David,

Yes, I did added it, means it was there, but is_first_hop() was blocking adding it. I think it's some leftovers from default config.

So, my conclusion, that is_first_hop() is ok with IP addresses, but not ok with FQDN in route. Although FQDN is added as alias

Regards,
Igor
On 07.05.2021 16:07, David Villasmil wrote:
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@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@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@A_IP_ADDRESS>;tag=0a3e31a6-96ad-42d0-9310-81b35cedbd3d
To: <sip:88290@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@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@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@A_IP_ADDRESS>;tag=0a3e31a6-96ad-42d0-9310-81b35cedbd3d
To: <sip:88290@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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www.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.com
www.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.com
www.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/
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Important: keep the mailing list in the recipients, do not reply only to the sender!
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Regards,

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

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