[asterisk-bugs] [JIRA] (ASTERISK-26386) chan_sip: ACK being sent is referencing the contact on the Request-URI instead of To

[Rajiv Duggal (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26386?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=232721#comment-232721 ] 

Rajiv Duggal commented on ASTERISK-26386:
-----------------------------------------

Hi Walter,

Thanks for your details on this. At this time we can close this ticket as it does not seem to be a asterisk issue. At this time I can replicate my issue on Asterisk11 as well.
Attaching 2 logs once where call works fine when calling from cell phone to a DID that points back to our PBX.
The other when we call from a local extensions or a sip trunk and it disconnects or goes silent.
Both logs are uploaded for your review. My sip provider is looking at their routing at this time.

Thanks for your help, much appreciated
.Rajiv.

> chan_sip: ACK being sent is referencing the contact on the Request-URI instead of To
> ------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-26386
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26386
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General, Channels/chan_sip/Interoperability
>    Affects Versions: 13.2.0
>         Environment: 64bit Linux
>            Reporter: Rajiv Duggal
>            Assignee: Unassigned
>         Attachments: sip.log
>
>
> per RFC protocol (https://tools.ietf.org/html/rfc3261#page-129) 
> the generated ACK should use the TO from the response being acknowledged
> This is causing call disconnects and continuous invites from some SIP providers who receive a contact from upstream which is not identical to the TO part of the header.
> This can be replicated for phone # 15033277805.
> received header per below
> {noformat}
> 17:03:36.856392 IP (tos 0x0, ttl 53, id 37874, offset 0, flags [none], proto UDP (17), length 1057)
> 65.254.44.194.sip > 10.2.1.40.sip-tls: [udp sum ok] SIP, length: 1029
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 100.32.192.11:5061;received=100.32.192.11;branch=z9hG4bK3da029f8;rport=5061
> From: <sip:8053881711 at gw1.sip.us:5061>;tag=as28725f49
> To: <sip:15033277805 at gw1.sip.us:5060>;tag=gK0ce47da4
> Call-ID: 0e760a970d24e31e22afca333f10989e at gw1.sip.us
> CSeq: 103 INVITE
> Record-Route: <sip:67.231.0.87:5060;lr=on;ftag=as28725f49>
> Record-Route: <sip:65.254.44.194:5060;lr=on;ftag=as28725f49;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAADo1MDYx;proxy_media=yes;dlgcor=018.b1f>
> Accept: application/sdp
> Contact: <sip:+15033277805 at 192.168.16.8:5060>
> Allow: INVITE,ACK,CANCEL,BYE,REFER,PRACK,OPTIONS
> Supported: timer
> Session-Expires: 1800;refresher=uas
> Content-Length: 258
> Content-Disposition: session; handling=required
> Content-Type: application/sdp
> {noformat}
> Note contact has a +prefixed above and below ACK generated by the system.
> {noformat}
> 17:03:36.857085 IP (tos 0x60, ttl 64, id 26156, offset 0, flags [none], proto UDP (17), length 629)
> 10.2.1.40.sip-tls > 65.254.44.194.sip: [udp sum ok] SIP, length: 601
> ACK sip:+15033277805 at 192.168.16.8:5060 SIP/2.0
> Via: SIP/2.0/UDP 100.32.192.11:5061;branch=z9hG4bK148ab810;rport
> Route: <sip:65.254.44.194:5060;lr=on;ftag=as28725f49;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAADo1MDYx
> ;proxy_media=yes;dlgcor=018.b1f>,<sip:67.231.0.87:5060;lr=on;ftag=as28725f49>
> Max-Forwards: 70
> From: <sip:8053881711 at gw1.sip.us:5061>;tag=as28725f49
> To: <sip:15033277805 at gw1.sip.us:5060>;tag=gK0ce47da4
> Contact: <sip:8053881711 at 100.32.192.11:5061>
> Call-ID: 0e760a970d24e31e22afca333f10989e at gw1.sip.us
> CSeq: 103 ACK
> User-Agent: FPBX-AsteriskNOW-12.0.76.4(13.2.0)
> Content-Length: 0
> {noformat}
> this is causing issues with some SIP providers. Complete log is attached.



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