Hi,
I tried all possibilities and now I'm stuck again. I tried to split register
and INVITE receive scenario, I tried to put all in the same file by using ///
in call ID and it has the same effect, but nothing works !
My super register - wait for invite scenario result look like this :
REGISTER ----------> 1 0
401 <---------- 1 0 0
REGISTER ----------> 1 0 0
200 <---------- 1 0 0
REGISTER ----------> 1 0
401 <---------- 1 0 0
REGISTER ----------> 1 0 0
200 <---------- 1 0 0
INVITE <---------- 0 0 0
180 ----------> 0 0
100 ----------> 0 0
200 ----------> 0 0
ACK <---------- E-RTD1 0 0 0
BYE <---------- 0 0 0
200 ----------> 0 0
Pause [ 4000ms] 0
At this point I make a call to my callee, but the Invite is not recognize by
sipp.
Here the scenario source , maybe you will see the mistake that I missed !!
<send>
<![CDATA[
REGISTER sip:[remote_ip] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: <sip:[fiel...@[local_ip]>;tag=[call_number]
To: <sip:[fiel...@[local_ip]>
Call-ID: A///[call_id]
CSeq: 1 REGISTER
Contact: *
Max-Forwards: 5
Expires: 0
User-Agent: SIPp/Linux
Content-Length: 0
]]>
</send>
<recv response="401" auth="true">
</recv>
<send retrans="500">
<![CDATA[
REGISTER sip:[remote_ip] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: <sip:[fiel...@[local_ip]>;tag=[call_number]
To: <sip:[fiel...@[local_ip]>
Call-ID: A///[call_id]
CSeq: 2 REGISTER
Contact: *
[field1]
Max-Forwards: 5
Expires: 0
User-Agent: SIPp/Linux
Content-Length: 0
]]>
</send>
<recv response="200">
</recv>
<send>
<![CDATA[
REGISTER sip:[remote_ip] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
[routes]
From: <sip:[fiel...@[local_ip]>;tag=[call_number]
To: <sip:[fiel...@[local_ip]>
Call-ID: B///[call_id]
CSeq: 3 REGISTER
Contact: sip:[fiel...@[local_ip]:[local_port]
Max-Forwards: 5
Expires: 40
User-Agent: SIPp/Linux
Content-Length: 0
]]>
</send>
<recv response="401" auth="true">
</recv>
<send retrans="500">
<![CDATA[
REGISTER sip:[remote_ip] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
[routes]
From: <sip:[fiel...@[local_ip]>;tag=[call_number]
To: <sip:[fiel...@[local_ip]>
Call-ID: B///[call_id]
CSeq: 4 REGISTER
Contact: sip:[fiel...@[local_ip]:[local_port]
[field1]
Max-Forwards: 5
Expires: 40
User-Agent: SIPp/Linux
Content-Length: 0
]]>
</send>
<recv response="200">
</recv>
<recv request="INVITE" crlf="true">
</recv>
<send >
<![CDATA[
SIP/2.0 180 Ringing
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
[routes]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<send>
<![CDATA[
SIP/2.0 100 Trying
[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>
<send>
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Type: application/sdp
Content-Length: 136
v=0
o=user1 53655765 2353687637 IN IP4 127.0.0.1
s=-
t=0 0
c=IN IP4 [media_ip]
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
</send>
<recv request="ACK"
rtd="true"
crlf="true">
</recv>
<recv request="BYE">
</recv>
<send>
<![CDATA[
SIP/2.0 200 OK
[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>
Because of our infrastructure, I have to use -aa option which is buggy on 3.1 (
seg fault ), I use the 2.0 stable and I apply this patch
> >
> > In call.cpp change line 3388 (you can find it searching for string
> > "NOTIFY") to:
> >
> > } else if (((strcmp(P_recv, "INFO") == 0) || (strcmp(P_recv, "NOTIFY") ==
> > 0) || (strcmp(P_recv, "UPDATE") == 0) || (strcmp(P_recv, "OPTIONS") == 0))
see on
http://www.mail-archive.com/sipp-users@lists.sourceforge.net/msg03427.html
Now, by using -aa , sipp answer properly to all OPTIONS NOTIFY and other kids
with an 200 ok, but it answer to my all INVITE message too !! So this is not
better for me...
ritesh.gupta I tried to use in your way, but I haven't the same results !!
Maybe it is because I have to enable OPTIONS - 200ok with -aa, could you please
tell me what is your work around ?
Thanks.
Ruhi ASLAN
Stagiaire ST40 - NOC/Operation
VTX SERVICES SA
Une société du groupe VTX Telecom
================================================================
Tél. direct : 021 721 12 18
Av. de Lavaux 101 - 1009 Pully
http://www.vtx.ch <http://www.vtx.ch/> - ruhi.as...@vtx-telecom.ch
----------------------------------------------------------------
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================================================================
________________________________
De : ritesh.gu...@bt.com [mailto:ritesh.gu...@bt.com]
Envoyé : mercredi, 21. avril 2010 18:46
À : himanshu.ra...@gmail.com
Cc : sipp-users@lists.sourceforge.net
Objet : Re: [Sipp-users] crazy problem on simple call scenario
I got the solution .. ...
first run the UAC (registration) once registration are done then exit this
SIPp instance.
Then run new instance for UAS (Receiving Invite).
Now it will work.
From: Himanshu Rawat [mailto:himanshu.ra...@gmail.com]
Sent: 20 April 2010 11:45
To: Gupta,R,Ritesh,DKH C
Cc: s...@hirschbichler.biz; sipp-users@lists.sourceforge.net
Subject: Re: [Sipp-users] crazy problem on simple call scenario
Me too getting same error in the error log files even though I'm just running
sipp as a client and connecting to the actual voice mail server. :( :(.
Please anyone can tell why its happening.??
Cheers,
Rawat
On Tue, Apr 20, 2010 at 15:07, <ritesh.gu...@bt.com> wrote:
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 <mailto:sip%3a...@10.230.53.225> >
From: "420"<sip:4...@10.230.53.225 <mailto:sip%3a...@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 <mailto:sip%3a...@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 <mailto:sip%3a...@10.230.53.225> >
From: "420"<sip:4...@10.230.53.225 <mailto:sip%3a...@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 <mailto:sip%3...@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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