24 Apr
2023
24 Apr
'23
10:22 a.m.
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>
*Sent:* Monday, April 24, 2023 4:11 PM
*To:* Kamailio (SER) - Users Mailing List
<sr-users(a)lists.kamailio.org>rg>; Kiss Zoltán <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 -- www.asipto.com
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com