7 May
2021
7 May
'21
5:07 p.m.
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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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/
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Important: keep the mailing list in the recipients, do not reply only to
the sender!
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email: david.villasmil.work(a)gmail.com
phone: +34669448337