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

[Carlos Chavez]

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
> 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,
> -- 
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-- 
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001