Hi Guys, me again, I increased the pause in uas after RINGING to 1000 milliseconds. With this value, works fine if I send 15 cps BUT if I send 25 I have to increase the pause to 2000 milliseconds.
<send> <![CDATA[ SIP/2.0 180 Ringing [last_Record-route] [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> <pause milliseconds="2000"/> If you see the first trace is kamailio who answers the BYE with a 404. Is it like the call wasn't established? Any help we'll be appreciated. Thanks in advance. Diego! El El mié, 14 de dic. de 2016 a las 18:22, Diego Nadares <dnada...@gmail.com> escribió: > Hi Mack, > > The only thing I added to the basic scenarios is rrs="true" > > In UAS > <recv rrs="true" request="INVITE" crlf="true"> > > In UAC > <recv response="200" rrs="true" crlf="true"> > > I use the same scenario with the same fields for every call. In a test of > 6000 calls I see ~5 dead calls. > > Diego > > > 2016-12-14 17:28 GMT-03:00 Mack Hendricks <m...@dopensource.com>: > > Hey Diego, > > This smells like a sipP scenario file issue. Did you customize the the > scenario file being used by sipP B? > > -Mack > > On Dec 14, 2016, at 3:23 PM, Diego Nadares <dnada...@gmail.com> wrote: > > Hi guys, > > We are testing kamailio with sipp. We are running it with 20cps and some > calls do the following. When Kamailio is processing the 'ringing' a '200ok' > arrives in the middle. First, kamailio forwards the 200 ok and then the > ringing. 'ACK' arrives and, I suppose, the call is established. The thing > is than when the 'BYE' arrives kamailio responds with a 404. > > This is the summary of the call > > Id Time Source Destination Protocol Len Info > > 6483 2016-12-14 11:26:06.264697 SIPP-A KAMAILIO SIP/SDP 692 Request: > INVITE sip:11111111@172.16.213.38:5060 | > > 6484 2016-12-14 11:26:06.264937 KAMAILIO SIPP-A SIP 367 Status: 100 > trying -- your call is important to us | > 6485 2016-12-14 11:26:06.266327 KAMAILIO SIPP-B SIP/SDP 1179 Request: > INVITE sip:22222222@172.16.213.31:5060 | > *6486 2016-12-14 11:26:06.267217 SIPP-B KAMAILIO SIP 566 Status: 180 > Ringing | * > 6487 2016-12-14 11:26:06.267268 SIPP-B KAMAILIO SIP/SDP 733 Status: 200 > OK | > 6488 2016-12-14 11:26:06.267758 KAMAILIO SIPP-A SIP/SDP 788 Status: 200 > OK | > *6489 2016-12-14 11:26:06.267833 *KAMAILIO SIPP-A* SIP 442 Status: 180 > Ringing | * > 6490 2016-12-14 11:26:06.268868 SIPP-A KAMAILIO SIP 493 Request: ACK > sip:127.0.0.8;line=sr-N6IAzBFwMJZfWJZLM.M7MlF-W.y6Mx14NEt7Nw05NhPQKjaP | > 6491 2016-12-14 11:26:06.269162 KAMAILIO SIPP-B SIP 609 Request: ACK > sip:172.16.213.31:5060;transport=UDP | > 6492 2016-12-14 11:26:06.269614 SIPP-A KAMAILIO SIP 404 Request: BYE > sip:11111111@172.16.213.38:5060 | > 6493 2016-12-14 11:26:06.269782 KAMAILIO SIPP-A SIP 348 Status: *404 Not > here* | > > We are using modules rtjson, evapi, uac, topoh, rtpproxy for all calls. My > debug level is -1. With higher levels this behavior increase. > > Kamailio is running in a virtual machine with centos7 with 8 cores and 8gb > of ram. > > Do you need any further information? I can send you a pcap or ngrep file. > > Best regards, > > Diego. > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > _______________________________________________ > > > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > > > sr-users@lists.sip-router.org > > > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > > > >
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