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.
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:12345678900@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_username@my.provider.ip;tag=as7d06fc50 To: sip:12345678900@my.provider.ip:5068 Contact:<provider_username@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:12345678900@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_username@my.provider.ip;tag=as7d06fc50 To: sip:12345678900@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_username@my.provider.ip;tag=as7d06fc50 To: sip:12345678900@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:12345678900@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_username@my.provider.ip;tag=as7d06fc50 To: sip:12345678900@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:12345678900@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_username@my.provider.ip;tag=as7d06fc50 To: sip:12345678900@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:12345678900@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_username@my.provider.ip;tag=as7d06fc50 To: sip:12345678900@my.provider.ip:5068;tag=v9g4HD4vrNFUH Contact:<provider_username@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
On 28 Aug 2014, at 14:14, Yuriy Gorlichenko ovoshlook@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:12345678900@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_username@my.provider.ip;tag=as7d06fc50 To: sip:12345678900@my.provider.ip:5068 Contact:<provider_username@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:12345678900@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_username@my.provider.ip;tag=as7d06fc50 To: sip:12345678900@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_username@my.provider.ip;tag=as7d06fc50 To: sip:12345678900@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:12345678900@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_username@my.provider.ip;tag=as7d06fc50 To: sip:12345678900@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:12345678900@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_username@my.provider.ip;tag=as7d06fc50 To: sip:12345678900@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:12345678900@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_username@my.provider.ip;tag=as7d06fc50 To: sip:12345678900@my.provider.ip:5068;tag=v9g4HD4vrNFUH Contact:<provider_username@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
On 28/08/14 14:45, Olle E. Johansson wrote:
On 28 Aug 2014, at 14:14, Yuriy Gorlichenko <ovoshlook@gmail.com mailto:ovoshlook@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.
While breaking the cseq incrementation for authentication (mentioned in the readme of uac), the Asterisk seems to do ok here, because the ACK is coming from asterisk, but it is not accepted by the provider.
The provider (having a plivo platform, based on the responses) is running kamailio 4.1.2 in front (looking at 100 trying).
Authentication from kamailio to another kamailio using uac module should work fine, as kamailio doesn't act as end user UAC and doesn't care much of cseq.
I didn't have time to look at the sip trace properly, but Asterisk should have nothing to do with the problem here, unless I missed something from the description.
Cheers, Daniel
All packets (INVITE,ACK,BYE) that comes from Asterisk and sends to Provider handled by Kamailio (changed tU, fU and td and from d). so I write to PLIVO this question, but they still answer to me nothing... As I see my trace there are no simple muistakes (such as wrong dst or wrong contact header).
AboutAsteirsk and Kamailio I think As Daniel that Auth is not Asterisk problem. Furthermore Asterisk works with kamailio without registration on kamailio: ip-based dialog.
So Daniel - If you will have some time to see my trace I will be happy.
Thanks for answers and help.
I will thinkabout problem to and waiting answer.
2014-08-28 16:57 GMT+04:00 Daniel-Constantin Mierla miconda@gmail.com:
On 28/08/14 14:45, Olle E. Johansson wrote:
On 28 Aug 2014, at 14:14, Yuriy Gorlichenko ovoshlook@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.
While breaking the cseq incrementation for authentication (mentioned in the readme of uac), the Asterisk seems to do ok here, because the ACK is coming from asterisk, but it is not accepted by the provider.
The provider (having a plivo platform, based on the responses) is running kamailio 4.1.2 in front (looking at 100 trying).
Authentication from kamailio to another kamailio using uac module should work fine, as kamailio doesn't act as end user UAC and doesn't care much of cseq.
I didn't have time to look at the sip trace properly, but Asterisk should have nothing to do with the problem here, unless I missed something from the description.
Cheers, Daniel
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA
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
You should send a pcap file with all packets, from first incoming INVITE to kamailio. It is important to have both sides of signaling from kamailio point of view, from first packet in that call.
Cheers, Daniel
On 28/08/14 15:11, Yuriy Gorlichenko wrote:
All packets (INVITE,ACK,BYE) that comes from Asterisk and sends to Provider handled by Kamailio (changed tU, fU and td and from d). so I write to PLIVO this question, but they still answer to me nothing... As I see my trace there are no simple muistakes (such as wrong dst or wrong contact header).
AboutAsteirsk and Kamailio I think As Daniel that Auth is not Asterisk problem. Furthermore Asterisk works with kamailio without registration on kamailio: ip-based dialog.
So Daniel - If you will have some time to see my trace I will be happy.
Thanks for answers and help.
I will thinkabout problem to and waiting answer.
2014-08-28 16:57 GMT+04:00 Daniel-Constantin Mierla <miconda@gmail.com mailto:miconda@gmail.com>:
On 28/08/14 14:45, Olle E. Johansson wrote:
On 28 Aug 2014, at 14:14, Yuriy Gorlichenko <ovoshlook@gmail.com <mailto:ovoshlook@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.
While breaking the cseq incrementation for authentication (mentioned in the readme of uac), the Asterisk seems to do ok here, because the ACK is coming from asterisk, but it is not accepted by the provider. The provider (having a plivo platform, based on the responses) is running kamailio 4.1.2 in front (looking at 100 trying). Authentication from kamailio to another kamailio using uac module should work fine, as kamailio doesn't act as end user UAC and doesn't care much of cseq. I didn't have time to look at the sip trace properly, but Asterisk should have nothing to do with the problem here, unless I missed something from the description. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 -http://www.asipto.com Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org <mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
On 28 Aug 2014, at 14:57, Daniel-Constantin Mierla miconda@gmail.com wrote:
On 28/08/14 14:45, Olle E. Johansson wrote:
On 28 Aug 2014, at 14:14, Yuriy Gorlichenko ovoshlook@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.
While breaking the cseq incrementation for authentication (mentioned in the readme of uac), the Asterisk seems to do ok here, because the ACK is coming from asterisk, but it is not accepted by the provider.
You are missing the fact that the ACK sent by Asterisk is already sent by the proxy. The INVITE w/AUth have a different cseq than the ACK.
The provider (having a plivo platform, based on the responses) is running kamailio 4.1.2 in front (looking at 100 trying).
Authentication from kamailio to another kamailio using uac module should work fine, as kamailio doesn't act as end user UAC and doesn't care much of cseq.
Won't the second ACK on the same transaction just be ignored, while it waits for an ACK on the new transaction?
I didn't have time to look at the sip trace properly, but Asterisk should have nothing to do with the problem here, unless I missed something from the description.
/O
Cheers, Daniel
-- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA _______________________________________________ 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
On 28/08/14 15:11, Olle E. Johansson wrote:
On 28 Aug 2014, at 14:57, Daniel-Constantin Mierla <miconda@gmail.com mailto:miconda@gmail.com> wrote:
On 28/08/14 14:45, Olle E. Johansson wrote:
On 28 Aug 2014, at 14:14, Yuriy Gorlichenko <ovoshlook@gmail.com mailto:ovoshlook@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.
While breaking the cseq incrementation for authentication (mentioned in the readme of uac), the Asterisk seems to do ok here, because the ACK is coming from asterisk, but it is not accepted by the provider.
You are missing the fact that the ACK sent by Asterisk is already sent by the proxy. The INVITE w/AUth have a different cseq than the ACK.
Kamailio is doing serial forking in this case, so the first ack is for the first branch that gets 407. This should be as usual for serial/parallel forking.
Then, Kamailio is not increasing the cseq here -- that's the limitation with uac auth module, because the authenticator should normally reject it. But if it is authentication against another kamailio, should just work.
The provider (having a plivo platform, based on the responses) is running kamailio 4.1.2 in front (looking at 100 trying).
Authentication from kamailio to another kamailio using uac module should work fine, as kamailio doesn't act as end user UAC and doesn't care much of cseq.
Won't the second ACK on the same transaction just be ignored, while it waits for an ACK on the new transaction?
It is the same transaction in this case, just two branches from kamailio downstream, which is serial forking case, as mentioned above.
Cheers, Daniel
My pcap file. Daniel sorry for first time sended message to your private mail. Was a mistake.
2014-08-28 17:27 GMT+04:00 Daniel-Constantin Mierla miconda@gmail.com:
On 28/08/14 15:11, Olle E. Johansson wrote:
On 28 Aug 2014, at 14:57, Daniel-Constantin Mierla miconda@gmail.com wrote:
On 28/08/14 14:45, Olle E. Johansson wrote:
On 28 Aug 2014, at 14:14, Yuriy Gorlichenko ovoshlook@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.
While breaking the cseq incrementation for authentication (mentioned in the readme of uac), the Asterisk seems to do ok here, because the ACK is coming from asterisk, but it is not accepted by the provider.
You are missing the fact that the ACK sent by Asterisk is already sent by the proxy. The INVITE w/AUth have a different cseq than the ACK.
Kamailio is doing serial forking in this case, so the first ack is for the first branch that gets 407. This should be as usual for serial/parallel forking.
Then, Kamailio is not increasing the cseq here -- that's the limitation with uac auth module, because the authenticator should normally reject it. But if it is authentication against another kamailio, should just work.
The provider (having a plivo platform, based on the responses) is running kamailio 4.1.2 in front (looking at 100 trying).
Authentication from kamailio to another kamailio using uac module should work fine, as kamailio doesn't act as end user UAC and doesn't care much of cseq.
Won't the second ACK on the same transaction just be ignored, while it waits for an ACK on the new transaction?
It is the same transaction in this case, just two branches from kamailio downstream, which is serial forking case, as mentioned above.
Cheers, Daniel
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA
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
I don't mind to receive directly attachments that can contain sensitive data. But you have to write an email via mailing list saying you do/did so.
With gmail I am checking mostly the emails that are filtered with various rules, otherwise, the rest land together with a lot of spam/advertising and check them very rarely, if ever.
So, as a reminder to anyone that did it -- if a conversation from mailing list was turned into a direct email to me only, it very unlikely to get answered soon, given the load I have -- better resend to mailing list.
I had no time to look yet at the trace.
Cheers, Daniel
On 29/08/14 07:08, Yuriy Gorlichenko wrote:
My pcap file. Daniel sorry for first time sended message to your private mail. Was a mistake.
2014-08-28 17:27 GMT+04:00 Daniel-Constantin Mierla <miconda@gmail.com mailto:miconda@gmail.com>:
On 28/08/14 15:11, Olle E. Johansson wrote:
On 28 Aug 2014, at 14:57, Daniel-Constantin Mierla <miconda@gmail.com <mailto:miconda@gmail.com>> wrote:
On 28/08/14 14:45, Olle E. Johansson wrote:
On 28 Aug 2014, at 14:14, Yuriy Gorlichenko <ovoshlook@gmail.com <mailto:ovoshlook@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.
While breaking the cseq incrementation for authentication (mentioned in the readme of uac), the Asterisk seems to do ok here, because the ACK is coming from asterisk, but it is not accepted by the provider.
You are missing the fact that the ACK sent by Asterisk is already sent by the proxy. The INVITE w/AUth have a different cseq than the ACK.
Kamailio is doing serial forking in this case, so the first ack is for the first branch that gets 407. This should be as usual for serial/parallel forking. Then, Kamailio is not increasing the cseq here -- that's the limitation with uac auth module, because the authenticator should normally reject it. But if it is authentication against another kamailio, should just work.
The provider (having a plivo platform, based on the responses) is running kamailio 4.1.2 in front (looking at 100 trying). Authentication from kamailio to another kamailio using uac module should work fine, as kamailio doesn't act as end user UAC and doesn't care much of cseq.
Won't the second ACK on the same transaction just be ignored, while it waits for an ACK on the new transaction?
It is the same transaction in this case, just two branches from kamailio downstream, which is serial forking case, as mentioned above. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 -http://www.asipto.com Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org <mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
The last .zip file is also showing only 3 packets (different ones) -- can you check the trace has all the packet before you compress? Maybe you can put it somewhere on a server for download and send me the link, so I get it over http.
Cheers, Daniel
On 29/08/14 09:36, Daniel-Constantin Mierla wrote:
I don't mind to receive directly attachments that can contain sensitive data. But you have to write an email via mailing list saying you do/did so.
With gmail I am checking mostly the emails that are filtered with various rules, otherwise, the rest land together with a lot of spam/advertising and check them very rarely, if ever.
So, as a reminder to anyone that did it -- if a conversation from mailing list was turned into a direct email to me only, it very unlikely to get answered soon, given the load I have -- better resend to mailing list.
I had no time to look yet at the trace.
Cheers, Daniel
On 29/08/14 07:08, Yuriy Gorlichenko wrote:
My pcap file. Daniel sorry for first time sended message to your private mail. Was a mistake.
2014-08-28 17:27 GMT+04:00 Daniel-Constantin Mierla <miconda@gmail.com mailto:miconda@gmail.com>:
On 28/08/14 15:11, Olle E. Johansson wrote:
On 28 Aug 2014, at 14:57, Daniel-Constantin Mierla <miconda@gmail.com <mailto:miconda@gmail.com>> wrote:
On 28/08/14 14:45, Olle E. Johansson wrote:
On 28 Aug 2014, at 14:14, Yuriy Gorlichenko <ovoshlook@gmail.com <mailto:ovoshlook@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.
While breaking the cseq incrementation for authentication (mentioned in the readme of uac), the Asterisk seems to do ok here, because the ACK is coming from asterisk, but it is not accepted by the provider.
You are missing the fact that the ACK sent by Asterisk is already sent by the proxy. The INVITE w/AUth have a different cseq than the ACK.
Kamailio is doing serial forking in this case, so the first ack is for the first branch that gets 407. This should be as usual for serial/parallel forking. Then, Kamailio is not increasing the cseq here -- that's the limitation with uac auth module, because the authenticator should normally reject it. But if it is authentication against another kamailio, should just work.
The provider (having a plivo platform, based on the responses) is running kamailio 4.1.2 in front (looking at 100 trying). Authentication from kamailio to another kamailio using uac module should work fine, as kamailio doesn't act as end user UAC and doesn't care much of cseq.
Won't the second ACK on the same transaction just be ignored, while it waits for an ACK on the new transaction?
It is the same transaction in this case, just two branches from kamailio downstream, which is serial forking case, as mentioned above. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 -http://www.asipto.com Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org <mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierla http://twitter.com/#!/miconda -http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 -http://www.asipto.com Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA
How did you grab the pcap? All the variants (including the zip via http download) are not recognized by ngrep or wireshark.
Anyhow, I looked with a text editor inside the and I could see that the ACK is changing the r-uri -- it comes to kamailio with the contact from 200ok, but goes out with a different hostname. You have some rules in kamailio.cfg that do that -- you should let the r-uri unchanged for ack -- routing is done via record-routing.
Cheers, Daniel
On 29/08/14 16:12, Daniel-Constantin Mierla wrote:
The last .zip file is also showing only 3 packets (different ones) -- can you check the trace has all the packet before you compress? Maybe you can put it somewhere on a server for download and send me the link, so I get it over http.
Cheers, Daniel
On 29/08/14 09:36, Daniel-Constantin Mierla wrote:
I don't mind to receive directly attachments that can contain sensitive data. But you have to write an email via mailing list saying you do/did so.
With gmail I am checking mostly the emails that are filtered with various rules, otherwise, the rest land together with a lot of spam/advertising and check them very rarely, if ever.
So, as a reminder to anyone that did it -- if a conversation from mailing list was turned into a direct email to me only, it very unlikely to get answered soon, given the load I have -- better resend to mailing list.
I had no time to look yet at the trace.
Cheers, Daniel
On 29/08/14 07:08, Yuriy Gorlichenko wrote:
My pcap file. Daniel sorry for first time sended message to your private mail. Was a mistake.
2014-08-28 17:27 GMT+04:00 Daniel-Constantin Mierla <miconda@gmail.com mailto:miconda@gmail.com>:
On 28/08/14 15:11, Olle E. Johansson wrote:
On 28 Aug 2014, at 14:57, Daniel-Constantin Mierla <miconda@gmail.com <mailto:miconda@gmail.com>> wrote:
On 28/08/14 14:45, Olle E. Johansson wrote:
On 28 Aug 2014, at 14:14, Yuriy Gorlichenko <ovoshlook@gmail.com <mailto:ovoshlook@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.
While breaking the cseq incrementation for authentication (mentioned in the readme of uac), the Asterisk seems to do ok here, because the ACK is coming from asterisk, but it is not accepted by the provider.
You are missing the fact that the ACK sent by Asterisk is already sent by the proxy. The INVITE w/AUth have a different cseq than the ACK.
Kamailio is doing serial forking in this case, so the first ack is for the first branch that gets 407. This should be as usual for serial/parallel forking. Then, Kamailio is not increasing the cseq here -- that's the limitation with uac auth module, because the authenticator should normally reject it. But if it is authentication against another kamailio, should just work.
The provider (having a plivo platform, based on the responses) is running kamailio 4.1.2 in front (looking at 100 trying). Authentication from kamailio to another kamailio using uac module should work fine, as kamailio doesn't act as end user UAC and doesn't care much of cseq.
Won't the second ACK on the same transaction just be ignored, while it waits for an ACK on the new transaction?
It is the same transaction in this case, just two branches from kamailio downstream, which is serial forking case, as mentioned above. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 -http://www.asipto.com Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org <mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierla http://twitter.com/#!/miconda -http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 -http://www.asipto.com Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA
-- Daniel-Constantin Mierla http://twitter.com/#!/miconda -http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 -http://www.asipto.com Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA
Apparently the file got corrupted, most probably by email client/server encoding, and it shows only three packets.
Can you resend it compress as tgz?
Cheers, Daniel
On 29/08/14 07:08, Yuriy Gorlichenko wrote:
My pcap file. Daniel sorry for first time sended message to your private mail. Was a mistake.
2014-08-28 17:27 GMT+04:00 Daniel-Constantin Mierla <miconda@gmail.com mailto:miconda@gmail.com>:
On 28/08/14 15:11, Olle E. Johansson wrote:
On 28 Aug 2014, at 14:57, Daniel-Constantin Mierla <miconda@gmail.com <mailto:miconda@gmail.com>> wrote:
On 28/08/14 14:45, Olle E. Johansson wrote:
On 28 Aug 2014, at 14:14, Yuriy Gorlichenko <ovoshlook@gmail.com <mailto:ovoshlook@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.
While breaking the cseq incrementation for authentication (mentioned in the readme of uac), the Asterisk seems to do ok here, because the ACK is coming from asterisk, but it is not accepted by the provider.
You are missing the fact that the ACK sent by Asterisk is already sent by the proxy. The INVITE w/AUth have a different cseq than the ACK.
Kamailio is doing serial forking in this case, so the first ack is for the first branch that gets 407. This should be as usual for serial/parallel forking. Then, Kamailio is not increasing the cseq here -- that's the limitation with uac auth module, because the authenticator should normally reject it. But if it is authentication against another kamailio, should just work.
The provider (having a plivo platform, based on the responses) is running kamailio 4.1.2 in front (looking at 100 trying). Authentication from kamailio to another kamailio using uac module should work fine, as kamailio doesn't act as end user UAC and doesn't care much of cseq.
Won't the second ACK on the same transaction just be ignored, while it waits for an ACK on the new transaction?
It is the same transaction in this case, just two branches from kamailio downstream, which is serial forking case, as mentioned above. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 -http://www.asipto.com Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org <mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users