Simply adding this -m 1 flag kept SIPp from sending over and over SIP Registration request.
Thank you very much, Pavel, for spotting this ! Le jeu. 25 oct. 2018 à 20:26, Šindelka Pavel <sinde...@ttc.cz> a écrit : > This time the -m 1 is missing on your command line, so sipp sends new > "calls" (actually, runs the scenario with new Call-ID values) without any > limit. > > P. > > Dne 25.10.2018 v 18:25 Olivier napsal(a): > > Thanks Pavel for replying. > > Thanks to your advice, I restarted all over, using another Asterisk client > instance as a guide to a canonical REGISTER dialog. > When I got a successful REGISTER dialog between 2 Asterisk instances, I > stopped Asterisk on client host and used SIPp instead. > I progressed step by step to the point I saw my Asterisk server replying > with a 200OK to the REGISTER ! > > I'm very happy with this result but the issue is that this REGISTER dialog > is repeated many times while I would like to play it only once. > How can I do that ? > > > SIPp is invoked with: > sipp 192.168.64.46:5062 -sf /home/foobar/my-uac-auth.xml -ap passsipp -s > sipp -i 192.168.64.45 -p 5062 > > > My /home/foobar/my-uac-auth.xml > file is now: > > <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 retrans="500"> > <![CDATA[ > > REGISTER sip:[remote_ip]:[remote_port] SIP/2.0 > Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] > From: <sip:[service]@[remote_ip]>;tag=[pid]SIPpTag00[call_number] > To: <sip:[service]@[remote_ip]> > Call-ID: [call_id] > CSeq: 1 REGISTER > Contact: <sip:s@[local_ip]:[local_port]> > Max-Forwards: 70 > Subject: Performance Test > Content-Type: application/sdp > Content-Length: [len] > ]]> > </send> > > <recv response="200" > optional="true" > next="auth_done"> > </recv> > > <recv response="401" > auth="true"> > </recv> > > <send retrans="500"> > <![CDATA[ > > REGISTER sip:[remote_ip]:[remote_port] SIP/2.0 > Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] > From: <sip:[service]@[remote_ip]>;tag=[pid]SIPpTag00[call_number] > To: <sip:[service]@[remote_ip]> > Call-ID: [call_id] > CSeq: 1 REGISTER > Contact: <sip:s@[local_ip]:[local_port]> > Max-Forwards: 70 > [authentication username=sipp password=passsipp] > Subject: Performance Test > Content-Type: application/sdp > Content-Length: [len] > ]]> > </send> > > <recv response="200"> > </recv> > > <label id="auth_done" /> > > <pause milliseconds="5000"/> > > </scenario> > > > Here is a trace of corresponding REGISTER dialog (captured on Asterisk and > received on SIPp host): > > <--- SIP read from UDP:192.168.64.45:5062 ---> > REGISTER sip:192.168.64.46:5062 SIP/2.0 > Via: SIP/2.0/UDP 192.168.64.45:5062;branch=z9hG4bK-19512-22-0 > From: <sip:sipp@192.168.64.46>;tag=19512SIPpTag0022 > To: <sip:sipp@192.168.64.46> > Call-ID: 22-19512@192.168.64.45 > CSeq: 1 REGISTER > Contact: <sip:s@192.168.64.45:5062> > Max-Forwards: 70 > Subject: Performance Test > Content-Type: application/sdp > Content-Length: 0 > > <-------------> > --- (11 headers 0 lines) --- > Sending to 192.168.64.45:5062 (no NAT) > Sending to 192.168.64.45:5062 (no NAT) > > <--- Transmitting (no NAT) to 192.168.64.45:5062 ---> > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.64.45:5062 > ;branch=z9hG4bK-19512-22-0;received=192.168.64.45 > From: <sip:sipp@192.168.64.46>;tag=19512SIPpTag0022 > To: <sip:sipp@192.168.64.46>;tag=as41ad7250 > Call-ID: 22-19512@192.168.64.45 > CSeq: 1 REGISTER > Server: Asterisk PBX 13.23.1~dfsg-1 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH, MESSAGE > Supported: replaces, timer > WWW-Authenticate: Digest algorithm=MD5, realm="canonical", nonce="3386ce74" > Content-Length: 0 > > > <------------> > Scheduling destruction of SIP dialog '22-19512@192.168.64.45' in 32000 ms > (Method: REGISTER) > > <--- SIP read from UDP:192.168.64.45:5062 ---> > REGISTER sip:192.168.64.46:5062 SIP/2.0 > Via: SIP/2.0/UDP 192.168.64.45:5062;branch=z9hG4bK-19512-22-3 > From: <sip:sipp@192.168.64.46>;tag=19512SIPpTag0022 > To: <sip:sipp@192.168.64.46> > Call-ID: 22-19512@192.168.64.45 > CSeq: 1 REGISTER > Contact: <sip:s@192.168.64.45:5062> > Max-Forwards: 70 > Authorization: Digest username="sipp",realm="canonical",uri="sip: > 192.168.64.46:5062 > ",nonce="3386ce74",response="b0388098db7f3fd69dbd4c8a030d8d28",algorithm=MD5 > Subject: Performance Test > Content-Type: application/sdp > Content-Length: 0 > > <-------------> > --- (12 headers 0 lines) --- > Sending to 192.168.64.45:5062 (no NAT) > > <--- Transmitting (no NAT) to 192.168.64.45:5062 ---> > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.64.45:5062 > ;branch=z9hG4bK-19512-22-3;received=192.168.64.45 > From: <sip:sipp@192.168.64.46>;tag=19512SIPpTag0022 > To: <sip:sipp@192.168.64.46>;tag=as41ad7250 > Call-ID: 22-19512@192.168.64.45 > CSeq: 1 REGISTER > Server: Asterisk PBX 13.23.1~dfsg-1 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH, MESSAGE > Supported: replaces, timer > Expires: 120 > Contact: <sip:s@192.168.64.45:5062>;expires=120 > Date: Thu, 25 Oct 2018 16:08:24 GMT > Content-Length: 0 > > > > Any clue ? > > Le jeu. 25 oct. 2018 à 13:26, Šindelka Pavel <sinde...@ttc.cz> a écrit : > >> Hi Olivier, >> >> looking at your command line with -m=1 and at the scenario, I suppose >> that the repeated REGISTER requests without the authentication header you >> can see are retransmissions of the initial one, implying that the sipp >> machine/process does not receive the responses from the Asterisk. >> >> This can have a number of reasons: >> >> - wrong population of the headers in the messages generated from the >> scenario (should not be the case as you've specified -i on the >> command line) >> - routing issue (unlikely unless you've intentionally split >> 192.168.64.0/24 into several subnets or misconfigured the network >> mask on either machine unintentionally) >> - Asterisk configuration issue (not permitting incoming registrations >> from this address/subnet) >> - firewall issue on either machine >> >> So SIPp logs, Asterisk logs, and tcpdump/Wireshark are your best friends. >> See whether the REGISTER arrives to the Asterisk, what is its contents, and >> whether the Asterisk responds at all and where it sends the responses. >> >> Pavel >> >> Dne 25.10.2018 v 11:17 Olivier napsal(a): >> >> Hello, >> >> I'm quite new to SIPp. >> I've just discovered [1]. >> I'm testing this uac-auth.xml file with the bellow command against an >> Asterisk instance: >> >> sipp -sf uac-auth.xml 192.168.64.250 -au 7005 -ap 7005 -s 7005 -i >> 192.168.64.45 -m 1 >> >> I see that Asterisk challenges incoming REGISTER with a WWW-Authenticate >> but SIPp does not reply with any new REGISTER with an Authorization header. >> Instead, it keeps sending first REGISTER. >> >> 1. Am I correct to expect, with referenced uac-auth.xml, SIPp to send a >> REGISTER with an Authorization header ? >> >> 2. If negative, what should be changed to in uac-auth.xml to implement >> this ? If positive, is it correct to expect [authentication] lines in a >> REGISTER to be replaced with an Authorization built with data coming from >> matching 401 reply (nonce, realm, ...) ? >> >> Best regards >> >> [1] https://github.com/rkday/sipp-samples/blob/master/uac-auth.xml >> >> >> >> >> _______________________________________________ >> Sipp-users mailing >> listSipp-users@lists.sourceforge.nethttps://lists.sourceforge.net/lists/listinfo/sipp-users >> >> >> _______________________________________________ >> Sipp-users mailing list >> Sipp-users@lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/sipp-users >> > > > > > _______________________________________________ > Sipp-users mailing > listSipp-users@lists.sourceforge.nethttps://lists.sourceforge.net/lists/listinfo/sipp-users > > > _______________________________________________ > Sipp-users mailing list > Sipp-users@lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/sipp-users >
_______________________________________________ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
