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 > > ------------------------------------------------------------------------------ Download Intel® Parallel Studio Eval Try the new software tools for yourself. Speed compiling, find bugs proactively, and fine-tune applications for parallel performance. See why Intel Parallel Studio got high marks during beta. http://p.sf.net/sfu/intel-sw-dev _______________________________________________ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
