[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=214525#comment-214525 ] 

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

I finally managed to get a trace in a similar scenario (albeit using chan_sip) and I am working on labbing this up to attempt to reproduce with actual phones under Asterisk 12/chan_pjsip.

The biggest difference I see is that for the grandstream's response the addresses in the To: and Contact: headers differ while in the preliminary trace I got that works (SNOMs), the addresses in the To: and Contact: headers are identical.
                
> 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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