I see the xml trace now.

the non vpn call is confusing ...

frame 23 the media invite to itsp is on

m=audio 30500 RTP/AVP 0 8 101

but the itsp sends back (frame 24)

m=audio 41302 RTP/AVP 0 101

and if you follow the same call from there on it seems like the media port
is changing. I don't know why.

(wish you could do pfsense where the filters are easier, so are pcaps).

On Thu, Jul 8, 2010 at 8:39 AM, [email protected] <[email protected]>wrote:

> Hi, Tony.
>
> Thank you very much again for your fast reply.
>
> May i ask you take a look at the trace that i attached already
> (sipx-trace-nonvpn-call.xml) file and see how all communication between
> user phone and ITSP goes? This is a call with audio problems. In this
> trace user does not use vpn. He is behind NAT and using local ip
> address.
>
> If is necessary i can send also a snapshot of the system, just let me
> know.
>
> About OpenVPN, i will check this and see how to go.
>
> On Thu, 2010-07-08 at 06:11 -0400, Tony Graziano wrote:
> >
> >
> > On Thu, Jul 8, 2010 at 4:21 AM, [email protected]
> > <[email protected]> wrote:
> >         Hi,
> >
> >         we have installed as you already know sipxecs behind NAT on a
> >         virtual
> >         box guest machine. You can see my expalation for our topology
> >         here in my
> >         previous topic:
> >
> http://forum.sipfoundry.org/index.php?t=msg&th=13717&start=0&S=60af1c90c584a680911ac1565d16b0ea
> >
> >         Now we have problem with one way audio on outgoing calls and a
> >         strange
> >         issue related with outgoing calls when we call via VPN
> >         network.
> >
> >         In the link above you will see that Tony Graziano sent to me a
> >         link with
> >         a diagram that is very useful.
> >
> >         I think that our problem is that RTP ports used in both legs
> >         are
> >         different and probably this is an issue with iptables rules,
> >         but i'm i
> >         would like someone to confirm me this that ports are different
> >         and if is
> >         possible to help me to solve this problem.
> >
> >         What i see in trace is that when the invite is sent to
> >         sipXproxy the
> >         audio port is one (30000), but in INVITE request from
> >         sipXproxy to
> >         sipxbridge is on different port ( 30248 ). Is that normal?
> >
> >         In traces i see also that when the user that do the call
> >         receive "183
> >         Session In Progress" then audio port is also different
> >         ( 30498 ). I
> >         suppose this is also wrong. Can you confirm this?
> >
> > To understand this, a siptrace of the call with audio problems would
> > tell a lot.
> >
> >         I attached also the iptables rules that i use right now. I
> >         followed Tony
> >         Graziano rules from his posts in my previous thread and also
> >         followed
> >         this article too:
> >
> >
> http://sipx-wiki.calivia.com/index.php/SipXbridge_Overview_and_Configuration#Firewall.2FNAT_Configuration
> >
> >         The other problem that we experience is related with calls
> >         from VPN
> >         network.
> >
> >         In trace that i have attached you can see on frame 23 that ACK
> >         package
> >         is sent to user's public IP address, not to the VPN address.
> >         Also in
> >         frame 23 is added a new VIA line that is wrong and this
> >         totally mess up
> >         the cominucation between sipx and user ( using vpn ):
> >         Via: SIP/2.0/UDP
> >         192.168.0.23:30256
> ;branch=z9hG4bK-d8754z-8375c91b95668768-1---d8754z-;rport
> >         Contact: <sip:[email protected]:30256>
> >
> >         So what can be the reason for this and how to solve this
> >         problem?
> >
> > Your openvpn should "push" the domain and resolve its records
> > internally. This way the softphone will register using the openvpn
> > address. BUT the audio needs to follow the vpn path. The only way I
> > know to get openvpn to do that is to have the client "route all
> > traffic" via the vpn, at least that's what I would start with.
> >
> >         P.S. We have these DNS records set:
> >
> >         ; SIP
> >         @  IN  NAPTR  10  0  "s"  "SIPS+D2T"  ""
> >          _sips._tcp.mydomain.net.
> >         @  IN  NAPTR  20  0  "s"  "SIP+D2U"   ""
> >          _sip._udp.mydomain.net.
> >         @  IN  NAPTR  30  0  "s"  "SIP+D2T"   ""
> >          _sip._tcp.mydomain.net.
> >
> >         ; SRV RECORDS
> >
> >         ; SIP
> >         _sips._tcp  IN  SRV  10  0  5060  sipx.mydomain.com.
> >         _sip._udp   IN  SRV  20  0  5060  sipx.mydomain.com.
> >         _sip._tcp   IN  SRV  30  0  5060  sipx.iguanait.com.
> >
> >         _sips._tcp  IN  SRV  40  0  5060  odin.mydomain.com.
> >         _sip._udp   IN  SRV  50  0  5060  odin.mydomain.com.
> >         _sip._tcp   IN  SRV  60  0  5060  odin.mydomain.com.
> >
> >
> >         odin.mydomain.com is the server with public ip address
> >         87.xxx.xxx.43
> >         sipx.mydomain.com is sipxecs server located behind NAt on
> >         virtual box.
> >         It has ip 10.1.1.2.
> >
> >         In traces the real domain is changed with 'mydomain' string
> >         and the
> >         public ip address is changed to 87.xxx.xxx.43.
> >
> >         IP: 91.2xx.xxx.17 is the user's public ip address that do the
> >         outgoing
> >         call.
> >
> >         IP: 10.1.1.5 is user's VPN address.
> >         IP: 192.168.0.23 is user's private IP address from his LAN's
> >         DHCP
> >         server.
> >
> >         The called number is: 883495466
> >
> >
> >         _______________________________________________
> >         sipx-users mailing list [email protected]
> >         List Archive: http://list.sipfoundry.org/archive/sipx-users
> >         Unsubscribe:
> >         http://list.sipfoundry.org/mailman/listinfo/sipx-users
> >         sipXecs IP PBX -- http://www.sipfoundry.org/
> >
> >
> >
> > --
> > ======================
> > Tony Graziano, Manager
> > Telephone: 434.984.8430
> > sip: [email protected]
> > Fax: 434.984.8431
> >
> > Email: [email protected]
> >
> > LAN/Telephony/Security and Control Systems Helpdesk:
> > Telephone: 434.984.8426
> > sip: [email protected]
> > Fax: 434.984.8427
> >
> > Helpdesk Contract Customers:
> > http://www.myitdepartment.net/gethelp/
> >
> > Why do mathematicians always confuse Halloween and Christmas?
> > Because 31 Oct = 25 Dec.
> >
>
>
>


-- 
======================
Tony Graziano, Manager
Telephone: 434.984.8430
sip: [email protected]
Fax: 434.984.8431

Email: [email protected]

LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: [email protected]
Fax: 434.984.8427

Helpdesk Contract Customers:
http://www.myitdepartment.net/gethelp/

Why do mathematicians always confuse Halloween and Christmas?
Because 31 Oct = 25 Dec.
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