Hi,
I am noticing strange thing. On one of our UAC's we are experiancing
strange behaviour.
UAC - INVITE-OPENSIPS
OPENSIPS-Proxy_Autehentication-UAC
UAC-ACK-Opensips
UAC-INVITE(with credentials)-OPENSIPS
UAC-INVITE(with credentials)-OPENSIPS
OPENSIPS-Proxy_Autehentication-UAC
..
.
.
.
Why is
Hello Craig,
As said, this is just side comment, not a try to cover the problem -
IMHO you are right (in regards to maddr handling in nat_traversal) and
that should be fixed.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 23.05.2014 04:38,
Hi Miha,
It looks like the UAC is doing a retransmission of the INVITE with
credentials - do you have enabled the nonce checking for re-usage (see
http://www.opensips.org/html/docs/modules/1.11.x/auth.html#id293610) ?
If is active, the first INVITE with be accepted by auth and pushed for
Hi Kaan,
If OpenSIPS is the one sending the 405 reply (you can check that by
simply doing a sip capture on the opensips machine), it is from the
script - check what method you have in your test traffic - that method
may be rejected from the script you use.
Regards,
Bogdan-Andrei Iancu
Hi Pradeep,
You can simply use the default script that comes with OpenSIPS - it has
support for registration; if you need something more, check make
menuconfig and get an auto-generated cfg from the Residential scenario.
Check:
http://www.opensips.org/Documentation/Tutorials-GettingStarted
Hello,
Is it possible to send media rtp packets from one rtpproxy server to
another rtpproxy server?
In my scenario , i am registering voip account via opensips proxy server.
We have rtpproxy and opensips server hosted on same place. opensips changes
c= and m= lines of SDP accordingly but when
Hi Bogdan,
Thanks for the quick response.
The method is Register. I am using standard load balancer configuration script
generated by opensips version 1.9 make menuconfig. Which lines I should add for
register method.?
BR,
Kaan
Samsung Mobile tarafından gönderildi
div Orjinal
Hi Bogdan,
tnx you for your quick respons and explenation.
br
miha
Dne 5/23/2014 12:14 PM, piše Bogdan-Andrei Iancu:
Hi Miha,
It looks like the UAC is doing a retransmission of the INVITE with
credentials - do you have enabled the nonce checking for re-usage (see
Kaan,
The LB module is exclusively for calls (nothing else than INVITEs) -
because the load is the actual number of calls (please refer to the
documentation).
If you want to distribute/LB other methods, I recommend you using the
dispatcher module.
Hi Kaushik,
As rtpproxy (by default) waits to first receive traffic (before pushing
traffic back), there is a dead-lock created between the 2 rtpproxies -
both are waiting for the other to send traffic.
What you have to do is, when using rtpproxy, to instruct rtpproxy to
trust (not to wait,
Going for a public exposure on this question to Maxim, maybe we will get
an answer here.
Original Message
Subject:RTPproxy project
Date: Mon, 14 Apr 2014 15:03:31 +0300
From: Bogdan-Andrei Iancu bog...@opensips.org
To: Maxim Sobolev sobo...@sippysoft.com
CC:
hello:
I install opensips-1.8.2 and asterisk together at my vmware. but when I
register the SIP phone or physical
Phone, the clients always show 483 too many Hops. I disable the fireware and
use:
ngrep -d lo -qt -W byline port 5060, the debug info is this:
Via: SIP/2.0/UDP
hello: all of users:
from opensips debug info, there are few bugs with no totag:
:47:17 [2915] DBG:core:parse_via_param: found param type 232, branch =
z9hG4bK-d87543-e049e36a4d0d0d08-1--d87543-; state=6
May 23 22:47:17 [2915] DBG:core:parse_via_param: found param type 235, rport
= 7147;
To be honest, i have stopped using rtpproxy for over 2 years now. It
is not evolving as fast as it should be, specially in the context of ICE
and WebRTC technologies.
I would like to suggest that opensips team
should consider adding support for rtpengine from SIPWise,
Thank you Bogdan for the reply.
I am wondering if I can use the get_dialog__info somehow to find out the
original Invite (A) is sent back.
I am still investigating if my SIP flow will always keep the contract (with the
VIA header solution).
Is there any other modules or functions that I could
Hi,
I'm just starting with OpenSIPs, but this means that the message isn't
being routed correctly. It's falling through the dialplan to a self-route,
which causes it to handle it again, eventually failing as it cuts the hop
count every time. I'm working on resolving the issue myself, but that is
Hey Bogdan,
I'm trying to do the same thing as Pradeep and failing. I've added a user
to the subscribers database, but I feel as though the default configuration
doesn't support registrar behavior. In the standard debian package, this is
the only line seemingly hinging on a REGISTER method:
Getting back on this horse, what do you mean alias setup. There seem to
be two different aliases: one for users (x@y - 1000@foo) and one for the
service itself:
Restarting opensips: opensipsListening on
udp: 192.168.0.0.1 [192.168.0.0.1]:5060
Aliases:
udp:
This totally makes sense! Thanks so much. Due to the intricacies of our
design (ec2 in action) the automatic alias detection isn't working (it gets
the name from the local DNS which isn't routable externally). How do I get
it to know that myself is a certain domain name? I set SIP_DOMAIN in the
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