From: Frank Bulk (no email)
Date: Sat Nov 15 2008 - 17:49:16 EST
Nathan hit the nail on the head in his first sentence. With a VPN, if the
latency is low enough to allow retransmission of the UDP-based RTP packets
before the ATA's jitter buffer is starved, there can definitely be an
improvement in audio quality. This was documented in a VoIP testing review
that Network Computing performed several years ago. Since most jitter
buffers range from 20 to 80 msec (2 to 4 times the sampling size), it's
unlikely a hosted PBX with service delivered over the internet would benefit
from a VPN.
From: Nathan Ward [mailto:]
Sent: Wednesday, November 12, 2008 6:01 AM
To: nanog list
Subject: Re: hosted PBX/VOIP thru VPN?
On 13/11/2008, at 12:39 AM, Aaron Wolfe wrote:
> Because the broadband connection was so fast, TCP was able to
> repair the impairments without reducing voice quality. "
That works fine if latency+window size is low, so that segments are
You really should also do the math and factor in the latency that
comes from doing something like this, assuming you lose a packet. G.
114 recommends an end to end latency of no more than 150ms for voice
applications, where over 400ms is unacceptable (between 150 and 400
you should indicate that performance is not ideal).
Finally, some audio codecs work well with fairly high amounts of loss
- I'd recommend doing something like that first. iLBC does this really
well. G.729 etc. do not - they rely on context, so a single packet
lost results in several packets of lost audio (and so, silence). iLBC
doesn't rely on context, and quality degrades during packet loss
before you get silence.
The i stands for Internet - so no surprise it works great in typical
-- Nathan Ward