[asterisk-bugs] [JIRA] (ASTERISK-26386) chan_sip: ACK being sent is referencing the contact on the Request-URI instead of To
Walter Doekes (JIRA)
noreply at issues.asterisk.org
Thu Oct 6 15:11:01 CDT 2016
[ https://issues.asterisk.org/jira/browse/ASTERISK-26386?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=232566#comment-232566 ]
Walter Doekes edited comment on ASTERISK-26386 at 10/6/16 3:09 PM:
-------------------------------------------------------------------
The problem being restated is that for the below INVITE from sip provider
{noformat}
To: <sip:15033277805 at gw1.sip.us:5060>;tag=gK0ce47da4
..
Contact: <sip:+15033277805 at 192.168.16.8:5060 SIP/2.0
{noformat}
the ACK generated by asterisk is per below
{noformat}
ACK sip:+15033277805 at 192.168.16.8:5060 SIP/2.0
..
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)
{noformat}
Per RFC 3261 Section 17.1.1.3:
{quote}
"...The ACK request constructed by the client transaction MUST contain
values for the Call-ID, From, and Request-URI that are equal to the
values of those header fields in the request passed to the transport
by the client transaction (call this the "original request")"
{quote}
So ACK should like per below
{noformat}
ACK sip:15033277805 at 192.168.16.8:5060 SIP/2.0
..
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)
{noformat}
thanks
Rajiv
was (Author: rduggal at dex.com):
The problem being restated is that for the below INVITE from sip provider
To: <sip:15033277805 at gw1.sip.us:5060>;tag=gK0ce47da4
..
Contact: <sip:+15033277805 at 192.168.16.8:5060 SIP/2.0
the ACK generated by asterisk is per below
ACK sip:+15033277805 at 192.168.16.8:5060 SIP/2.0
..
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)
Per RFC 3261 Section 17.1.1.3:
"...The ACK request constructed by the client transaction MUST contain
values for the Call-ID, From, and Request-URI that are equal to the
values of those header fields in the request passed to the transport
by the client transaction (call this the "original request")"
So ACK should like per below
ACK sip:15033277805 at 192.168.16.8:5060 SIP/2.0
..
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)
thanks
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.
--
This message was sent by Atlassian JIRA
(v6.2#6252)
More information about the asterisk-bugs
mailing list