My goal is to have SIP and media go over the split tunnel VPN, with the SIP traffic going through FS (which it does) and the RTP stream between the VPN IP addresses of the two endpoints. I have inbound-bypass-media set to true as I don't want the media to go via FS.

VoIP on Cisco Meraki: F.A.Q. and Troubleshooting Tips Voice over IP (VoIP) is a common technology used in enterprise networks, allowing users on a network to make internal and outbound phone calls over the network. This article outlines a number of frequently asked questions regarding VoIP systems and technologies on Cisco Meraki networks, as well as some general troubleshooting tips and tricks. One-Way Audio via VPN on VOIP Solutions | Experts Exchange I recently had an issue just like this with a cisco voip installation where we had 1 way audio over a VPN. I'm reading about RTP being a necessity, and I don't see any RTP packets coming from the client side PC. However, here's something interesting. If I use the OpenVPN client to connect to the office, I get a symptom where the distant end vicidial.org • View topic - SIP/RTP Over SSL VPN Using ASA5510 Feb 14, 2013 RFC 7201 - Options for Securing RTP Sessions

Also don't overlook protocol inspection for SIP/RTP. It is enabled globally be default and rewrites the SDP information with post-NAT IPv4 addresses. This needs to be disabled for the VPN-connected clients.

SIP traffic not passing through the VPN. - J-Net Community A ping from local LAN to R-LAN1&2 is going through the VPN (seen in “get session”), but the RTP traffic is not going through the VPN. I have the following questions: - Does SIP ALG support this configuration? (SIP control traffic and RTP traffic use different subnets.) - Why the RTP traffic is not forwarded through the VPN? Solved: Disable NAT on SIP payload - breaks ICE - Check

Cisco IP Phone Certificates and Secure Communications

The VPN server has a iptables rule that masquerades all outgoing traffic. With a tcpdump at FreePBX server i see packets coming from 192.168.x.x but from asterisk console i see that the extension address is 10.8.x.x (the VPN client IP). In practice, i am able to call from remote (client over VPN… Solved: jabber via anyconnect vpn no vioce and - Cisco Also don't overlook protocol inspection for SIP/RTP. It is enabled globally be default and rewrites the SDP information with post-NAT IPv4 addresses. This needs to be disabled for the VPN-connected clients.