[SR-Users] ACK Relay Issue Transaction Not Found
me.projects at yahoo.com
me.projects at yahoo.com
Tue Nov 24 18:22:52 CET 2020
Thanks for your reply. I can't see the ACK being relayed to MSTeams in network capture through in Relay Route
if(is_method("ACK|BYE")) { sip_trace(); setflag(22); } but call has been established successfully. is ACK is being relayed hiddenly by Kamailio ?
On Tuesday, November 24, 2020, 09:35:32 PM GMT+5, Alex Balashov <abalashov at evaristesys.com> wrote:
Hi,
If you take a look at the documentation for t_check_trans(), you'll see
that it behaves in very different ways depending on the type of message
it is handling, and is supposed to return a negative/false value in some
scenarios or abort script execution -- context is very important:
https://kamailio.org/docs/modules/stable/modules/tm.html#tm.f.t_check_trans
There are two types of ACKs - hop-by-hop ACKs, which acknowledge
negative final replies, and end-to-end ACKs, which are for successful
final 2xx replies and meant to be relayed by Kamailio end-to-end, in the
manner of a normal in-dialog request.
An ACK of this nature is a separate request and a separate transaction,
though its CSeq is identical to that of the INVITE transaction it is
meant to acknowledge. However, a 2xx final reply will close an INVITE
transaction, subject to `max_inv_lifetime` and `wt_timer`[1] and all
that stuff:
The documentation speaks to this point:
"For end-to-end ACKs (ACKs to 2xx responses for forwarded
INVITE transactions) it will return true if the corresponding
INVITE transaction is found and still active and false if not.
Note that the e2e ACK matching is more of a hint then a
certainty. A delayed e2e ACK might arrive after the
transaction wait time elapses, when the INVITE transaction
no longer exists and thus would not match anything. There
are also cases when tm would not keep all the information
needed for e2e ACK matching (since this is not needed for
a stateful proxy and it requires additional memory,
tm will not keep this information unless needed by
some other module or callbacks)."
So, in other words, you should not expect t_check_trans() to return true
for an end-to-end ACK. It is reasonable to expect that in the vast
majority of cases, it will return false. The ACK should be relayed in
the same manner as any other loose-routed in-dialog request, e.g. a
reinvite or a BYE. If you follow the logic of the stock configuration
that Kamailio ships with, this will happen correctly by default.
-- Alex
[1] https://kamailio.org/docs/modules/stable/modules/tm.html#tm.p.wt_timer
On 11/24/20 3:24 AM, me.projects at yahoo.com wrote:
> Hi When ACK is received on Kamailio from PBX t_check_trans() fails.
> 200 OK sent PBX from Teams and ACK from PBX is provided here.
>
> What could be the issue.
>
> SIP/2.0 200 OK
> FROM: "Ext1"<sip:1102 at PBX_IP>;tag=e08f0a03-1120-4e13-b0e2-90d1a027c05f
> TO: <sip:1101 at KAM_IP>;tag=e667397cb47c457a860222931e320b41
> CSEQ: 25772 INVITE
> CALL-ID: a6685754-b402-4a15-82b2-5fb2551e5b58
> VIA: SIP/2.0/UDP
> PBX_IP:7790;received=PBX_IP;branch=z9hG4bKPj170912fc-0908-40bb-8248-9f6cb560332d;rport=7790
>
> RECORD-ROUTE:
> <sip:sip-du-a-us.pstnhub.microsoft.com:5061;transport=tls;lr>,<sip:sbc.callsprout.com;transport=tls;ftag=e08f0a03-1120-4e13-b0e2-90d1a027c05f;lr>,<sip:KAM_IP:7790;transport=udp;ftag=e08f0a03-1120-4e13-b0e2-90d1a027c05f;lr>
>
> CONTACT:
> <sip:api-du-b-usea.pstnhub.microsoft.com:443;x-i=c825abba-7203-442c-8461-994522d14365;x-c=25f3c2ce3a635971b809dc3be872e51b/s/1/63c9eea743284db99366cabf5f7dcbf5;ias=52.114.132.46~5061~3>
>
> CONTENT-LENGTH: 353
> SUPPORTED: timer
> CONTENT-TYPE: application/sdp
> ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY
> REQUIRE: timer
> SESSION-EXPIRES: 1800;refresher=uac
> SERVER: Microsoft.PSTNHub.SIPProxy v.2020.11.16.5 i.USEA.5
> X-Siptrace-Fromip: tls:KAM_IP:7790
> X-Siptrace-Toip: udp:PBX_IP:7790
> X-Siptrace-Time: 1606205334 607029
> X-Siptrace-Method: INVITE
> X-Siptrace-Dir: out
>
> ACK
> sip:api-du-b-usea.pstnhub.microsoft.com:443;x-i=c825abba-7203-442c-8461-994522d14365;x-c=25f3c2ce3a635971b809dc3be872e51b/s/1/63c9eea743284db99366cabf5f7dcbf5;alias=.114.132.46~5061~3
> SIP/2.0
> Via: SIP/2.0/UDP
> PBX_IP:7790;rport;branch=z9hG4bKPjaa30c14c-c965-473d-bc49-9f223f9f634c
> From: "Ext1" <sip:1102 at PBX_IP>;tag=e08f0a03-1120-4e13-b0e2-90d1a027c05f
> To: <sip:1101 at KAM_IP>;tag=e667397cb47c457a860222931e320b41
> Call-ID: a6685754-b402-4a15-82b2-5fb2551e5b58
> CSeq: 25772 ACK
> Route: <sip:KAM_IP:7790;lr;ftag=e08f0a03-1120-4e13-b0e2-90d1a027c05f>
> Route:
> <sip:sbc.callsprout.com;transport=tls;lr;ftag=e08f0a03-1120-4e13-b0e2-90d1a027c05f>
>
> Route: <sip:sip-du-a-us.pstnhub.microsoft.com:5061;transport=tls;lr>
> Max-Forwards: 70
> User-Agent: Asterisk
> Content-Length: 0
> X-Siptrace-Fromip: udp:PBX_IP:7790
> X-Siptrace-Toip: udp:KAM_IP:7790
> X-Siptrace-Time: 1606205334 615495
> X-Siptrace-Method: ACK
> X-Siptrace-Dir: in
>
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>
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
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