Hi,
So, the problem is occuring when a device, which are using sips schema initiating a call. If we calling the device from another device which are using „plain” sip, then its working.
The RTP is okay, even if the call dropping after 6 sec.
I think the problem can be spotted in the last BYE from backend Asterisk to Kamailio:
2023/04/25 13:01:35.134587 10.0.5.6:5060 -> 172.16.10.211:5060
BYE sips:292@10.0.5.192:5062;transport=tls;alias=86.101.146.235~57217~3 SIP/2.0
Via: SIP/2.0/UDP 10.0.5.6:5060;branch=z9hG4bK1168e2fd;rport
Route: <sip:172.16.10.211;r2=on;lr=on;ftag=2063230821;nat=yes>,<sip:79.172.213.175:5061;transport=tls;r2=on;lr=on;ftag=2063230821;nat=yes>
Max-Forwards: 70
From: <sips:291@teszt:5061>;tag=as4a869947
To: "292" <sips:292@teszt:5061>;tag=2063230821
Call-ID: 66383220-5062-7@BA.A.F.BJC
CSeq: 102 BYE
User-Agent: Asterisk PBX certified/18.9-cert4
Proxy-Authorization: Digest username="292", realm="asterisk", algorithm=MD5, uri="sip:teszt", nonce="xxxXXX", response=" xxxXXX "
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0
As you can see the from is also sips which is interesting, because the 291 is using plain „sip”, so i think Asterisk handles this as its specified in the RFC, but we dont want to use sips on the
backend communication.
The route for the packets is simple:
10.0.5.6 (PBX) -> 172.16.10.211 (Kamailio 5060/udp), 79.172.213.175 (Kamailio 5061 TCP/TLS) -> client.
So thats why i think we have to rewrite the schema itself for the backend asterisk boxes from sips to sip.
What do you think?
Cheers,
Zoltan
From: Sergey Safarov <s.safarov@gmail.com>
Sent: Tuesday, April 25, 2023 9:32 AM
To: Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.org>
Cc: Daniel-Constantin Mierla <miconda@gmail.com>; Kiss Zoltán <kiss.zoltan@adertis.hu>
Subject: Re: [SR-Users] Re: sips to sip with TLS proxy
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
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:
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.
just to clarify: you cannot initiate calls from the phone or you can't sent calls to the phone?
On 24.04.23 15:58, Kiss Zoltán wrote:
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?
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Daniel-Constantin Mierla -- www.asipto.com
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Daniel-Constantin Mierla -- www.asipto.com
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