[SR-Users] loose_route() and FQDN

David Villasmil david.villasmil.work at gmail.com
Fri May 7 16:07:24 CEST 2021


Did you add the handle_ruri_alias() as suggested by Daniel? I had something
like this where I would get “unable to resolve blah blah blah" and it’s
because the RURI is the actual wss “address” which is unresolvable, so
executing the function forces kamailio to take the alias instead.


On Fri, 7 May 2021 at 13:48, Igor Olhovskiy <igorolhovskiy at gmail.com> wrote:

> Daniel,
>
> Seems to be it's really the case, but with other function
>
> With FQDN in RR
> is_first_hop()
>
> is not acting correctly for reply.
>
> For incoming SIP replies, it means that top Record-Route URI is 'myself'
> and source address is not matching it
>
> But in Record-Route we have "myself", but *is_first_hop()* returning 0.
>
> Thanks!
>
> Regards,
> Igor
>
> On 07.05.2021 14:22, Daniel-Constantin Mierla wrote:
>
> OK, because looping was something that should not have happened in this
> case.
>
> Then the problem is that you do not do nat-traversal-like processing for
> websocket/webrtc traffic. You have to use set_contact_alias() +
> handle_ruri_alias() because the webrtc endpoints do not set "valid" contact
> addresses.
>
> Cheers,
> Daniel
> On 07.05.21 14:13, Igor Olhovskiy wrote:
>
> Ah, no, sorry, I was wrong at this one,
>
> This just is not sent with "unable to resolve address
> toleivu2gdbh.invalid".
>
> Sorry. Looping were something else during my tests, this just with
> *advertise* added
>
> Regards,
> Igor
>
> On 07.05.2021 14:02, Daniel-Constantin Mierla wrote:
>
> This looks like incoming ACK, because there is only one Via header, so it
> is not what proxy forwards -- that one is relevant to see what headers were
> consumed and added.
>
> Cheers,
> Daniel
> On 07.05.21 13:51, Igor Olhovskiy wrote:
>
> Sure.
>
> ACK sip:88290 at toleivu2gdbh.invalid;transport=wss SIP/2.0
> Via: SIP/2.0/UDP
> A_IP_ADDRESS:5060;rport;branch=z9hG4bKPj8d05548a-91ef-4332-8617-32f8eeebf8f2
> From: <sip:88881 at A_IP_ADDRESS>;tag=0a3e31a6-96ad-42d0-9310-81b35cedbd3d
> To: <sip:88290 at KAMAILIO_FQDN>;tag=hvra7mj3q0
> Call-ID: 46f44741-d155-4dd5-8fd8-78e540fc1acb
> CSeq: 18326 ACK
> Route:
> <sip:KAMAILIO_FQDN;lr;r2=on;ftag=0a3e31a6-96ad-42d0-9310-81b35cedbd3d;nat=wss>
> Route:
> <sip:KAMAILIO_FQDN:8089;transport=ws;lr;r2=on;ftag=0a3e31a6-96ad-42d0-9310-81b35cedbd3d;nat=wss>
> Max-Forwards: 70
> User-Agent: Asterisk PBX 13.33.0
> Content-Length:  0
>
>
> By loop I meant, Kamailio just relaying it back to self and discard.
>
> Regards,
> Igor
>
> On 07.05.2021 13:48, Daniel-Constantin Mierla wrote:
>
> Can you paste the ACK that loops, after being handled once by Kamailio?
>
> Cheers,
> Daniel
> On 07.05.21 13:25, Igor Olhovskiy wrote:
>
> Daniel,
>
> Yes, it is.
>
> alias=...
>
> Also tried with
>
> listen = IP advertise FQDN
>
> same behavior, loose_route() stops acting correctly.
>
> PS: Forgot to add, Kamailio 5.4.3 / 5.4.4
>
> Regards,
> Igor
>
> On 07.05.2021 13:21, Daniel-Constantin Mierla wrote:
>
> Hello,
>
> is the KAMAILIO_FQDN set as local domain for Kamailio (via alias parameter
> or domain module+register myself)?
>
> Cheers,
> Daniel
> On 07.05.21 11:17, Igor Olhovskiy wrote:
>
> Hello,
>
> I saw there are some topics on this already and carefully walked through
> all of them, but can't solve following issue.
>
> For a reason I do need that in Record-Route header sent to endpoint would
> present FQDN. If it matters, it's UDP/WSS conversion done on Kamailio.
>
> So, scheme is quite simple
>
> Enpoint A  ->UDP-> Kamailio ->WSS-> Endpoint B (NAT)
>
> Main issue here, that if in Record-Route headers it's FQDN, but not IP
> addresses, on a new transactions with a dialog (ACK on 200, PRACK, BYE),
> Kamailio* loose_route()* function resolves address of destination not
> current dialog, but actual R-URI (or itself, if R-URI is something from
> WebRTC world) that is not correct due to NAT.
>
> If in RR headers IP addresses are present - all is working as expected.
>
> As an example (RR with FQDN)
>
> B answers 200
>
> SIP/2.0 200 OK
> Record-Route:
> <sip:KAMAILIO_FQDN:8089;transport=ws;r2=on;lr=on;ftag=0a3e31a6-96ad-42d0-9310-81b35cedbd3d;nat=wss>
> Record-Route:
> <sip:KAMAILIO_FQDN;r2=on;lr=on;ftag=0a3e31a6-96ad-42d0-9310-81b35cedbd3d;nat=wss>
> Via: SIP/2.0/UDP <A_IP_ADDRESS>:5060;received=A IP
> ADDRESS;rport=5060;branch=z9hG4bKPj67fb6d86-97d7-4231-995b-e54b0f62881e
> To: <sip:88290@<KAMAILIO_FQDN>>;tag=hvra7mj3q0
> From: <sip:+XXXX7688881@
> <KAMAILIO_FQDN>>;tag=0a3e31a6-96ad-42d0-9310-81b35cedbd3d
> Call-ID: 46f44741-d155-4dd5-8fd8-78e540fc1acb
> CSeq: 18326 INVITE
> Contact: <sip:88290 at toleivu2gdbh.invalid;transport=wss>
> Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
> Supported: outbound
> Content-Type: application/sdp
> Content-Length: 817
>
>
> ACK looks like
>
> ACK sip:88290 at toleivu2gdbh.invalid;transport=wss SIP/2.0
> Via: SIP/2.0/UDP
> A_IP_ADDRESS:5060;rport;branch=z9hG4bKPj8d05548a-91ef-4332-8617-32f8eeebf8f2
> From: <sip:88881 at A_IP_ADDRESS>;tag=0a3e31a6-96ad-42d0-9310-81b35cedbd3d
> To: <sip:88290 at KAMAILIO_FQDN>;tag=hvra7mj3q0
> Call-ID: 46f44741-d155-4dd5-8fd8-78e540fc1acb
> CSeq: 18326 ACK
> Route:
> <sip:FQDN;lr;r2=on;ftag=0a3e31a6-96ad-42d0-9310-81b35cedbd3d;nat=wss>
> Route:
> <sip:FQDN:8089;transport=ws;lr;r2=on;ftag=0a3e31a6-96ad-42d0-9310-81b35cedbd3d;nat=wss>
> Max-Forwards: 70
> User-Agent: Asterisk PBX 13.33.0
> Content-Length:  0
>
> And Kamailio on *loose_route()* loops ACK to itself. (with result of
> function == 1)
>
> In a case if in Record-Route/Route headers just IP address of Kamailio
> present, all works as expected, but it's not really behavior I want to
> achieve.
>
> I've tried to play with *record_route_preset("...")* specifying only WSS
> part (as suggested in https://skalatan.de/de/blog/kamailio-sbc-teams)
> with FQDN, but no luck.
>
> Also wanted to try approach using record_route_preset() function in branch
> route, but it was working only with first branch, not affecting others (but
> I assume having different RR headers across branches is not a good idea)
>
> --
> Regards,
> Igor
>
>
> __________________________________________________________
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> --
> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda
> Kamailio Advanced Training - Online
> May 17-20, 2021 (Europe Timezone) - June 7-10, 2021 (America Timezone)
>   * https://www.asipto.com/sw/kamailio-advanced-training-online/
>
>
> __________________________________________________________
> Kamailio - Users Mailing List - Non Commercial Discussions
>   * sr-users at lists.kamailio.org
> Important: keep the mailing list in the recipients, do not reply only to the sender!
> Edit mailing list options or unsubscribe:
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>
> --
> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda
> Kamailio Advanced Training - Online
> May 17-20, 2021 (Europe Timezone) - June 7-10, 2021 (America Timezone)
>   * https://www.asipto.com/sw/kamailio-advanced-training-online/
>
> --
> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda
> Kamailio Advanced Training - Online
> May 17-20, 2021 (Europe Timezone) - June 7-10, 2021 (America Timezone)
>   * https://www.asipto.com/sw/kamailio-advanced-training-online/
>
> --
> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda
> Kamailio Advanced Training - Online
> May 17-20, 2021 (Europe Timezone) - June 7-10, 2021 (America Timezone)
>   * https://www.asipto.com/sw/kamailio-advanced-training-online/
>
> __________________________________________________________
> Kamailio - Users Mailing List - Non Commercial Discussions
>   * sr-users at lists.kamailio.org
> Important: keep the mailing list in the recipients, do not reply only to
> the sender!
> Edit mailing list options or unsubscribe:
>   * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
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