15 Feb
2023
15 Feb
'23
8:53 p.m.
Hi Federico,
Thank you so much for your answer and help.
I know this version is old but it is difficult to change at this time
as there are some challenges. I will definitely do the upgrade in the next
few months.
Now regarding my issue. It might be silly but, my question is, the
NATDETECT route that is on top of request_route is not sufficient to add
the alias with the set_contact_alias that the below code has? Why is it not
added on the re-invite as it is added on the initial invite?
route[NATDETECT] {
if (nat_uac_test("19")) {
if (is_method("REGISTER")) {
fix_nated_register();
} else {
if(is_first_hop()) {
set_contact_alias();
}
}
setflag(FLT_NATS);
}
return;
}
Regards,
Angelo
Στις Τετ 15 Φεβ 2023 στις 8:54 μ.μ., ο/η Federico Cabiddu <
federico.cabiddu(a)gmail.com> έγραψε:
Hi,
your Kamailio version is very old (4.3) and unsupported, so my first
advice would be to upgrade to a recent version (5.6.3 is the latest stable).
Anyhow, your problem is not the kamailio version you are using but the
fact that the Contact header URI of the re-INVITE 200 OK only contains the
private port and not the public one, thus the R-URI in the ACK only
contains the private one and Kamailio cannot do anything else than
forwarding the ACK there. The 200 OK Contact URI relied to the re-INVITE's
sender should contain an 'alias' parameter (like in the re-INVITE R-URI)
with the public IP/port. You can add it using natheper's add_contact_alias
function (module
https://www.kamailio.org/docs/modules/5.6.x/modules/nathelper.html#nathelpe…):
be sure that you call it in the reply route that processes the 200 OK. Then
upon receiving the ACK use handle_ruti_alias function from the same module
to forward it to the public/ip/port.
Hope this helps,
Federico
On Wed, Feb 15, 2023 at 7:06 PM Angelo Sipper <sippro97(a)gmail.com> wrote:
None?
Στις Σάβ 11 Φεβ 2023 στις 5:55 π.μ., ο/η Angelo Sipper <
sippro97(a)gmail.com> έγραψε:
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Any sugestions?
Στις Παρ 27 Ιαν 2023 στις 9:04 μ.μ., ο/η Angelo Sipper <
sippro97(a)gmail.com> έγραψε:
> Hi,
>
> Any help on this topic?
>
> Regards,
> Angelo
>
> Στις Πέμ 26 Ιαν 2023 στις 9:33 μ.μ., ο/η Angelo Sipper <
> sippro97(a)gmail.com> έγραψε:
>
>> Hi,
>>
>> My setup has: (TLS)UAC 77.1.1.1 -> (TLS)Kamailio 17.17.17.211 (UDP) ->
>> (UDP)Kamailio 17.17.17.181 (UDP) -> Carrier(UDP)
>> Both Kamailio are 4.3 with minor modifications on default cfg, and
>> Topoh and TLS modules are used.
>>
>> I am having an issue on some outbound calls from nated UACs. On these
>> calls our SBC sometimes receives reverse invites from the carrier side
>> within the call. When this happens, my SBC(17.17.17.211) sends the invite
>> to UAC and the UAC sends back 200 OK (SDP). Unfortunately my
>> SBC(17.17.17.211) sends back the ACK not to the correct UAC public port but
>> instead to the UAC internal port making the call to crash with several
>> retries from my SBC to send 200 OK as ACK was never received from UAC!
>>
>> If you look the below traces of reverse invite past within the call,
>> the reinvite has port private 20125 and alias the public port 2314 of UAC
>> but the ACK bellow is send to 20125 and not to 2314
>>
>> 2023/01/26 09:48:30.668634 17.17.17.181:5060 -> 17.17.17.211:5060
>> INVITE sip:301234567890@77.1.1.1:20125;alias=77.1.1.1~2314~1 SIP/2.0
>> Via: SIP/2.0/UDP
>> 17.17.17.181;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0
>> From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9
>> To: "301234567890"
<sip:301234567890@sip.mydomain.com>;tag=07ece97c
>> Call-ID: 1q5nE37RGZ8a8ChI6rqShw..
>> CSeq: 29227 INVITE
>> Max-Forwards: 69
>> Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS
>> Accept: application/sdp, application/isup, application/dtmf,
>> application/dtmf-relay, multipart/mixed
>> Supported: timer
>> Session-Expires: 600;refresher=uas
>> Min-SE: 90
>> Content-Length: 297
>> Content-Disposition: session; handling=required
>> Content-Type: application/sdp
>> Route:
>>
<sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes>
>> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181>
>>
>> v=0
>> o=Sonus_UAC 14218 13519 IN IP4 14.1.1.30
>> s=SIP Media Capabilities
>> c=IN IP4 14.1.1.30
>> t=0 0
>> m=audio 18600 RTP/AVP 8 18 101
>> a=maxptime:20
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:18 G729/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:18 annexb=no
>> a=fmtp:101 0-15
>> a=sendrecv
>> a=rtcp:18601
>>
>>
>> 2023/01/26 09:48:30.679743 17.17.17.211:5060 -> 17.17.17.181:5060
>> SIP/2.0 100 trying -- your call is important to us
>> Via: SIP/2.0/UDP
>> 17.17.17.181;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0;rport=5060
>> From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9
>> To: "301234567890"
<sip:301234567890@sip.mydomain.com>;tag=07ece97c
>> Call-ID: 1q5nE37RGZ8a8ChI6rqShw..
>> CSeq: 29227 INVITE
>> Server: Supervoice
>> Content-Length: 0
>>
>>
>>
>> 2023/01/26 09:48:30.740803 17.17.17.211:5060 -> 77.1.1.1:2314
>> INVITE sip:301234567890@77.1.1.1:20125 SIP/2.0
>> Via: SIP/2.0/UDP
>> 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0
>> Via: SIP/2.0/UDP
>> 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0
>> From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9
>> To: "301234567890" <sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c
>> Call-ID: 1q5nE37RGZ8a8ChI6rqShw..
>> CSeq: 29227 INVITE
>> Max-Forwards: 68
>> Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS
>> Accept: application/sdp, application/isup, application/dtmf,
>> application/dtmf-relay, multipart/mixed
>> Supported: timer
>> Session-Expires: 600;refresher=uas
>> Min-SE: 90
>> Content-Length: 297
>> Content-Disposition: session; handling=required
>> Content-Type: application/sdp
>> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181>
>>
>> v=0
>> o=Sonus_UAC 14218 13519 IN IP4 14.1.1.28
>> s=SIP Media Capabilities
>> c=IN IP4 14.1.1.28
>> t=0 0
>> m=audio 13288 RTP/AVP 8 18 101
>> a=maxptime:20
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:18 G729/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:18 annexb=no
>> a=fmtp:101 0-15
>> a=sendrecv
>> a=rtcp:13289
>>
>>
>> 2023/01/26 09:48:30.893512 77.1.1.1:2314 -> 17.17.17.211:5060
>> SIP/2.0 100 Trying
>> Via: SIP/2.0/UDP
>> 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0
>> Via: SIP/2.0/UDP
>> 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0
>> To: "301234567890" <sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c
>> From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9
>> Call-ID: 1q5nE37RGZ8a8ChI6rqShw..
>> CSeq: 29227 INVITE
>> Content-Length: 0
>>
>>
>>
>> 2023/01/26 09:48:31.026742 77.1.1.1:2314 -> 17.17.17.211:5060
>> SIP/2.0 200 OK
>> Via: SIP/2.0/UDP
>> 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0
>> Via: SIP/2.0/UDP
>> 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0
>> Require: timer
>> Contact: <sip:301234567890@77.1.1.1:20125>
>> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c
>> From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9
>> Call-ID: 1q5nE37RGZ8a8ChI6rqShw..
>> CSeq: 29227 INVITE
>> Session-Expires: 600;refresher=uas
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY,
>> REFER, INFO, MESSAGE, UPDATE
>> Content-Type: application/sdp
>> Supported: replaces, timer
>> User-Agent: 3CXPhoneSystem 18.0.6.905 (905)
>> Content-Length: 259
>>
>> v=0
>> o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1
>> s=3cxPS Audio call
>> c=IN IP4 77.1.1.1
>> t=0 0
>> m=audio 9054 RTP/AVP 8 18 101
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:18 G729/8000
>> a=fmtp:18 annexb=no
>> a=rtpmap:101 telephone-event/8000
>> a=sendrecv
>>
>>
>> 2023/01/26 09:48:31.093656 17.17.17.211:5060 -> 17.17.17.181:5060
>> SIP/2.0 200 OK
>> Via: SIP/2.0/UDP
>> 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0
>> Require: timer
>> Contact: <sip:301234567890@77.1.1.1:20125>
>> To: "301234567890"
<sip:301234567890@sip.mydomain.com>;tag=07ece97c
>> From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9
>> Call-ID: 1q5nE37RGZ8a8ChI6rqShw..
>> CSeq: 29227 INVITE
>> Session-Expires: 600;refresher=uas
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY,
>> REFER, INFO, MESSAGE, UPDATE
>> Content-Type: application/sdp
>> Supported: replaces, timer
>> User-Agent: 3CXPhoneSystem 18.0.6.905 (905)
>> Content-Length: 276
>>
>> v=0
>> o=3cxPS 792019937001472 2393157908037634 IN IP4 14.1.1.28
>> s=3cxPS Audio call
>> c=IN IP4 14.1.1.28
>> t=0 0
>> m=audio 13278 RTP/AVP 8 18 101
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:18 G729/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:18 annexb=no
>> a=sendrecv
>> a=rtcp:13279
>>
>>
>> 2023/01/26 09:48:31.254591 17.17.17.181:5060 -> 17.17.17.211:5060
>> ACK sip:301234567890@77.1.1.1:20125 SIP/2.0
>> Via: SIP/2.0/UDP
>> 17.17.17.181;branch=z9hG4bK7b5d.48f49235d45a0f6f9efdad9d6d8224bd.0
>> From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9
>> To: "301234567890"
<sip:301234567890@sip.mydomain.com>;tag=07ece97c
>> Call-ID: 1q5nE37RGZ8a8ChI6rqShw..
>> CSeq: 29227 ACK
>> Max-Forwards: 69
>> Content-Length: 0
>> Route:
>>
<sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes>
>> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181>
>>
>>
>>
>> 2023/01/26 09:48:31.260901 17.17.17.211:5060 -> 77.1.1.1:20125
>> ACK sip:301234567890@77.1.1.1:20125 SIP/2.0
>> Via: SIP/2.0/UDP
>> 17.17.17.211;branch=z9hG4bK7b5d.e8fb34e79623843d516ae77257014e2d.0
>> Via: SIP/2.0/UDP
>> 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.48f49235d45a0f6f9efdad9d6d8224bd.0
>> From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9
>> To: "301234567890" <sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c
>> Call-ID: 1q5nE37RGZ8a8ChI6rqShw..
>> CSeq: 29227 ACK
>> Max-Forwards: 68
>> Content-Length: 0
>> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181>
>>
>> Any help will be much appreciated.
>>
>> Thank you!
>> Regards,
>> Angelo
>>
> __________________________________________________________
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