Sebastian Nielsen
2017-Apr-18 22:39 UTC
[asterisk-users] SIP connections over OpenVPN connection get one-way voice.
You need to ensure that traffic to the SIP box is sent to the correct IP. Also
if you use split-tunnel (eg: not redirect-gateway def1) you must make sure NAT
and traffic redirection works as is so the Asus router knows it should send the
traffic through tunnel and not via WAN.
IMPORTANT: Then you must, in the ASUS RT-N66U make a port forward inwards from
TUN to the phone client.
I would suggest wiresharking on the client side and see which IP Asterisk
suggest the client should connect back to. This should be the internal IP of the
asterisk server as seen from the openvpn server's point of view.
Another important thing: The local network in the Openvpns machine locatiin, may
NOT have same subnet as the network behind the asus.All these must be separate,
like:server network: 192.168.1.0/24Openvpn tunnel network: 192.168.2.0/24Asus
network: 192.168.3.0/24
Else you get bizarre routing problems when states appear in the state table.
-------- Originalmeddelande --------Fr?n: Ernie Dunbar <maillist at
lightspeed.ca> Datum: 2017-04-19 00:25 (GMT+01:00) Till: 'Asterisk
Users Mailing List - Non-Commercial Discussion' <asterisk-users at
lists.digium.com> Rubrik: [asterisk-users] SIP connections over OpenVPN
connection get one-way voice.
Hi everyone. I'm having some trouble with an OpenVPN tunnel that
isn't working *quite* as well as we'd hoped.
First, here's our technical details:
The OpenVPN server (v2.3.4-5+deb8u1) is a Debian 8 box behind a NAT
router. The router has UDP port 1194 forwarded to our server. This
server also runs our office Asterisk PBX, so there isn't any
networking hardware or firewall between the VPN tunnel and the
Asterisk PBX.
The OpenVPN client is an Asus RT-N66U router, which if I'm not
mistaken, runs a somewhat modified version of Tomato.
I've got the VPN tunnel working well enough. I can do practically
anything from a computer hooked up to the client router as if I were
in the main office where the server is. But any SIP client I use -
whether it's a hardware SIP phone or a soft phone like Zoiper, can
connect to the Asterisk server without issue. Making calls can work,
accepting calls works, but I only get 1 way voice traffic. I can
hear voice data coming in FROM the Asterisk PBX, but I cannot send
any.
In my experience with SIP, this usually means a firewall is breaking
the connection from the client phone to the Asterisk server. I just
can't for the life of me find what could be wrong. None of the other
traffic is being blocked. The ipfw firewall on the Asterisk PBX is
extremely open (see below). The firewall on the client router is
turned off, and as far as I can tell, most NAT routers don't even
block outbound traffic in the first place.
I can't see how traffic from the TUN interface on the OpenVPN server
even can be blocked going to another IP address on the same box, but
here are the IPFW rules:
root at ldinfo:/etc/asterisk# iptables -L -n
Chain INPUT (policy ACCEPT)
target prot opt source destination
ACCEPT all -- 192.168.0.0/24 192.168.0.3
ACCEPT all -- 192.168.1.0/24 192.168.0.3
ACCEPT all -- 10.8.0.0/24 192.168.0.3
ACCEPT all -- X.X.X.X 192.168.0.3
ACCEPT all -- 192.168.0.3 X.X.X.X
ACCEPT udp -- 0.0.0.0/0 0.0.0.0/0 udp dpt:1194
REJECT all -- 112.220.127.26 0.0.0.0/0 reject-with
icmp-port-unreachable
Chain FORWARD (policy ACCEPT)
target prot opt source destination
Chain OUTPUT (policy ACCEPT)
target prot opt source destination
Chain POSTROUTING (0 references)
target prot opt source destination
192.168.0.0/24 is the network the Asterisk PBX and OpenVPN server
are on.
192.168.1.0/24 is the network that the remote router is on.
10.8.0.0/24 is the network that the TUN device creates.
X.X.X.X is our datacenter.
192.168.0.3 is the IP address of our PBX.
Any assistance would be greatly appreciated.
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20170419/84dc1850/attachment.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: smime.p7s
Type: application/pkcs7-signature
Size: 6298 bytes
Desc: S/MIME Cryptographic Signature
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20170419/84dc1850/attachment.bin>
Ernie Dunbar
2017-Apr-18 23:40 UTC
[asterisk-users] SIP connections over OpenVPN connection get one-way voice.
<html>
<head>
<meta content="text/html; charset=windows-1252"
http-equiv="Content-Type">
</head>
<body text="#000000" bgcolor="#FFFFFF">
On 2017-04-18 03:39 PM, Sebastian Nielsen wrote:<br>
<blockquote
cite="mid:576123822.112.1492555173381.JavaMail.djigzo@linuxlite-desktop"
type="cite">
<meta http-equiv="Content-Type" content="text/html;
charset=windows-1252">
You need to ensure that traffic to the SIP box is sent to the
correct IP. Also if you use split-tunnel (eg: not redirect-gateway
def1) you must make sure NAT and traffic redirection works as is
so the Asus router knows it should send the traffic through tunnel
and not via WAN.</blockquote>
<br>
I'm not that well versed in OpenVPN, but it's worth noting that we
have the `push "redirect-gateway def1 bypass-dhcp"` directive set
on
the server. I have two independent DHCP servers on either side of
the VPN, so that the clients are getting their IP addresses for
their appropriate networks - 192.168.0.0/24 on the server side, and
192.168.1.0/24 on the client side. <br>
<br>
<blockquote
cite="mid:576123822.112.1492555173381.JavaMail.djigzo@linuxlite-desktop"
type="cite">
<div>IMPORTANT: Then you must, in the ASUS RT-N66U make a port
forward inwards from TUN to the phone client.</div>
</blockquote>
<br>
I'll give that a shot, but it will have to wait until tomorrow.
:)<br>
<br>
<blockquote
cite="mid:576123822.112.1492555173381.JavaMail.djigzo@linuxlite-desktop"
type="cite">
<div><br>
</div>
<div>I would suggest wiresharking on the client side and see which
IP Asterisk suggest the client should connect back to. This
should be the internal IP of the asterisk server as seen from
the openvpn server's point of view.</div>
<div><br>
</div>
<div>Another important thing: The local network in the Openvpns
machine locatiin, may NOT have same subnet as the network behind
the asus.</div>
<div>All these must be separate, like:</div>
<div>server network: 192.168.1.0/24</div>
<div>Openvpn tunnel network: 192.168.2.0/24</div>
<div>Asus network: 192.168.3.0/24</div>
</blockquote>
<br>
I'm pretty sure that I've got this subnet separation in place. If I
didn't cover it in my original post, the network looks like
this:<br>
<br>
Server network: 192.168.0.0/24<br>
OpenVPN network: 10.8.0.0/24<br>
Asus network: 192.168.1.0/24<br>
<br>
The Asterisk SIP registration appears to be responding properly to
this - this is what I see when I do a 'sip show peer' for an Aastra
phone that's connecting through the VPN (Asterisk output is
truncated): <br>
<br>
ToHost : <br>
Addr->IP : 10.8.0.6:5060<br>
Defaddr->IP : (null)<br>
Prim.Transp. : UDP<br>
Allowed.Trsp : UDP<br>
Def. Username: FrontDesk1<br>
SIP Options : (none)<br>
Codecs : (ulaw|alaw)<br>
Codec Order : (ulaw:20,alaw:20)<br>
Auto-Framing : No <br>
Status : Unmonitored<br>
Useragent : Aastra 6731i/3.2.2.1136<br>
Reg. Contact : <a class="moz-txt-link-abbreviated"
href="mailto:sip:FrontDesk1@10.8.0.6:5060;transport=udp">sip:FrontDesk1@10.8.0.6:5060;transport=udp</a><br>
<br>
<br>
<blockquote
cite="mid:576123822.112.1492555173381.JavaMail.djigzo@linuxlite-desktop"
type="cite">
<div><br>
</div>
<div>Else you get bizarre routing problems when states appear in
the state table.</div>
<div><br>
</div>
<div style="font-size:100%;color:#000000"><!--
originalMessage -->
<div>-------- Originalmeddelande --------</div>
<div>Från: Ernie Dunbar <a
class="moz-txt-link-rfc2396E"
href="mailto:maillist@lightspeed.ca"><maillist@lightspeed.ca></a>
</div>
<div>Datum: 2017-04-19 00:25 (GMT+01:00) </div>
<div>Till: 'Asterisk Users Mailing List - Non-Commercial
Discussion' <a class="moz-txt-link-rfc2396E"
href="mailto:asterisk-users@lists.digium.com"><asterisk-users@lists.digium.com></a>
</div>
<div>Rubrik: [asterisk-users] SIP connections over OpenVPN
connection get one-way voice. </div>
<div><br>
</div>
</div>
Hi everyone. I'm having some trouble with an OpenVPN tunnel that
isn't working *quite* as well as we'd hoped.<br>
<br>
First, here's our technical details:<br>
<br>
The OpenVPN server (v2.3.4-5+deb8u1) is a Debian 8 box behind a
NAT router. The router has UDP port 1194 forwarded to our server.
This server also runs our office Asterisk PBX, so there isn't any
networking hardware or firewall between the VPN tunnel and the
Asterisk PBX.<br>
<br>
The OpenVPN client is an Asus RT-N66U router, which if I'm not
mistaken, runs a somewhat modified version of Tomato. <br>
<br>
I've got the VPN tunnel working well enough. I can do practically
anything from a computer hooked up to the client router as if I
were in the main office where the server is. But any SIP client I
use - whether it's a hardware SIP phone or a soft phone like
Zoiper, can connect to the Asterisk server without issue. Making
calls can work, accepting calls works, but I only get 1 way voice
traffic. I can hear voice data coming in FROM the Asterisk PBX,
but I cannot send any. <br>
<br>
In my experience with SIP, this usually means a firewall is
breaking the connection from the client phone to the Asterisk
server. I just can't for the life of me find what could be wrong.
None of the other traffic is being blocked. The ipfw firewall on
the Asterisk PBX is extremely open (see below). The firewall on
the client router is turned off, and as far as I can tell, most
NAT routers don't even block outbound traffic in the first
place.<br>
<br>
I can't see how traffic from the TUN interface on the OpenVPN
server even can be blocked going to another IP address on the same
box, but here are the IPFW rules:<br>
<br>
root@ldinfo:/etc/asterisk# iptables -L -n<br>
Chain INPUT (policy ACCEPT)<br>
target prot opt source destination<br>
ACCEPT all -- 192.168.0.0/24 192.168.0.3<br>
ACCEPT all -- 192.168.1.0/24 192.168.0.3<br>
ACCEPT all -- 10.8.0.0/24 192.168.0.3<br>
ACCEPT all -- X.X.X.X 192.168.0.3<br>
ACCEPT all -- 192.168.0.3 X.X.X.X<br>
ACCEPT udp -- 0.0.0.0/0 0.0.0.0/0 udp dpt:1194<br>
REJECT all -- 112.220.127.26 0.0.0.0/0 reject-with
icmp-port-unreachable<br>
<br>
Chain FORWARD (policy ACCEPT)<br>
target prot opt source destination<br>
<br>
Chain OUTPUT (policy ACCEPT)<br>
target prot opt source destination<br>
<br>
Chain POSTROUTING (0 references)<br>
target prot opt source destination<br>
<br>
192.168.0.0/24 is the network the Asterisk PBX and OpenVPN server
are on.<br>
192.168.1.0/24 is the network that the remote router is on.<br>
10.8.0.0/24 is the network that the TUN device creates.<br>
X.X.X.X is our datacenter.<br>
192.168.0.3 is the IP address of our PBX.<br>
<br>
Any assistance would be greatly appreciated.<br>
<br>
<p>
<meta http-equiv="content-type" content="text/html;
charset=windows-1252">
</p>
<br>
<fieldset class="mimeAttachmentHeader"></fieldset>
<br>
</blockquote>
<br>
</body>
</html>
Mark Wiater
2017-Apr-19 19:38 UTC
[asterisk-users] SIP connections over OpenVPN connection get one-way voice.
On 4/18/2017 7:40 PM, Ernie Dunbar wrote:> Server network: 192.168.0.0/24 > OpenVPN network: 10.8.0.0/24 > Asus network: 192.168.1.0/24 > > The Asterisk SIP registration appears to be responding properly to > this - this is what I see when I do a 'sip show peer' for an Aastra > phone that's connecting through the VPN (Asterisk output is truncated): > > ToHost : > Addr->IP : 10.8.0.6:5060If the Asus network is 192.168.1.0/24, and the phone is registering as 10.0.8.6, it looks like NAT is taking place. Would your asterisk server know how to route traffic to 192.168.1.0/24? I've always used site-to-site OpenVPN tunnels where the vpn's terminate on the gateway for both the phones and the asterisk server. I've always had rock solid connections between phones and Asterisk.
Victor Villarreal
2017-Apr-20 01:13 UTC
[asterisk-users] SIP connections over OpenVPN connection get one-way voice.
Hi Ernie, When one-way audio appear (no matters if there is a VPN or NAT server on the diagram) I simply : * Enable SIP debug on Asterisk server. Excecute 'sip set debug ip x.x.x.x' on Astrisk CLI, where x.x.x.x is the IP of the phone or SIP peer you want to debug. * Make a test call and replicate the issue. * Stop debug with 'sip set debug off'. * Follow the SIP conversation. Verify that the INVITE message has the correct IP on the contact field and any other related fields. * On SDP handshake, verify that the ports where the sound is send, is correct. Normally, one-way audio is faced when one audio stream (example the called audio) is send to the correct IP and Port destination, on the other audio stream (example the caller audio) don't. Last, if Asterisk is 'behind' another server, you need tell Asterisk what is the external IP so it can inform this IP to your clients. If you dont want to follow the SIP conversation on plain text, you can make a packet capture on the Asterisk server, instead of SIP debug. El 19 abr. 2017 16:38, "Mark Wiater" <mark.wiater at greybeam.com> escribi?:> On 4/18/2017 7:40 PM, Ernie Dunbar wrote: > >> Server network: 192.168.0.0/24 >> OpenVPN network: 10.8.0.0/24 >> Asus network: 192.168.1.0/24 >> >> The Asterisk SIP registration appears to be responding properly to this - >> this is what I see when I do a 'sip show peer' for an Aastra phone that's >> connecting through the VPN (Asterisk output is truncated): >> >> ToHost : >> Addr->IP : 10.8.0.6:5060 >> > > If the Asus network is 192.168.1.0/24, and the phone is registering as > 10.0.8.6, it looks like NAT is taking place. Would your asterisk server > know how to route traffic to 192.168.1.0/24? > > I've always used site-to-site OpenVPN tunnels where the vpn's terminate on > the gateway for both the phones and the asterisk server. I've always had > rock solid connections between phones and Asterisk. > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170419/dc5d255d/attachment.html>