[asterisk-users] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?

[bruce bruce]

Thanks for that Carlos. I am playing with that right now. What do you
suggest localnet should say?

Server A = OpenVPN Server:
localnet=127.0.01
localnet=192.168.100.0/255.255.255.0

Where 192.168.100.0 is the DHCPd subnet of Server B (the openvpn client)

Server A doesn't have any localnet other than the loop back and then a Vnet
to internet (public ip address).

Thanks,
Bruce

On Wed, Sep 22, 2010 at 11:36 AM, Carlos Chavez <cursor at telecomabmex.com>wrote:

> Do you have a localnet statement in your sip.conf?  That and using
> nat=no will make sure Asterisk does not replace the IP address in the
> Invite.
>
> On Wed, 2010-09-22 at 01:27 -0400, bruce bruce wrote:
> > Hi Everyone,
> >
> >
> > I have setup an OpenVPN tunnel between Server A (running Asterisk) and
> > Server B suppling it's SIP Phones with DHCP pool of IPs.
> >
> >
> > So, the tunnel is established nicely and everyone can ping others.
> > "sip show peers" shows the local subnet of the SIP Phones registered
> > (192.168.100.0/24).
> >
> >
> > But there is the old bad one-way audio. Calls also drop after few
> > seconds. In the SIP debug I can see that asterisk uses it's external
> > public IP address to communicate to endpoints that are known to it as
> > the 192.168.100.0/24 endpoints and the endpoints identify themselves
> > with the OpenVPN tunnel IP address scheme in one part of the sip
> > handshake. How can this be fixed? After all, with the OpenVPN this
> > should all look like an internal network to Asterisk.
> >
> >
> > I have added my comments followed by # to lines below that are
> > problematic.
> >
> >
> > <--- SIP read from UDP:192.168.100.5:5060 --->    #This line is good
> > as it uses the local DHCP supplied network address scheme
> > INVITE sip:203 at 172.16.0.1:5060 SIP/2.0 #This line is BAD. Why are we
> > inviting Ext. 203 with it's OpenVPN IP while it's on the same network
> > of 192.168.50.0/24 as 202?
> > Via: SIP/2.0/UDP
> > 192.168.100.5:5060;branch=z9hG4bK695f8c1cfc7cdee96.1c65dc2eb25a46fc6
> Max-Forwards: 70
> > From: "SIP Phone - Ext. 202" <sip:202 at 172.16.0.1:5060>;tag=6d6f8c4226
> >    #BAD line again. Should be SIP:202 at 192.168.100.6<SIP%3A202 at 192.168.100.6>
> > To: "203" <sip:203 at 172.16.0.1:5060> #Bad again....
> > Call-ID: 43af67a634e06e75
> > CSeq: 32058 INVITE
> > Allow:  INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE,
> > PRACK, SUBSCRIBE, INFO
> > Allow-Events: talk, hold, conference, LocalModeStatus
> > Contact: "SIP Phone - Ext. 202"
> > <sip:202 at 192.168.50.5:5060;transport=udp>;
> > +sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D25B72F>"
> > Supported: gruu, path, timer, 100rel, replaces
> > User-Agent: Aastra 55i/2.5.2.1500
> > Content-Type: application/sdp
> > Content-Length: 594
> >
> >
> > Basically the phones should only send with FROM their local
> > 192.168.100.0/24 address and Asterisk should only send ANSWER and ACK
> > back to 192.168.100.0/24 rather than sending it to 172.16.0.0/24
> > (which is the openvpn client ip).
> >
> >
> > Once above is fixed, I think all the audio and call cut will go away.
> > I hate to use a sip proxy in this situation since I already have an
> > openvpn connection.
> >
> >
> > Any feed back is appreciated.
> >
> >
> > Thanks,
> > --
> > _____________________________________________________________________
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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> --
> Carlos Chavez
> Director de Tecnología
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Tel: +52-55-91169161 Ext 2001
>
>
> --
> _____________________________________________________________________
> -- 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
>
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