Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

VoIP Mailing List Archives
Mailing list archives for the VoIP community
 SearchSearch 

[asterisk-users] Erratic calls through NAT-ed server


 
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users
View previous topic :: View next topic  
Author Message
norman.laidla at teleg...
Guest





PostPosted: Thu Nov 13, 2014 9:24 am    Post subject: [asterisk-users] Erratic calls through NAT-ed server Reply with quote

Morning,

We recently pushed our Asterisk video bridge into a DMZ and since then, local calls have been unreliable to say the least. While offsite calls work nicely, calls on our internal server usually fail to ring the far end. Two test calls that were made 4 minutes apart yielded different results: one rang the far end, the other kept trying to transmit the Invite. The configuration didn't change at all between the two calls. I've been going over the debug logs, but haven't noticed any possible reasons why one call failed. It's the same all the way to the part where the far end is called.


The endpoints use different ports for UDP signaling and Asterisk is set to expect UDP packets from those ports. The RTP port range is the same between the ends (at least where it's configurable), Asterisk and the firewall. All ports that we're using have been opened in the firewall and incoming UDP traffic is routed to Asterisk. In Asterisk settings, localnet is defined as the LAN that both endpoints are on, externip is the public address of the server. Directrtpsetup and directmedia are both set to "no" and nat is set to "yes".


So, what could be causing this issue?


Best wishes,
Norman
Back to top
rnewton at digium.com
Guest





PostPosted: Fri Nov 14, 2014 5:39 pm    Post subject: [asterisk-users] Erratic calls through NAT-ed server Reply with quote

On Thu, Nov 13, 2014 at 8:23 AM, Norman Laidla
<norman.laidla@telegrupp.ee> wrote:
Quote:
Morning,

We recently pushed our Asterisk video bridge into a DMZ and since then,
local calls have been unreliable to say the least. While offsite calls work
nicely, calls on our internal server usually fail to ring the far end. Two
test calls that were made 4 minutes apart yielded different results: one
rang the far end, the other kept trying to transmit the Invite. The
configuration didn't change at all between the two calls. I've been going
over the debug logs, but haven't noticed any possible reasons why one call
failed. It's the same all the way to the part where the far end is called.

The endpoints use different ports for UDP signaling and Asterisk is set to
expect UDP packets from those ports. The RTP port range is the same between
the ends (at least where it's configurable), Asterisk and the firewall. All
ports that we're using have been opened in the firewall and incoming UDP
traffic is routed to Asterisk. In Asterisk settings, localnet is defined as
the LAN that both endpoints are on, externip is the public address of the
server. Directrtpsetup and directmedia are both set to "no" and nat is set
to "yes".

So, what could be causing this issue?

If out of multiple calls, some work and some don't - you either have
found a bug or something is really changing between the calls. That is
assuming the failing/working behavior does not fit an obvious pattern
(e.g. unique to a particular dialed remote party).

If you pastebin two Asterisk logs that show the working and failing
calls then someone may be able to look through them and spot an issue.

Be sure the Asterisk logs show VERBOSE and DEBUG channels at level 5
or above. See: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

You might also mention the exaction version of Asterisk you are using
and which channel driver (though it sounds like chan_sip based on the
options described).

--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200

Check us out at: http://digium.com & http://asterisk.org

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Back to top
norman.laidla at teleg...
Guest





PostPosted: Mon Nov 17, 2014 2:04 am    Post subject: [asterisk-users] Erratic calls through NAT-ed server Reply with quote

Problem fixed. The issue was that the timeout for the Options packet was longer than our NAT timeout.

Best wishes,
Norman
________________________________________
From: asterisk-users-bounces@lists.digium.com on behalf of Rusty Newton
Sent: Saturday, November 15, 2014 12:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Erratic calls through NAT-ed server

On Thu, Nov 13, 2014 at 8:23 AM, Norman Laidla
<norman.laidla@telegrupp.ee> wrote:
Quote:
Morning,

We recently pushed our Asterisk video bridge into a DMZ and since then,
local calls have been unreliable to say the least. While offsite calls work
nicely, calls on our internal server usually fail to ring the far end. Two
test calls that were made 4 minutes apart yielded different results: one
rang the far end, the other kept trying to transmit the Invite. The
configuration didn't change at all between the two calls. I've been going
over the debug logs, but haven't noticed any possible reasons why one call
failed. It's the same all the way to the part where the far end is called.

The endpoints use different ports for UDP signaling and Asterisk is set to
expect UDP packets from those ports. The RTP port range is the same between
the ends (at least where it's configurable), Asterisk and the firewall. All
ports that we're using have been opened in the firewall and incoming UDP
traffic is routed to Asterisk. In Asterisk settings, localnet is defined as
the LAN that both endpoints are on, externip is the public address of the
server. Directrtpsetup and directmedia are both set to "no" and nat is set
to "yes".

So, what could be causing this issue?

If out of multiple calls, some work and some don't - you either have
found a bug or something is really changing between the calls. That is
assuming the failing/working behavior does not fit an obvious pattern
(e.g. unique to a particular dialed remote party).

If you pastebin two Asterisk logs that show the working and failing
calls then someone may be able to look through them and spot an issue.

Be sure the Asterisk logs show VERBOSE and DEBUG channels at level 5
or above. See: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

You might also mention the exaction version of Asterisk you are using
and which channel driver (though it sounds like chan_sip based on the
options described).

--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200

Check us out at: http://digium.com & http://asterisk.org

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Back to top
Display posts from previous:   
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users All times are GMT - 5 Hours
Page 1 of 1

 
Jump to:  
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum


Powered by phpBB © 2001, 2005 phpBB Group

VoiceMeUp - Corporate & Wholesale VoIP Services