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RE: hosted PBX/VOIP thru VPN?
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. Frank -----Original Message----- From: Nathan Ward [mailto:[email protected]] 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 retransmitted quickly. 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 Internet conditions. -- Nathan Ward