Thanks Daniel, you’re totally right! I hadn’t spotted the wrong IP in the ACK, so I’m trying to correct that with our provider ;)
If it doesn’t, I’d go for solution number 2 as we have multiple Asterisk behind each Kamailio. Cheers, Francisco. From: Daniel-Constantin Mierla [mailto:mico...@gmail.com] Sent: 08 June 2016 22:50 To: Kamailio (SER) - Users Mailing List <sr-users@lists.sip-router.org>; Francisco Valentin Vinagrero <francisco.valentin.vinagr...@cern.ch> Subject: Re: [SR-Users] ACK does not match transaction after stripping Route header 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@10.15.1.30:5060 SIP/2.0 Via: SIP/2.0/UDP 10.15.1.12:5060;branch=z9hG4bKdd1m7b00aom47rggc700.1 To: <sip: mynumber@10.15.1.30:5060><sip:mynumber@10.15.1.30:5060> From: <sip:a-number@10.15.1.12;user=phone><sip:a-number@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@10.15.1.12:5060;transport=udp><sip:41754112601@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><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@10.15.1.30:5060><sip:mynumber@10.15.1.30:5060> From: <sip:a-number@10.15.1.12;user=phone><sip:a-number@10.15.1.12;user=phone>;tag=SDkbo9901-42090 P-Asserted-Identity: <sip:a-number@10.15.1.12><sip:a-number@10.15.1.12> Call-ID: SDkbo9901-71a1d17456b829b4c422af61de9eee7e-ao32g50 CSeq: 1 INVITE Contact: <sip:a-number@10.15.1.12:5060;transport=udp><sip:a-number@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><sip:10.15.1.30;lr;ftag=SDkbo9901-42090> Call-ID: SDkbo9901-71a1d17456b829b4c422af61de9eee7e-ao32g50 From: <sip:a-number@10.15.1.12;user=phone><sip:a-number@10.15.1.12;user=phone>;tag=SDkbo9901-42090 To: <sip:mynumber@10.15.1.30><sip:mynumber@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><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><sip:10.15.1.30;lr;ftag=SDkbo9901-42090> Call-ID: SDkbo9901-71a1d17456b829b4c422af61de9eee7e-ao32g50 From: <sip:a-number@10.15.1.12;user=phone><sip:a-number@10.15.1.12;user=phone>;tag=SDkbo9901-42090 To: <sip:mynumber@10.15.1.30><sip:mynumber@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><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@10.15.1.30><sip:mynumber@10.15.1.30>;tag=2e3c2071-c895-4069-afc2-37a19b20637a From: <sip:a-number@10.15.1.12;user=phone><sip:a-number@10.15.1.12;user=phone>;tag=SDkbo9901-42090 Call-ID: SDkbo9901-71a1d17456b829b4c422af61de9eee7e-ao32g50 CSeq: 1 ACK Contact: <sip:a-number@10.15.1.12:5060;transport=udp><sip:a-number@10.15.1.12:5060;transport=udp> *Route: 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@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@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 ☺ 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@lists.sip-router.org<mailto:sr-users@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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