24 Mar
2020
24 Mar
'20
10:24 p.m.
On 24.03.20 18:11, Ben Kaufman wrote:
I may be misinterpreting this, but RFC3261 section
18.2.2 ( Sending Responses ):
- The server transport uses the value of the top Via header field in
- order to determine where to send a response. It MUST follow the
- following process:
-
- o If the "sent-protocol" is a reliable transport protocol such as
- TCP or SCTP, or TLS over those, the response MUST be sent using
- the existing connection to the source of the original request
- that created the transaction, if that connection is still open.
- This requires the server transport to maintain an association
- between server transactions and transport connections. If that
- connection is no longer open, the server SHOULD open a
- connection to the IP address in the "received" parameter, if
- present, using the port in the "sent-by" value, or the default
- port for that transport, if no port is specified. If that
- connection attempt fails, the server SHOULD use the procedures
- in [4] for servers in order to determine the IP address and
- port to open the connection and send the response to.
The message is constructed by the application, and the TCP source port is determined
afterwards, so there's no way that the application can know what that port will be
ahead of time.
I do not get why the local port on client system (or the source port of
the tcp connection from server point of view) cannot be figured out
before constructing the SIP message. As I said in the previous email,
the app can connect, learn the local port then construct the message and
then send it over the connection. The port is allocated at connect time,
not when sending first data over the connection.
The telephone app on macos does it, here is the relevant part of a
register generated by it:
2020/03/24 22:10:46.857098 192.168.188.20:61851 -> asipto.lab:5060
REGISTER sip:asipto.lab SIP/2.0
Via: SIP/2.0/TCP
192.168.188.20:61851;rport;branch=z9hG4bKPjZO5u8nv3.APD7kvptujo-1wMF0KCFfxw;alias
Route: <sip:asipto.lab;transport=tcp;lr>
Max-Forwards: 70
From: "dcmtest00"
<sip:dcmtest00@asipto.lab>;tag=Lsnew1BlAcXdVJ9C9JZK57hKAvS6X11s
To: "dcmtest00" <sip:dcmtest00@asipto.lab>
Call-ID: 8HyZKCb1xJvY8Mh9wI21-w8TXBQbBKsX
CSeq: 40327 REGISTER
User-Agent: Telephone 1.4
Supported: outbound, path
Contact: "dcmtest00"
<sip:dcmtest00@192.168.188.20:61851;transport=TCP;ob>;reg-id=1;+sip.instance="<urn:uuid:00000000-0000-0000-0000-0000871a7549>"
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
Content-Length: 0
The port in the SIP message is the source port of the IP packet.
When using TCP, it should be expected that message
source port will be different than the port indicated in the Via.
From my experience, in most of the cases it is the
same, unless it is a
natted device using or not stun.
If port would be impossible to be known, then it would be no sense to
just have a value there. In WebRTC/websocket, when the JS app cannot
know the local IP, the specs say it can be a random string.
With that said, there's nothing wrong here. nat_uac_test(flags) bit value 16 is :
* 16 - Test if the source port is different from the port in the "Via"
header
This tests exactly what it states, so it's not a false positive. It's probably
a bad idea to use bit value 16 to check for NAT if using TCP.
Kamailio has options to create aliases for tcp connections, as well as
don't connect for forwarding, if the connection is expected to be open
(or is known it cannot be created by the server when device is behind
nat). Otherwise, from my point of view, I don't see it a bad idea to use
this test to detect that a device is behind nat even for tcp/tls.
Cheers,
Daniel
Ben Kaufman
ben.kaufman(a)altigen.com
-----Original Message-----
From: Daniel-Constantin Mierla <miconda(a)gmail.com>
Sent: Tuesday, March 24, 2020 10:47 AM
To: Ben Kaufman <ben.kaufman(a)altigen.com>om>; Kamailio (SER) - Users Mailing List
<sr-users(a)lists.kamailio.org>rg>; Juha Heinanen <jh(a)tutpro.com>
Subject: Re: [SR-Users] about kamailio.cfg NAT test
Yes, tcp is typically using ephemeral source port, in the way that is allocated by the
tcp stack/kernel, not easy to be specified by the client app. But the client app can get
the local (ephemeral) port after
connect() using getsockname() and use it to build the SIP message that is going to be
sent through the connection. I expect this is the way how (most of the) sip phone apps do
it when the port in SIP headers is matching the source port of the connection.
Cheers,
Daniel
On 24.03.20 15:15, Ben Kaufman wrote:
--
Daniel-Constantin Mierla -- www.asipto.com
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Shouldn't *TCP* use an ephemeral source port,
and expect replies to be to the source port?
Ben Kaufman
-----Original Message-----
From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of
Daniel-Constantin Mierla
Sent: Tuesday, March 24, 2020 9:01 AM
To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>rg>;
Juha Heinanen <jh(a)tutpro.com>
Subject: Re: [SR-Users] about kamailio.cfg NAT test
Hello,
this is the test to detect devices behind NAT that use STUN, so they discover properly
the public IP of the NAT router, but the port allocation is different for STUN and SIP
traffic.
If you have an asymmetric signalling client, then this test is not useful -- actually nat
traversal cannot be done for asymmetric signalling (when client sends from one address
(ip+port) and expects traffic back to another address). You haven't included the
contact address, but anyhow probably the reply is not expected on port 44717 as it is
indicated in Via. The phone expects some natted environment, as it requested
"rport" handling (send back the reply to the port from where the request was
received). If it wanted asymmetric signalling, then rport is not needed. If it knew it
sends and receives to the same port, also rport would not be needed.
Maybe this is just baresip approach not to detect local port of the tcp connection,
putting a random value in Via and setting rport parameter.
You can tune the test for your specific environment/client apps, but for the vast
majority of the cases I encountered over the years, this test is the best one to detect
clients behind nat that are using stun.
Cheers,
Daniel
On 24.03.20 13:01, Juha Heinanen wrote:
--
Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda --
www.linkedin.com/in/miconda
STATEMENT OF CONFIDENTIALITY:
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or copying is strictly prohibited. Please notify AltiGen Communications
immediately at either (888)258-4436 or via email to
administrator(a)altigen.com, and destroy all copies of this message and any
attachments. In kamailio/etc/kamailio.cfg NAT test is based on
nat_uac_test("19").
19 includes test 16:
16 - Test if the source port is different from the port in the “Via”
header. If the “Via” header contains no port, it uses the default SIP
port 5060
Based on a couple of tests using baresip, looks like that results in
false positive when UA connects to SIP proxy via TCP.
An example:
T 2020/03/24 13:57:12.685441 192.26.134.10:38940 -> 192.26.134.1:5060
[AP] #127 REGISTER sip:test.tutpro.com SIP/2.0.
Via: SIP/2.0/TCP 192.26.134.10:44717;branch=z9hG4bK5b6aa423104ef942;rport.
Does someone know why test 16 is included?
-- Juha
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--
Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda --
www.linkedin.com/in/miconda
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If you are not the intended recipient, or the person responsible for
delivering the e-mail to the intended recipient, be advised you have
received this message in error and that any use, dissemination,
forwarding, printing, or copying is strictly prohibited. Please notify
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