Just made another test with TLS, not WSS. Same behavior
A -> Kamailio -> B
INVITE sip:88882@KAMAILIO_FQDN:5060 SIP/2.0
Via: SIP/2.0/UDP
A_IP_ADDRESS:5060;rport;branch=z9hG4bKPj84577fd7-611e-40ce-b095-835583310eab
From: <sip:88881@A_IP_ADDRESS>;tag=62ce37c8-b186-40d9-ae88-31321338031d
To: <sip:88882@KAMAILIO_FQDN>
Contact: <sip:asterisk@A_IP_ADDRESS:5060>
Call-ID: 8c9f68bf-1514-4dfa-8ba0-72cd8922a22b
CSeq: 597 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL,
UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 13.33.0
Content-Type: application/sdp
Content-Length: 387
180 reply
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
A_IP_ADDRESS:5060;received=A_IP_ADDRESS;rport=5060;branch=z9hG4bKPj84577fd7-611e-40ce-b095-835583310eab
From:
<sip:+XXXXX688881@tone.cern.ch>;tag=62ce37c8-b186-40d9-ae88-31321338031d
To: "TEST 88882" <sip:88882@KAMAILIO_FQDN>;tag=5AE57984-651CED77
CSeq: 597 INVITE
Call-ID: 8c9f68bf-1514-4dfa-8ba0-72cd8922a22b
Contact: <sip:88882@192.168.0.30:54224;transport=tls>
Record-Route:
<sip:KAMAILIO_FQDN:5061;transport=tls;r2=on;lr=on;ftag=62ce37c8-b186-40d9-ae88-31321338031d;nat=tls>,
<sip:KAMAILIO_FQDN;r2=on;lr=on;ftag=62ce37c8-b186-40d9-ae88-31321338031d;nat=tls>
User-Agent: PolycomVVX-VVX_411-UA/6.3.1.11465
Allow-Events: conference,talk,hold
Accept-Language: en
Require: 100rel
RSeq: 8193
Content-Length: 0
And PRACK from A
PRACK sip:88882@192.168.0.30:54224;transport=tls SIP/2.0
Via: SIP/2.0/UDP
A_IP_ADDRESS:5060;rport;branch=z9hG4bKPjc9d866c0-a483-404e-9b97-d0738490fd74
From: <sip:88881@A_IP_ADDRESS>;tag=62ce37c8-b186-40d9-ae88-31321338031d
To: <sip:88882@KAMAILIO_FQDN>;tag=5AE57984-651CED77
Call-ID: 8c9f68bf-1514-4dfa-8ba0-72cd8922a22b
CSeq: 598 PRACK
Route:
<sip:KAMAILIO_FQDN;lr;r2=on;ftag=62ce37c8-b186-40d9-ae88-31321338031d;nat=tls>
Route:
<sip:KAMAILIO_FQDN:5061;transport=tls;lr;r2=on;ftag=62ce37c8-b186-40d9-ae88-31321338031d;nat=tls>
RAck: 8193 597 INVITE
Max-Forwards: 70
User-Agent: Asterisk PBX 13.33.0
Content-Length: 0
on *loose_route() *Kamailio is trying to send it to 192.168.0.30, which
is just Contact address in 180 responce (incorrect one).
If using IP addresses in RR - all is working as expected
Maybe I'm missing some part of RFC?
Regards,
Igor
On 07.05.2021 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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> --
> Daniel-Constantin Mierla --www.asipto.com
>
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