[SR-Users] ACK does not match transaction after stripping Route header
Daniel-Constantin Mierla
miconda at gmail.com
Wed Jun 8 22:49:38 CEST 2016
Hello,
the problem is that the ACK doesn't have in R-URI the address from
Contact header in 200ok, as required by RFC.
The R-URI and the Route are both having the IP of Kamailio, so there is
no other address where to send the ACK.
ACK sip:10.15.1.30:5060 SIP/2.0
*Route: sip:10.15.1.30;lr;ftag=SDkbo9901-42090
You have to track why the ACK is not coming with R-URI having the
address from Contact of 200ok. See how 200ok is received by caller and
how caller's devices sets the R-URI. Might be the sbc or other hop
breaking it.
If you can't fix the device/application messing up the R-URI for the
ACK, then you can try some solutions in kamailio:
1) use set_contact_alias() for 18x/200ok responses and then
handle_ruri_alias() for ACK and other requests within dialog.
if(has_totag() && uri==myself && $ru=~";alias=") {
handle_ruri_alias();
if($du != $null) {
$ru = $du;
}
}
# continue with loose_route(), etc ...
It works if the devices messes up only the host/port of the R-URI in
ACK, but keeps the other parameters.
2) if 1 doesn't work, use htable to store association between
callid+from-tag and the contact address of 200ok, then use it for
requests within dialog that have uri==myself
reply_route {
...
if(is_method("INVITE") && status=="200") {
$sht(ct=>$ci::$ft) = $sel(contact.uri);
}
...
}
request_route {
...
if(has_totag() && uri==myself) {
if($sht(ct=>$ci::$ft) != $null) {
$ru = $sht(ct=>$ci::$ft);
}
}
# continue with loose_route(), etc ...
...
}
You have to set the auto-expire for htable ct to the maximum lifetime
for a dialog. You can delete the items from ct hash table (or reduce
their expire value) when processing the BYE or the response to BYE. Note
that for bye you have to try the combination $ci:$ft as well as $ci:$tt
3) if you have only one asterisk, then if the request has to-tag and a
route header and uri==myself, then just set $ru to sip:ip:port of asterisk
Cheers,
Daniel
On 08/06/16 12:18, Francisco Valentin Vinagrero wrote:
>
> Hi there,
>
>
>
> I’m having an issue in a SBC (ACME) -> KAMAILIO -> Asterisk scenario
> with an ACK that gets ignored in Kamailio because it does not match
> any transaction.
>
>
>
> The INVITE coming from the SBC looks like this (only relevant headers
> and hidden numbers for simplicity – SBC has IP .12 , Kamailio .30 and
> Asterisk .34)
>
>
>
> INVITE sip:mynumber at 10.15.1.30:5060
> SIP/2.0
>
> Via: SIP/2.0/UDP
> 10.15.1.12:5060;branch=z9hG4bKdd1m7b00aom47rggc700.1
>
> To: <sip:
> mynumber at 10.15.1.30:5060>
>
> From:
> <sip:a-number at 10.15.1.12;user=phone>;tag=SDkbo9901-42090
>
> P-Asserted-Identity: <sip: a-number
> @10.15.1.12>
>
> Call-ID:
> SDkbo9901-71a1d17456b829b4c422af61de9eee7e-ao32g50
>
> CSeq: 1
> INVITE
>
> Contact:
> <sip:41754112601 at 10.15.1.12:5060;transport=udp>
>
>
>
> And its forwarded to Asterisk with the Record-Route header:
>
>
>
>
>
> INVITE sip: mynumber @10.15.1.30:5060 SIP/2.0
>
> *Record-Route: <sip:10.15.1.30;lr=on;ftag=SDkbo9901-42090>
>
> *Via: SIP/2.0/UDP
> 10.15.1.30;branch=z9hG4bK1c02.7dc1b94be22d8780df5141f9ba3c5b7b.0
>
> Via: SIP/2.0/UDP
> 10.15.1.12:5060;branch=z9hG4bKdd1m7b00aom47rggc700.1
>
> To: <sip:mynumber at 10.15.1.30:5060>
>
> From: <sip:a-number at 10.15.1.12;user=phone>;tag=SDkbo9901-42090
>
> P-Asserted-Identity: <sip:a-number at 10.15.1.12>
>
> Call-ID: SDkbo9901-71a1d17456b829b4c422af61de9eee7e-ao32g50
>
> CSeq: 1 INVITE
>
> Contact: <sip:a-number at 10.15.1.12:5060;transport=udp>
>
>
>
>
>
> Then, 200 OK from Asterisk:
>
>
>
> SIP/2.0 200 OK
>
> *Via: SIP/2.0/UDP
> 10.15.1.30;rport=5060;received=10.15.1.30;branch=z9hG4bK1c02.7dc1b94be22d8780df5141f9ba3c5b7b.0
>
> Via: SIP/2.0/UDP
> 10.15.1.12:5060;branch=z9hG4bKdd1m7b00aom47rggc700.1
>
> *Record-Route: <sip:10.15.1.30;lr;ftag=SDkbo9901-42090>
>
> Call-ID: SDkbo9901-71a1d17456b829b4c422af61de9eee7e-ao32g50
>
> From: <sip:a-number at 10.15.1.12;user=phone>;tag=SDkbo9901-42090
>
> To:
> <sip:mynumber at 10.15.1.30>;tag=2e3c2071-c895-4069-afc2-37a19b20637a
>
> CSeq: 1 INVITE
>
> Server: Asterisk PBX 13.8.0
>
> Contact: <sip:10.15.1.34:5060>
>
>
>
> Which is sent to the SBC like this:
>
>
>
> SIP/2.0 200 OK
>
> Via: SIP/2.0/UDP
> 10.15.1.12:5060;branch=z9hG4bKdd1m7b00aom47rggc700.1
>
> *Record-Route: <sip:10.15.1.30;lr;ftag=SDkbo9901-42090>
>
> Call-ID: SDkbo9901-71a1d17456b829b4c422af61de9eee7e-ao32g50
>
> From: <sip:a-number at 10.15.1.12;user=phone>;tag=SDkbo9901-42090
>
> To:
> <sip:mynumber at 10.15.1.30>;tag=2e3c2071-c895-4069-afc2-37a19b20637a
>
> CSeq: 1 INVITE
>
> Server: Asterisk PBX 13.8.0
>
> Contact: <sip:10.15.1.34:5060>
>
>
>
>
>
> And finally the SBC sends the ACK:
>
>
>
> ACK sip:10.15.1.30:5060 SIP/2.0
>
> Via: SIP/2.0/UDP
> 10.15.1.12:5060;branch=z9hG4bKdt7p9k00dounet8ic600.1
>
> To:
> <sip:mynumber at 10.15.1.30>;tag=2e3c2071-c895-4069-afc2-37a19b20637a
>
> From: <sip:a-number at 10.15.1.12;user=phone>;tag=SDkbo9901-42090
>
> Call-ID: SDkbo9901-71a1d17456b829b4c422af61de9eee7e-ao32g50
>
> CSeq: 1 ACK
>
> Contact: <sip:a-number at 10.15.1.12:5060;transport=udp>
>
> *Route: sip:10.15.1.30;lr;ftag=SDkbo9901-42090
> <sip:10.15.1.30;lr;ftag=SDkbo9901-42090>
>
>
>
>
>
> The problem: this ACK gets not retransmitted to Asterisk
>
>
>
> At first, I thought it was some sanity check but after disabling that
> I realized that it was in the WITHINDLG route.
>
>
>
> For the incoming ACK I get in the logs:
>
>
>
> Jun 8 11:56:47 tone-0866-fe-2-qa /usr/local/sbin/kamailio[53240]:
> ALERT: <script>: Inside LOOSE route for ACK proto=UDP trans=4194304
> from=sip:00754112601 at 10.15.1.12;user=phone
> route=sip:10.15.1.30;lr;ftag=SDkbo9901-42090 src_ip=10.15.1.12
>
>
>
>
>
> And once the ACK is ready to be sent to Asterisk, the Route header has
> been removed and no Record-Route has been added so it fails.
>
>
>
> Jun 8 11:56:44 tone-0866-fe-2-qa /usr/local/sbin/kamailio[53238]:
> INFO: rr [rr_mod.c:402]: pv_get_route_uri_f(): No route header present.
>
> Jun 8 11:56:44 tone-0866-fe-2-qa /usr/local/sbin/kamailio[53238]:
> ALERT: <script>: ACK does not match transaction!! proto=UDP
> trans=4194304 from=sip:00754112601 at 10.15.1.12;user=phone route=
> src_ip=10.15.1.30
>
>
>
>
>
> My WITHINDLG route looks like this:
>
>
>
>
>
> # Handle requests within SIP
> dialogs
>
> route[WITHINDLG] {
>
>
> if (has_totag())
> {
>
> # sequential request withing a dialog
> should
>
> # take the path determined by
> record-routing
>
> if (loose_route())
> {
>
> if (is_method("BYE"))
> {
>
> xlog("L_ALERT","Inside LOOSE
> route\n");
>
> setflag(FLT_ACC); # do accounting
> ...
>
>
> setflag(FLT_ACCFAILED); # ... even if the transaction
> fails
>
> }
>
>
> if ( is_method("ACK") )
> {
>
> xlog("L_ALERT","Inside LOOSE route for ACK proto=$rP
> trans=$mf from=$fu route=$route_uri src_ip=$si \n");
>
> # ACK is forwarded
> statelessy
>
> route(NATMANAGE);
>
>
> }
>
>
> route(RELAY);
>
>
> } else
> {
>
> if (is_method("SUBSCRIBE") && uri == myself)
> {
>
> # in-dialog subscribe
> requests
>
>
> route(PRESENCE);
>
>
> exit;
>
>
> }
>
>
> if ( is_method("ACK") )
> {
>
>
> if ( t_check_trans() )
> {
>
> # no loose-route, but stateful
> ACK;
>
>
> # must be an ACK after a
> 487
>
> # or e.g. 404 from upstream server
>
>
> t_relay();
>
>
> exit;
>
>
> } else
> {
>
> # ACK without matching transaction ... ignore and
> discard
>
> xlog("L_ALERT","ACK does not match transaction!!
> proto=$rP trans=$mf from=$fu route=$route_uri src_ip=$si \n");
>
> exit;
>
>
> }
>
>
> }
>
>
> sl_send_reply("404","Not
> here");
>
>
> }
>
>
> exit;
>
>
> }
>
>
> }
>
>
>
>
>
>
> Thanks for reading this J Any idea about how to validate the
> transaction? t_check_trans is not being validated…
>
>
>
> Cheers, Francisco.
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla
http://www.asipto.com - http://www.kamailio.org
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
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