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<http://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]><sip:[service]@[remote_ip]>;tag=[pid]SIPpTag00[call_number] To: <sip:[service]@[remote_ip]><sip:[service]@[remote_ip]> Call-ID: [call_id] CSeq: 1 REGISTER Contact: <sip:s@[local_ip]:[local_port]><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]><sip:[service]@[remote_ip]>;tag=[pid]SIPpTag00[call_number] To: <sip:[service]@[remote_ip]><sip:[service]@[remote_ip]> Call-ID: [call_id] CSeq: 1 REGISTER Contact: <sip:s@[local_ip]:[local_port]><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<http://192.168.64.45:5062> ---> REGISTER sip:192.168.64.46:5062<http://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<mailto:sip%3Asipp@192.168.64.46>>;tag=19512SIPpTag0022 To: <sip:sipp@192.168.64.46<mailto:sip%3Asipp@192.168.64.46>> Call-ID: 22-19512@192.168.64.45<mailto:22-19512@192.168.64.45> CSeq: 1 REGISTER Contact: <sip:s@192.168.64.45:5062<http://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<http://192.168.64.45:5062> (no NAT) Sending to 192.168.64.45:5062<http://192.168.64.45:5062> (no NAT) <--- Transmitting (no NAT) to 192.168.64.45:5062<http://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<mailto:sip%3Asipp@192.168.64.46>>;tag=19512SIPpTag0022 To: <sip:sipp@192.168.64.46<mailto:sip%3Asipp@192.168.64.46>>;tag=as41ad7250 Call-ID: 22-19512@192.168.64.45<mailto: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<mailto:22-19512@192.168.64.45>' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:192.168.64.45:5062<http://192.168.64.45:5062> ---> REGISTER sip:192.168.64.46:5062<http://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<mailto:sip%3Asipp@192.168.64.46>>;tag=19512SIPpTag0022 To: <sip:sipp@192.168.64.46<mailto:sip%3Asipp@192.168.64.46>> Call-ID: 22-19512@192.168.64.45<mailto:22-19512@192.168.64.45> CSeq: 1 REGISTER Contact: <sip:s@192.168.64.45:5062<http://sip:s@192.168.64.45:5062>> Max-Forwards: 70 Authorization: Digest username="sipp",realm="canonical",uri="sip:192.168.64.46:5062<http://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<http://192.168.64.45:5062> (no NAT) <--- Transmitting (no NAT) to 192.168.64.45:5062<http://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<mailto:sip%3Asipp@192.168.64.46>>;tag=19512SIPpTag0022 To: <sip:sipp@192.168.64.46<mailto:sip%3Asipp@192.168.64.46>>;tag=as41ad7250 Call-ID: 22-19512@192.168.64.45<mailto: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<http://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<mailto: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<http://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 list Sipp-users@lists.sourceforge.net<mailto:Sipp-users@lists.sourceforge.net> https://lists.sourceforge.net/lists/listinfo/sipp-users _______________________________________________ Sipp-users mailing list Sipp-users@lists.sourceforge.net<mailto:Sipp-users@lists.sourceforge.net> https://lists.sourceforge.net/lists/listinfo/sipp-users _______________________________________________ Sipp-users mailing list Sipp-users@lists.sourceforge.net<mailto:Sipp-users@lists.sourceforge.net> https://lists.sourceforge.net/lists/listinfo/sipp-users
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