Displaying 20 results from an estimated 20000 matches similar to: "Does Asterisk support STUN or TURN? How to configure asterisk NAT traversal?"
2004 Oct 05
0
SIP and symmetric NAT
Hello,
I have a problem with a Grandstream being behind a symmetric nat. The
box which does the nat is a german "Fritz Box". This one does nat for
the internal network. In the internal network is a Granstream
BudgeTone 100. The nat router has a dial-up connection, so ip changes
on every dial-in.
|------------| |------------| |--------|
|Grandstream
2004 May 22
1
Re: Sipura and STUN (was: rejected NOTIFY requests)
At 7:36 PM +0200 on 5/22/04, Olle E. Johansson wrote:
>>[snip]
>Sending NOTIFY to Asterisk is an error, but a workaround. Since Asterisk
>can handled the NAT traversal all by itself with Qualify (as John points
>out) disabling the NOTIFY will not change anything.
>
>The NOTIFY will in no way affect the status - unreachable/reachable.
>
>Another problem with the SIPURA is
2004 May 22
1
Re: Sipura and STUN (was: rejected NOTIFY re quests)
Sipura does include STUN as an option. It has for quite some time. We are
using it with all of our Sipuras behind NAT'd gateways and it works great!
Try upgrading to the latest Sipura firmware rev.
Darren Nay
> -----Original Message-----
> From: John Todd [mailto:jtodd@loligo.com]
> Sent: Saturday, May 22, 2004 1:57 PM
> To: asterisk-users@lists.digium.com
> Subject:
2005 Mar 08
3
NAT Far End Traversal
Hi List,
After some research, it seems the only reasonable thing to do in order
to get SIP phones behind NAT working reasonably well without fiddling
with the DSL router is to have some kind of far end nat traversal mechanism.
Is there any way to do this with open source tools? I've seen somewhere
that far end nat traversal can be achieved with SER + nathelper does the
job... has anybody
2006 Apr 13
2
NAT/STUN Server
Hi,
I am trying to register SIP clients which are behind NAT on different
network. In order to achieve this goal I think I need STUN Server . I
downloaded STUN Server from
http://internap.dl.sourceforge.net/sourceforge/stun/stund_0.96_Aug13.tgz
But I don't know how to install/configure it.
And please advice me that STUN server is good idea for this scenario?
Thanks in advance
Wazb
2004 Jul 15
1
*, NAT & STUN
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?
Hi friends
I have some doubt in connecting my firefly3rd party softphone from windows machine to asterisk server in linux .
My asterisk is behind the "Port Restricted NAT". I am using STUN server to cross the
2006 Dec 23
1
SNOM 200 behind NAT and other xmas woes
I decided to give the whole family IP phones for christmas,
all hooked into my asterisk server, so all the nephews can
have their own lines.
However, one of the phones I got was the SNOM 200. That's worked
fine for me on my own network, but I'm having bad luck getting
it to work behind NAT talking to Asterisk. It talks to my
termination/origination provider, which seems to ruthlessly
2005 Jul 05
0
Re: [Serusers] NAT considerations...
You will also need your SIP clients that are behind the same NAT to
support ICE (Interactive Connectivty Establishment) if you want calls
between them. Xten Eyebeam and Snom phones are the only ones I'm
aware of that support it.
On 7/5/05, Ricardo Martinez <rmartinez@redvoiss.net> wrote:
> And even worst.
> There are some kind of NAT that STUN does not work.
> You can check
2007 Apr 27
1
How to configure a stun server for a sip peer
HI all!
I'm looking for some infos to configure stun server support for a SIP peer.
I've installed Asterisk 1.4.3, but searching for stun support in
chan_sip (sip.conf) i've found nothing, only a "misterious" externip = stun...
But where i have to put the ip of stun server?
No infos around Google and forum! :-)
Thank all, regards
--
Marco Ciacci
Asterisk Admin
Windows
2005 Mar 28
1
Asterisk, SER, NAT, STUN and the whole debate
Guys.
Im reading a lot about ser, nat, stun, etc. And I noticed there are a lot of
ways to get around nat but I would like to hear some success stories about
handling nat users with multiple voip phones behind nat.
I have my asterisk box behind but ports are forwarded (5060 5004 10000-20000
for rtp and 4569 for iax2) but still.. I can quite figure out what ser and
stund have to do on this
2006 Dec 19
0
Cisco devices (without STUN) and dynamic NAT
Cisco devices (7912, ata-168, 7960 etc.) don't support STUN.
However, they do let you define a static external NAT IP
address, and parameters to send a keep-alive out through the
NAT on a regular basis.
However, I want to make these devices work in an environment
where they are behind a NAT which has a dynamic IP which
might change (though in fact it changes only rarely.)
They're
2007 Oct 22
2
NAT traversal packet loss measurement
How can one measure the effect of NAT traversal packet loss?
We currently have no solution for NAT traversal for our SIP clients. There
is no doubt that packets are getting lost. What is not clear is how much
damage this does. On the face of it, everything seems fine. Could this be
so? Perhaps we're suffering a degradation in quality or our call setup times
could be improved. How can we
2010 Jul 30
1
Please test: STUN patch for Asterisk behind NAT
Hi there!
David has put up a patch to fix the STUN issues that has plagued Asterisk
1.6 ever since that feature was introduced. Now we need testers to verify
the patch, so please grab the patch (or check out the SVN branch) and add
your comments:
https://issues.asterisk.org/view.php?id=17622
Thanks, Philipp
2003 Sep 02
2
STUN server from Vovida
Not sure if it's alright to talk about this here???
compiled the STUN server from Vovida on RedHat 7.3. Looks simple to
configure. It isn't starting...it tries to for a long time and then just
craps out. Here is my config:/etc/sysconfig/stund
#!/bin/echo Not to execute.
# Path to stund
STUND=/usr/sbin/stund
# Set the required args for STUND
STUNDPRIMARYHOSTNAME=208.x.x.x
# The hostname
2003 Jan 06
3
ipsec nat-traversal
It seems to me that ipsecnat tunnel type is not complete.
Latest drafts of ipsec nat-traversal use udp port 4500 for nat-traversal
communications. (It''s called port floating). That is needed to get rid
of ugly ipsec passthru devices.
Now ipsecnat opens port udp/500 from any source port.
And I think ipsecnat won''t work at all with gw zone defined? I''m not
sure about
2016 Mar 07
4
Differences between Chan_SIP and PJSIP with NAT and STUN
> Joshua Colp wrote:
>
> There should be nothing different, except for how you configure things.
> What is the full PJSIP configuration? What is the environment where
> Asterisk is running? Is ICE actually in use on the other side? What is
> the full SIP trace?
>
The full configuration is here:
http://pastebin.com/XqZG1m5X
I am connection over TLS / SRTP on port 5063.
When
2005 Sep 14
2
STUN vs NAT Helper
I'm wondering if anyone can recommend one over the other. I'm mostly
interested in running open source solutions, so I would prefer if
your recommendations are within the open source arena.
Basically, I contemplated the idea of using SER as a NAT Helper and
possibly as a SIP server for a portion of our user base. We prefer to
have Asterisk in the mix because of the additional
2005 Jan 21
0
Grandstreams+Nat
Good day all
I cant get my grandstream bt-100 to register
My asterisk is on a public ip and the phone behind a nat firewall
I added nat=yes in sip.conf and did this on my grandstream
set the GS to "SIP server=asterisk.yourhost.com" and leave "Outbound
Proxy" empty
* set the GS to SIP port 5060 and RTP port 5004 (and "Use random
port=No")
* set the GS to
2003 Oct 30
1
NAT type router database?
Is anybody aware of a database containing the types of nat
implementation in todays soho/consumer routers? I think it would make
sense for the community to have this database in order to avoid
symmetric nats.
If one such thing does not exist how about starting this database?
A stunclient for linux can be found at
http://sourceforge.net/projects/stun/
I can contribute this information for two
2005 May 10
2
Stun & codec
I have two phones, one does not need stun, the other one needs.
All settings are identically, except the number/password and said above
stun - not stun
I use codec in the order:
g729
g711u
g711a
Any ideas, why the user can hear me, but I cannot hear him (stun) while
the other user without stun has no problem.
bye
Ronald