Hi Michael,
Can you please let me know how to split two UAC and UAS?
Do we need to run two separate SIPp instance one for Register and one for
Invite?
In that case how they are going to map because Registration is done for
particular number so how Invite instance going to understand that it should
receive Invite for particular number?
I tried two split UAC and UAS..
I run two separate sipp instance.
"Instance A" for Register and "Instance B" for Invite.
In that case also I am receiving same error on "Instance A" ---Discarding
message which can't be mapped to a known..
Any suggestion any idea any help ?
Thanks for support... Please find my XML for "Instance A" and "Instance B"
"Instance A"---- XML
<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 >
<![CDATA[
REGISTER sip:10.230.53.225 SIP/2.0
Via: SIP/2.0/UDP 10.230.53.227:5060;branch=[branch];rport
Max-Forwards: 70
Contact: <sip:4...@10.230.53.227:5060>
To: "420"<sip:4...@10.230.53.225>
From: "420"<sip:4...@10.230.53.225>;tag=[call_number]
Call-ID: [call_id]
CSeq: [cseq] REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO
Content-Length: [len]
]]>
</send>
<recv response="200" crlf="true">
</recv>
<send >
<![CDATA[
SUBSCRIBE sip:4...@10.230.53.225 SIP/2.0
Via: SIP/2.0/UDP 10.230.53.227:5060;branch=[branch];rport
Max-Forwards: 70
Contact: <sip:4...@10.230.53.227:5060>
To: "420"<sip:4...@10.230.53.225>
From: "420"<sip:4...@10.230.53.225>;tag=[call_number]
Call-ID: [call_id]
CSeq: [cseq] SUBSCRIBE
Expires: 300
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO
User-Agent: X-Lite release 4204o stamp 56125
Event: message-summary
Content-Length: [len]
]]>
</send>
<recv response="501" crlf="true">
</recv>
"Instance B" xml sample
<scenario name="Basic Sipstone UAS">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<recv request="INVITE">
</recv>
<send>
<![CDATA[
SIP/2.0 180 Ringing
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
-----Original Message-----
From: Michael Hirschbichler [mailto:s...@hirschbichler.biz]
Sent: 12 April 2010 07:35
To: sipp-users@lists.sourceforge.net
Subject: Re: [Sipp-users] crazy problem on simple call scenario
This scenario as described below won't work.
If I understood the description correctly, the signalling-flow is
UA Proxy
---REGISTER-->
<---401-------
---REGISTER-->
<---200-------
<--INVITE-----
....
In sipp, the mapping of a message (request/reply) is done by parsing for
the SIP Call-ID - if a message is incoming with another call-id than the
call-id in the originating request, the message is dropped as an
unexpected message.
In general, one sipp instance is not able to act as a UAC (for the
registration process) and as an UAS (for the incomming invite request)
at the same time. You have to split up the functionality to two
sequenced sipp-instances:
UA_C_ Proxy
---REGISTER-->
<---401-------
---REGISTER-->
<---200-------
and after that
UA_S_ Proxy
<--INVITE-----
---180-------->
---200-------->
....
hth and br
Michael
On 2010-04-09 17:12, Ruhi Aslan wrote:
> ------------------------------------------------------------------------
> *De :* Ruhi Aslan
> *Envoyé :* vendredi, 9. avril 2010 16:56
> *À :* 'sipp-users-requ...@lists.sourceforge.net'
> *Objet :* help
>
> Hi all,
>
> Sipp is a great tool and I currently pull my hair out...
>
> I have some trouble with a very simple scenario. I even can't make a
> call to sipp registered phone.
> I first registered my phone :
>
> sipp -sf callee_hangup_process_test.xml -inf
> csv/register_client.csv asterisk.ch -trace_err -r1 -m 1
>
> ## register my sipp phone to get calls
>
> <send>
> <![CDATA[
>
> REGISTER sip:sipproxy SIP/2.0
> Via: SIP/2.0/UDP mycomputerIP:5060;branch=z9hG4bK-ID
> From: <sip:4...@mycomputerip>;tag=1
> To: <sip:4...@mycomputerip>
> Call-ID: 1...@mycomputerip <mailto:1...@mycomputerip>
> CSeq: 1 REGISTER
> Contact: *
> Max-Forwards: 5
> Expires: 0
> User-Agent: SIPp/Linux
> Content-Length: 0
>
> ]]>
> </send>
> <recv response="404" optional="true" next="1">
> </recv>
>
> <recv response="401" auth="true">
> </recv>
>
> ******* Register Process *******
>
> <send retrans="500">
> <![CDATA[
>
> REGISTER sip:sipproxy SIP/2.0
> Via: SIP/2.0/UDP mycomputerIP:5060;branch=z9hG4bK-ID
> From: <sip:4...@mycomputerip>;tag=1
> To: <sip:4...@mycomputerip>
> Call-ID: 1...@mycomputerip <mailto:1...@mycomputerip>
> CSeq: 1 REGISTER
> Contact: *
> [AUTHENTICATION LINE]
> Max-Forwards: 5
> Expires: 0
> User-Agent: SIPp/Linux
> Content-Length: 0
>
> ]]>
>
> </send>
> <recv response="200">
> </recv>
>
> ### phone registered, sip show peer 44 tell me it's OK and reachable on
> mycomputerIP
>
>
> Then I ask to it to wait until an INVITE comes :
>
> <recv request="INVITE" crlf="true">
> </recv>
>
>
> In another window, I make a call with another phone number 43 ( correct
> scenarios and successfully tested )
>
> sipp -sf callee_hangup.xml -inf csv/caller.cvs asterisk.ch -trace_err
> -r 1 -m 1
>
> BUT, callee_hangup_process_test.xml doesn't get the INVITE from
> callee_hangup.xml scenario.
> The crazy thing is that wireshark says that it sends the expected INVITE
> to callee_hangup_process_test.xml ( on the right computer, on the right
> port ). But on my previous INVITE recv request, the count persist on 0 !
>
>
> Here the INVITE sended to mycomputerIP ( supposed to make the INVITE
> recv reauest count up to 1 )
>
> INVITE sip:4...@mycomputerip:5060 SIP/2.0
> Record-Route: <sip:sipproxy;lr=on;ftag=ftag;vsf=some...;did=...>
> Via: SIP/2.0/UDP sipproxy;branch=z9hG4bK-ID2
> Via: SIP/2.0/UDP
> asterisk.ch:5060;received=asterisk.ch;branch=z9hG4b-ID;rport=5060
> From: "43" <sip:4...@voip.vtx.ch>;tag=as1cf8af76
> To: <sip:4...@mycomputerip:5060>
> Contact: <sip:4...@_asterisk.ch_>
> Call-ID: call...@asterisk.ch <mailto:call...@asterisk.ch>
> CSeq: 102 INVITE
> User-Agent: voipua
> Max-Forwards: 69
> Date: Fri, 09 Apr 2010 13:54:19 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
> Content-Type: application/sdp
> Content-Length: 242
> P-hint: outbound
>
> v=0
> o=root 26199 26199 IN IP4 _asterisk.ch_
> s=session
> c=IN IP4 _asterisk.ch_
> t=0 0
> m=audio 18150 RTP/AVP 8 0 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
>
>
> more info :
>
> I already use -aa option for OPTIONS NOTIFY request, and on the second
> OPTIONS, sipp crash on seg fault :-\
>
>
>
> So where is my mistake ?
>
> Ruhi ASLAN
> Stagiaire ST40 - NOC/Operation
>
>
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