On 28 Aug 2014, at 14:14, Yuriy Gorlichenko <ovoshl...@gmail.com> wrote:
> Hello. I try to provide call scheme: > > internal client -> asterisk -> Kamailio -> provider -> external endpoint call > > when I make call I see this: > > asterisk kamailio provider > invite --> invite --> > <-- 407 > ACK --> > invite w/Auth --> > <-- 100 <-- 100 > <-- 180 <-- 180 > <-- 183 <-- 183 > <-- 200 <-- 200 > ACK --> ACK --> > > My problem with last ACK, that I send to provider. Provider ignores it, and > sends me some OK packets. As resultI can notend session ( answer to BYE 481 - > transaction does not exists). I think it is wrong ACK but can not undrtand > where I do mistake. Well, by letting the proxy handle authentication the INVITE tranction i closed without Asterisk knowing about it. So the ACK sent from the proxy and from Asterisk is for the same transaction, which messes things up. Asterisk does not know anything about the second invite. Letting the proxy handle authentiction breaks the SIP protocol in bad ways and is generally not a good solution. You may want to send another response to asterisk when you get the 407 so Asterisk retries and use the retry as a trigger for the second INVITE and add auth to that. /O > > Please help me to find it: > > My invite (with Auth creditans): > > IP 10.0.1.18.5068 > my.provider.ip.5060: UDP, length 1606 > E...]. .@..R > ...6........N0TINVITE sip:12345678...@my.provider.ip:5060 SIP/2.0 > Record-Route: <sip:my.external.ip:5068;nat=yes;ftag=as7d06fc50;lr=on> > Via: SIP/2.0/UDP > my.external.ip:5068;branch=z9hG4bK48ba.74ed5eb56b172cd802c50dcc201cce56.1 > Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK258b5220;rport=50600 > Max-Forwards: 70 > From: "John" <sip:provider_usern...@my.provider.ip>;tag=as7d06fc50 > To: <sip:12345678...@my.provider.ip:5068> > Contact:<provider_usern...@my.external.ip:5068> > Call-ID: 2122fc6a3cbe2e64253289cf23c3dd2a@10.0.1.6:50600 > CSeq: 102 INVITE > User-Agent: Asterisk PBX 12.5.0 > Date: Wed, 27 Aug 2014 22:02:58 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH, MESSAGE > Supported: replaces, timer > Content-Type: application/sdp > Content-Length: 544 > Proxy-Authorization: Digest username="provider_username", > realm="my.provider.ip", nonce="U/5Wv1P+VZNjFBLf6fwPizgd6iLto5St", > uri="sip:12345678...@my.provider.ip:5060", qop=auth, nc=00000001, > cnonce="2888860875", response="9f23110471fe9ff751cd55466e70ded2", > algorithm=MD5 > > v=0 > o=root 1370647246 1370647246 IN IP4 12.34.56.78 > s=Asterisk PBX 12.5.0 > c=IN IP4 12.34.56.78 > t=0 0 > a=ice-lite > m=audio 30296 RTP/AVP 0 3 8 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=maxptime:150 > a=sendrecv > a=rtcp:30297 > a=ice-ufrag:p5k92ynl > a=ice-pwd:FIOYKt96NlBfEqKsQipUuadUev1g > a=candidate:vV3V06Tv > > > > Provider trying > > > IP my.provider.ip.5060 > 10.0.1.18.5068: UDP, length 500 > E.........PX6... > ..........ySIP/2.0 100 trying -- your call is important to us > Via: SIP/2.0/UDP > my.external.ip:5068;branch=z9hG4bK48ba.74ed5eb56b172cd802c50dcc201cce56.1;rport=5068;received=12.34.56.78 > Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK258b5220;rport=50600 > From: "John" <sip:provider_usern...@my.provider.ip>;tag=as7d06fc50 > To: <sip:12345678...@my.provider.ip:5068> > Call-ID: 2122fc6a3cbe2e64253289cf23c3dd2a@10.0.1.6:50600 > CSeq: 102 INVITE > Server: kamailio (4.1.2 (x86_64/linux)) > Content-Length: 0 > > > > > provider ringing > > > > > IP my.provider.ip.5060 > 10.0.1.18.5068: UDP, length 1098 > E..f......M.6... > ........RV.SIP/2.0 180 Ringing > Via: SIP/2.0/UDP > my.external.ip:5068;rport=5068;received=12.34.56.78;branch=z9hG4bK48ba.74ed5eb56b172cd802c50dcc201cce56.1 > Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK258b5220;rport=50600 > Record-Route: <sip:my.provider.ip;lr=on;ftag=as7d06fc50;did=5bc.33f1> > Record-Route: <sip:my.external.ip:5068;nat=yes;ftag=as7d06fc50;lr=on> > From: "John" <sip:provider_usern...@my.provider.ip>;tag=as7d06fc50 > To: <sip:12345678...@my.provider.ip:5068>;tag=v9g4HD4vrNFUH > Call-ID: 2122fc6a3cbe2e64253289cf23c3dd2a@10.0.1.6:50600 > CSeq: 102 INVITE > Contact: <sip:12345678900@67.192.253.160:5060;transport=udp> > User-Agent: Plivo > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, > NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Allow-Events: talk, hold, conference, presence, dialog, line-seize, > call-info, sla, include-session-description, presence.winfo, message-summary, > refer > Content-Length: 0 > Remote-Party-ID: "12345678900" > <sip:12345678...@my.provider.ip>;party=calling;privacy=off;screen=no > > > > provider seesion in progress > > > > IP my.provider.ip.5060 > 10.0.1.18.5068: UDP, length 1887 > E..... ...,.6... > ........g.DSIP/2.0 183 Session Progress > Via: SIP/2.0/UDP > my.external.ip:5068;rport=5068;received=12.34.56.78;branch=z9hG4bK48ba.74ed5eb56b172cd802c50dcc201cce56.1 > Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK258b5220;rport=50600 > Record-Route: <sip:my.provider.ip;lr=on;ftag=as7d06fc50;did=5bc.33f1> > Record-Route: <sip:my.external.ip:5068;nat=yes;ftag=as7d06fc50;lr=on> > From: "John" <sip:provider_usern...@my.provider.ip>;tag=as7d06fc50 > To: <sip:12345678...@my.provider.ip:5068>;tag=v9g4HD4vrNFUH > Call-ID: 2122fc6a3cbe2e64253289cf23c3dd2a@10.0.1.6:50600 > CSeq: 102 INVITE > Contact: <sip:12345678900@67.192.253.160:5060;transport=udp> > User-Agent: Plivo > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, > NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Allow-Events: talk, hold, conference, presence, dialog, line-seize, > call-info, sla, include-session-description, presence.winfo, message-summary, > refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 742 > Remote-Party-ID: "12345678900" > <sip:12345678...@my.provider.ip>;party=calling;privacy=off;screen=no > > v=0 > o=FreeSWITCH 1409149800 1409149801 IN IP4 67.192.253.160 > s=FreeSWITCH > c=IN IP4 67.192.253.160 > t=0 0 > a=msid-semantic: WMS uIWGGSqM8mUp5NEgQ9CU0svyzqjzisqD > m=audio 27180 RTP/AVP 0 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=ssrc:326362635 cnam > > > > > provider OK > > > > IP my.provider.ip.5060 > 10.0.1.18.5068: UDP, length 2026 > E..... ...,.6... > ...........SIP/2.0 200 OK > Via: SIP/2.0/UDP > my.external.ip:5068;rport=5068;received=12.34.56.78;branch=z9hG4bK48ba.74ed5eb56b172cd802c50dcc201cce56.1 > Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK258b5220;rport=50600 > Record-Route: <sip:my.provider.ip;lr=on;ftag=as7d06fc50;did=5bc.33f1> > Record-Route: <sip:my.external.ip:5068;nat=yes;ftag=as7d06fc50;lr=on> > Fл2rom: "John" <sip:provider_usern...@my.provider.ip>;tag=as7d06fc50 > To: <sip:12345678...@my.provider.ip:5068>;tag=v9g4HD4vrNFUH > Call-ID: 2122fc6a3cbe2e64253289cf23c3dd2a@10.0.1.6:50600 > CSeq: 102 INVITE > Contact: <sip:12345678900@67.192.253.160:5060;transport=udp> > User-Agent: Plivo > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, > NOTIFY, PUBLISH, SUBSCRIBE > SupлЛ > o=FreeSWITCH 1409149800 1409149801 IN IP4 67.192.253.160 > s=FreeSWITCH > c=л2IN IP4 67.192.253.160 > t=0 0 > a=msid-semantic: WMS uIWGGSqM8mUp5NEgQ9CU0svyzqjzisqD > m=audio 27180 RTP/AVP 0 > > > > > my ACK > > > > > IP 10.0.1.18.5068 > my.provider.ip.5060: UDP, length 614 > E...]...@... > ...6........n.hACK sip:12345678...@my.provider.ip:5060;transport=udp SIP/2.0 > Via: SIP/2.0/UDP > my.external.ip:5068;branch=z9hG4bK48ba.4250e4d315c4aa6697b6d7f70e861b62.0 > Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK4d28fc11;rport=50600 > Route: <sip:my.provider.ip;lr=on;ftag=as7d06fc50;did=5bc.33f1> > Max-Forwards: 70 > From: "John" <sip:provider_usern...@my.provider.ip>;tag=as7d06fc50 > To: <sip:12345678...@my.provider.ip:5068>;tag=v9g4HD4vrNFUH > Contact:<provider_usern...@my.external.ip:5068> > Call-ID: 2122fc6a3cbe2e64253289cf23c3dd2a@10.0.1.6:50600 > CSeq: 102 ACK > User-Agent: Asterisk PBX 12.5.0 > Content-Length: 0 > > > > So after this ACK provider still sends me 200 OK and my server still sends > ACK.... > > tags and call-id always one. > > > Thanks > _______________________________________________ > 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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