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.
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
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:
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> <miconda(a)gmail.com>
*Sent:* Monday, April 24, 2023 4:11 PM
*To:* Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>
<sr-users(a)lists.kamailio.org>rg>; Kiss Zoltán <kiss.zoltan(a)adertis.hu>
<kiss.zoltan(a)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 --
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www.kamailioworld.com
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Daniel-Constantin Mierla --
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www.kamailioworld.com
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