.whatever.com:[local_port]>;tag=3D[call_number]
>   To: <sip:[field0]@hal9004.whatever.com:[local_port]>
>   Contact: <sip:[field0]@[local_ip]:[local_port]>
>   Call-ID: [call_id]
>   CSeq: [cseq] REGISTER
>   Expires: 1800
>   Max-Forwards: 70
>   Content-Length: 0
>  ]]>
> </send>
> =20
> <nop><action><log message=3D"sent REGISTER."/></action></nop>
> =20
> <recv response=3D"200" crlf=3D"true" rrs=3D"true">  <action><log=20
> message=3D"received OK."/></action> </recv>
> =20
> <recv request=3D"INVITE" crlf=3D"true" rrs=3D"true">  <action><log=20
> message=3D"received INVITE."/></action> </recv>
> =20
> <send>
>  <![CDATA[
>   SIP/2.0 180 Ringing
>   [last_Via:]
>   [last_From:]
>   [last_To:];tag=3D[call_number]
>   [last_Call-ID:]
>   [last_CSeq:]
>   [last_Record-Route:]
>   Contact: <sip:9782060836@[local_ip]:[local_port]>
>   Content-Length: [len]
>  ]]>
> </send>
> <nop><action><log message=3D"sent RINGING."/></action></nop> <send> =20
> <![CDATA[
>   SIP/2.0 200 OK
>   [last_Via:]
>   [last_From:]
>   [last_To:];tag=3D[call_number]
>   [last_Call-ID:]
>   [last_CSeq:]
>   [last_Record-Route:]
>   Contact: <sip:9782060836@[local_ip]:[local_port]>
>   Content-Type: application/sdp
>   Content-Length: [len]
> =20
>   v=3D0
>   o=3D9782060836 [pid] [pid][cseq] IN IP4 [local_ip]
>   s=3DQA Test
>   c=3DIN IP4 [local_ip]
>   t=3D0 0
>   m=3Daudio [media_port] RTP/AVP 8
>   a=3Drtpmap:8 pcmu/8000
>   a=3Drtpmap:101 telephone-event/8000
>   a=3Dfmtp:101 0-11,16
>  ]]>
> </send>
> <nop><action><log message=3D"sent OK."/></action></nop> <recv=20
> request=3D"ACK" crlf=3D"true" rrs=3D"true">  <action><log =
message=3D"received=20
> ACK."/></action> </recv> <nop><action><log message=3D"CALL=20
> INPROGRESS."/></action></nop> <recv request=3D"BYE" crlf=3D"true"=20
> rrs=3D"true">  <action><log message=3D"received BYE."/></action> =
</recv>=20
> <send>  <![CDATA[
>   SIP/2.0 200 OK
>   [last_Via:]
>   [last_From:]
>   [last_To:];tag=3D[call_number]
>   [last_Call-ID:]
>   [last_CSeq:]
>   [last_Record-Route:]
>   Contact: <sip:9782060836@[local_ip]:[local_port]>
>   Content-Length: [len]
>  ]]>
> </send>
> <nop><action><log message=3D"sent OK."/></action></nop>=20
> <nop><action><log message=3D"CALL OVER."/></action></nop>
> =20
> </scenario>
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