MTU doesn't appear to be the problem here. I have a feeling it is a symptom of soft-phone usage. The RTT that asterisk shows when doing a 'sip show peers' is based on the sending and return of a SIP OPTIONS packet to the user agent, in this case X-Lite. Soft phones unfortunately are fighting for resources on a machine and therefore don't always answer in the most timely fashion. If I use a hard phone, the problems disappear. I'm going to have to "close the book" on this one and chalk it up to soft-phone silliness.
Not of useful consequence, but I do find it interesting that many desktop phones have large varying differences in their RTT. I've got a Grandstream and Polycom phone on my desk connected to the same switch. The Polycom typically stays within 3-10ms while the Grandstream is generally 20-30ms. Meh. --- Tim Nelson RockBochs Inc. ----- "Chris Buechler" <[EMAIL PROTECTED]> wrote: > On Fri, Sep 19, 2008 at 7:29 AM, Paul Mansfield > <[EMAIL PROTECTED]> wrote: > > Tim Nelson wrote: > >> > >> Any ideas on what I can do to decrease the effect OpenVPN is having > on the traffic? All suggestions welcome and appreciated! > > > > a wild thought, but could you have a problem with MTU? try reducing > it > > on the remote client? > > > > That came to mind for me as well, but since this is VoIP I very > seriously doubt if that's the case. VoIP uses small packets to get > data (voice, actually) sent as quickly as possible. I would check > what > the frame sizes are being sent by the phone and whatever is on the > other end, but doubt if this will be the case. > > --------------------------------------------------------------------- > To unsubscribe, e-mail: [EMAIL PROTECTED] > For additional commands, e-mail: [EMAIL PROTECTED] --------------------------------------------------------------------- To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
