[SR-Users] loose_route() and FQDN

Fri May 7 13:43:40 CEST 2021

Just made another test with TLS, not WSS. Same behavior

A -> Kamailio -> B

INVITE sip:88882 at 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 at A_IP_ADDRESS>;tag=62ce37c8-b186-40d9-ae88-31321338031d
To: <sip:88882 at KAMAILIO_FQDN>
Contact: <sip:asterisk at 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 at tone.cern.ch>;tag=62ce37c8-b186-40d9-ae88-31321338031d
To: "TEST 88882" <sip:88882 at KAMAILIO_FQDN>;tag=5AE57984-651CED77
CSeq: 597 INVITE
Call-ID: 8c9f68bf-1514-4dfa-8ba0-72cd8922a22b
Contact: <sip:88882 at 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 at 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 at A_IP_ADDRESS>;tag=62ce37c8-b186-40d9-ae88-31321338031d
To: <sip:88882 at 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 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.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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