[asterisk-bugs] [JIRA] (ASTERISK-23106) pjsip: ACK to 200 OK sent to private IP address on outbound channel's INVITE request

[Kinsey Moore (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-23106?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=213989#comment-213989 ] 

Kinsey Moore commented on ASTERISK-23106:
-----------------------------------------

I have managed to reproduce this in the testsuite with only the contact address being private and only the 200 OK being sent back without the 100 and 180. This problem does not occur if the 200 has the correct address in the contact header regardless of the address in provisional messages.
                
> pjsip: ACK to 200 OK sent to private IP address on outbound channel's INVITE request
> ------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-23106
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-23106
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_pjsip, Resources/res_pjsip_nat
>    Affects Versions: 12.0.0
>            Reporter: Matt Jordan
>            Assignee: Kinsey Moore
>         Attachments: freepbx_grandstream.tar.gz
>
>
> Note: this only happens with GrandStream devices.
> Scenario:
> * Dial between two GrandStream phones
> * Outbound channel's device sends 200 OK
> * Asterisk sends ACK to the private IP address of the outbound channel's device
> Note that we do send the 200 OK to the inbound channel device's public IP address.
> {noformat}
> [Jan  6 16:36:13] VERBOSE[19775] res_pjsip_logger.c: <--- Received SIP response (846 bytes) from UDP:74.87.121.99:53997 --->
> ÿSIP/2.0 200 OK
> ÿVia: SIP/2.0/UDP 199.102.239.103:5060;rport=5060;branch=z9hG4bKPj533c2053-eb63-4896-a67d-d190de484d09
> ÿFrom: "GXV3175" <sip:5003 at 199.102.239.103>;tag=7ef05a61-4081-4bcd-ad09-ec82e4cf11bf
> ÿTo: <sip:5002 at 74.87.121.99>;tag=547897211
> ÿCall-ID: 475af947-6311-45a8-b058-49ec80e4a15e
> ÿCSeq: 3964 INVITE
> ÿContact: <sip:5002 at 10.4.0.148:53997>
> ÿSupported: replaces, path, timer, eventlist
> ÿUser-Agent: Grandstream GXV3140 1.0.7.74
> ÿSession-Expires: 1800;refresher=uac
> ÿRequire: timer
> ÿAllow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
> ÿContent-Type: application/sdp
> ÿContent-Length:   208
> ÿ
> ÿv=0
> ÿo=5002 8000 8000 IN IP4 10.4.0.148
> ÿs=SIP Call
> ÿc=IN IP4 10.4.0.148
> ÿt=0 0
> ÿm=audio 36682 RTP/AVP 0 101
> ÿa=sendrecv
> ÿa=rtpmap:0 PCMU/8000
> ÿa=ptime:20
> ÿa=rtpmap:101 telephone-event/8000
> ÿa=fmtp:101 0-15
> ÿ
> [Jan  6 16:36:13] VERBOSE[19776] res_pjsip_logger.c: <--- Transmitting SIP request (407 bytes) to UDP:10.4.0.148:53997 --->
> ÿACK sip:5002 at 10.4.0.148:53997 SIP/2.0
> ÿVia: SIP/2.0/UDP 199.102.239.103:5060;rport;branch=z9hG4bKPj0000bcab-f6c6-4ab6-bf2d-df9b08839477
> ÿFrom: "GXV3175" <sip:5003 at 199.102.239.103>;tag=7ef05a61-4081-4bcd-ad09-ec82e4cf11bf
> ÿTo: <sip:5002 at 74.87.121.99>;tag=547897211
> ÿCall-ID: 475af947-6311-45a8-b058-49ec80e4a15e
> ÿCSeq: 3964 ACK
> ÿMax-Forwards: 70
> ÿUser-Agent: FPBX-12.0.1alpha6(12.0.0)
> ÿContent-Length:  0
> ÿ
> ÿ
> {noformat}

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