Hi Everyone,<div><br></div><div>I have setup an OpenVPN tunnel between Server A (running Asterisk) and Server B suppling it's SIP Phones with DHCP pool of IPs.</div><div><br></div><div>So, the tunnel is established nicely and everyone can ping others. "sip show peers" shows the local subnet of the SIP Phones registered (<a href="http://192.168.100.0/24">192.168.100.0/24</a>).</div>
<div><br></div><div>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 <a href="http://192.168.100.0/24">192.168.100.0/24</a> 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.</div>
<div><br></div><div>I have added my comments followed by # to lines below that are problematic.</div><div><br></div><div><div><--- SIP read from UDP:<a href="http://192.168.100.5:5060">192.168.100.5:5060</a> ---> #This line is good as it uses the local DHCP supplied network address scheme</div>
<div>INVITE <a href="http://sip:203@172.16.0.1:5060">sip:203@172.16.0.1:5060</a> 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 <a href="http://192.168.50.0/24">192.168.50.0/24</a> as 202?</div>
<div>Via: SIP/2.0/UDP 192.168.100.5:5060;branch=z9hG4bK695f8c1cfc7cdee96.1c65dc2eb25a46fc6 Max-Forwards: 70</div><div>From: "SIP Phone - Ext. 202" <<a href="http://sip:202@172.16.0.1:5060">sip:202@172.16.0.1:5060</a>>;tag=6d6f8c4226 #BAD line again. Should be <a href="mailto:SIP%3A202@192.168.100.6">SIP:202@192.168.100.6</a></div>
<div>To: "203" <<a href="http://sip:203@172.16.0.1:5060">sip:203@172.16.0.1:5060</a>> #Bad again....</div><div>Call-ID: 43af67a634e06e75</div><div>CSeq: 32058 INVITE</div><div>Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO</div>
<div>Allow-Events: talk, hold, conference, LocalModeStatus</div><div>Contact: "SIP Phone - Ext. 202" <sip:202@192.168.50.5:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D25B72F>"</div>
<div>Supported: gruu, path, timer, 100rel, replaces</div><div>User-Agent: Aastra 55i/2.5.2.1500</div><div>Content-Type: application/sdp</div><div>Content-Length: 594</div></div><div><br></div><div>Basically the phones should only send with FROM their local <a href="http://192.168.100.0/24">192.168.100.0/24</a> address and Asterisk should only send ANSWER and ACK back to <a href="http://192.168.100.0/24">192.168.100.0/24</a> rather than sending it to <a href="http://172.16.0.0/24">172.16.0.0/24</a> (which is the openvpn client ip).</div>
<div><br></div><div>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.</div><div><br></div><div>Any feed back is appreciated.</div>
<div><br></div><div>Thanks,</div>