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(a)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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Kamailio Advanced Training - Online
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Kamailio Advanced Training - Online
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Regards,
David Villasmil
email: david.villasmil.work(a)gmail.com
phone: +34669448337
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