25 Apr
2023
25 Apr
'23
5:21 a.m.
Probably in the specific language of the rfc for sip, the udp may not be
considered as a reliable one, but it is actually used to build reliable
protocols, like:
- https://en.wikipedia.org/wiki/Secure_Reliable_Transport
With retransmissions, the communication over udp becomes reliable,
otherwise it wouldn't have been so popular. Like Olle said, there are
quite many interpretation that one could do from sips related specs that
is hard to get to some common grounds for "reliable" implementations ...
Then there could be DTLS, that could be considered secure, but not
reliable from the same perspective, ...
Daniel
On 25.04.23 09:32, Sergey Safarov wrote:
some links from RFC
if the Request-URI contains a SIPS URI, TLS MUST be used to
communicate with that proxy.
A SIPS URI specifies that the resource be contacted securely. This
means, in particular, that TLS is to be used between the UAC and the
domain that owns the URI.
For a SIPS URI, the transport parameter MUST indicate a reliable
transport.
https://datatracker.ietf.org/doc/html/rfc3261
I think clean UDP cannot be used.
On Tue, Apr 25, 2023 at 10:08 AM Olle E. Johansson <oej(a)edvina.net> wrote:
Agree, the SIPS: URL was probably a good idea at the time of
writing the SIP RFC (20 years ago) since other protocols had
secure variants, like HTTPS, LDAPS etc.
But the specs wasn’t very well considered and is today generally
thought of as a bad idea. There has been a few attempts to fix it,
but nothing that got implemented by a large amount of implementations.
As an example: If your device registers with a SIPS: contact it
has to have a server cert and accept incoming TLS connections from
the server. This will not work if the phone is behind NAT.
Better to use the SIP: URI and set transport to TLS.
/O
--
Daniel-Constantin Mierla -- www.asipto.com
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com
On 24 Apr 2023, at 16:22, Daniel-Constantin
Mierla
<miconda(a)gmail.com> wrote:
The sips scheme is misleading because people expect to be SIP
over TLS, but it is not, it is SIP over secure network, which can
be a private network or a vpn. So the sips can meet the
requirements even for sip over udp.
But if you say that the call get's connected, only that is no
audio and ends quickly, likely the issue is with the RTP layer,
when the sips endpoint expect srtp and the other endpoint does
not do it.
Probably you have to share the ngrep output or pcap with all sip
messages of such call.
Cheers,
Daniel
On 24.04.23 16:14, Kiss Zoltán wrote:
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Hi,
We have to test every scenario, but the latest issue was we have
one way rtp and the call is dropped after 6 seconds cc.
In the test the calle was the GS phone which is registered via
Kamailio, and the called party was an another phone witch was
registered directly tot he backend Asterisk.
After switching GrandStream phone to sip scheme, then everything
is working fine again.
Zoltan
*From:* Daniel-Constantin Mierla <miconda(a)gmail.com>
<mailto:miconda@gmail.com>
*Sent:* Monday, April 24, 2023 4:11 PM
*To:* Kamailio (SER) - Users Mailing
List <sr-users(a)lists.kamailio.org>
<mailto:sr-users@lists.kamailio.org>; Kiss
Zoltán <kiss.zoltan(a)adertis.hu> <mailto:kiss.zoltan@adertis.hu>
*Subject:* Re: [SR-Users] sips to sip with TLS proxy
Hello,
just to clarify: you cannot initiate calls from the phone or you
can't sent calls to the phone?
Cheers,
Daniel
On 24.04.23 15:58, Kiss Zoltán wrote:
Hi all,
We have a working Kamailio setup, lets call it a transparent
proxy for Asterisk boxes. Its based on domain and dispatcher
modules and everything is working as expected with the test
clients (more or less microsip, softphone for ios, etc). We
are tried to register with a Grandstream deskphone today,
and we see that the phone sending sips:xxx in the Reg
Contact field for example. Because the sips schema, the
register is working, but we cannot initiate calls from this
phone. If we are turning SIP scheme to sip from sips in the
phone, then everything is working as expected.
I think we can transform those requests from sips to sip
with Kamailio, but currently we dont know where can we start.
Has anybody a suggestion about this issue? I know that we
can transform ruri, contact, etc with textops, nathelper and
a lot of other modules, but what is the best for this
sips->sip translation?
Thanks for your help.
With kind regards,
Zoltan
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--
Daniel-Constantin Mierla -- www.asipto.com <http://www.asipto.com/>
www.twitter.com/miconda <http://www.twitter.com/miconda> --
www.linkedin.com/in/miconda <http://www.linkedin.com/in/miconda>
Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com
<http://www.kamailioworld.com/>
--
Daniel-Constantin Mierla -- www.asipto.com <http://www.asipto.com/>
www.twitter.com/miconda <http://www.twitter.com/miconda> --
www.linkedin.com/in/miconda <http://www.linkedin.com/in/miconda>
Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com
<http://www.kamailioworld.com/>
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