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
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@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@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.ZoltanFrom: Daniel-Constantin Mierla <miconda@gmail.com>
Sent: Monday, April 24, 2023 4:11 PM
To: Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.org>; Kiss Zoltán <kiss.zoltan@adertis.hu>
Subject: Re: [SR-Users] sips to sip with TLS proxyHello,just to clarify: you cannot initiate calls from the phone or you can't sent calls to the phone?Cheers,
DanielOn 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.comwww.twitter.com/miconda -- www.linkedin.com/in/micondaKamailio World Conference - June 5-7, 2023 - www.kamailioworld.com-- 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__________________________________________________________
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