25 Apr
2023
25 Apr
'23
2:43 a.m.
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> <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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<http://www.asipto.com/>
www.twitter.com/miconda <http://www.twitter.com/miconda> --
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Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com
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