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
From: "John" <sip:provider_username@my.provider.ip>;tag=as7d06fc50
To: <sip:12345678900@my.provider.ip:5068>;tag=v9g4HD4vrNFUH
CSeq: 102 INVITE
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...]...@...
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
Max-Forwards: 70
From: "John" <sip:provider_username@my.provider.ip>;tag=as7d06fc50
To: <sip:12345678900@my.provider.ip:5068>;tag=v9g4HD4vrNFUH
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 
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