Can Remote IP phone sets work correctly over a VPN?

[freedomseeker]

We have a client that is considering purchasing a IP Office platform to tie together three of their facilities as a single extension range, centralized voicemail and PSTN lines.

But our data connectivity between the two remote facilities is over the raw Internet with a VPN w/ Triple DES from a NetGear Router. (We are running terminal emualtion to the remote locations.) The bandwidth (via a CLEC) is only 512k.

Currently, the IP Office dealer that we are talking with has not been able to explain or commit to us that we are not going to experience any type of voice latency, dropped packets or create problems with our existing data network activity. I have always heard that for IP voice - install a dedicated Point to Point pipe.

Anyone with experience with this type of application have any suggestions or advise before I make a recommendation and get myself into trouble?

Thanks,

Jim

 

[intrigrant]

You definitely must do a network assessment and test the quality of the connection.
I do not think the quality can be guaranteed on a VPN link if there no Qos mechanisme all through the VPN link.

 

[freedomseeker]

Does the issue of having the VPN using IPSEC for security encryption effect the phone sets ability to signal properly.

Example - time & date display, busy lamp lights. Basically any type of multicast signaling to the remote phone sets to update the above examples.

Your comment "I do not think the quality can be guaranteed on a VPN link if there no Qos mechanisme all through the VPN link". Are you saying that the voice quality is not going to be guarenteed for business type voice applications - dropped packets / Internet Router latency, etc...

Sounds like creating a P-to-P router managed WAN configuration would resolve the above issues to ensure QOS, multicast signalling, and security.

Thanks again,

Jim

 

[intrigrant]

Messaging between IP Offices is never a issue as this is ordinary TCP traffic.
Yes, a P-to-P router managed WAN configuration with low latency queuing would be a good option to get proper voice quality.