[VoiceOps] High Quality, Reliable Voice via the Internet / SIPNOC

[Frank Bulk]

In a previous life my $PARTIME_JOB did some VoIP testing and found that VoIP
testing through a VPN tunnel resulted in a higher MOS. What we eventually
realized is that undelivered packets over the WAN were automatically
re-transmitted by the VPN.  This only works if the missing packet can be
re-transmitted before the far side's jitter buffer is drained.  So raising
the jitter buffer to 60 or 80 msec can help if the RTT is less than 60 msec.

Frank

-----Original Message-----
From: VoiceOps [mailto:voiceops-bounces at voiceops.org] On Behalf Of Alex
Balashov
Sent: Monday, June 09, 2014 3:14 PM
To: voiceops at voiceops.org
Subject: Re: [VoiceOps] High Quality, Reliable Voice via the Internet /
SIPNOC

On 06/09/2014 02:50 PM, Mark R Lindsey wrote:

> 2. Increase the ptime from 20 ms to 30-40 ms to reduce packet-drop
exposure

Question: does this actually reduce packet-drop exposure? One would 
think that with a longer duration of audio captured in a given packet, 
the loss of any individual packet would have more negative impact upon 
voice quality as subjectively experienced.

Or does this thesis lean on countervailing tendencies, such as overall 
reduced PPS in a higher ptime scenario?

-- 
Alex Balashov - Principal
Evariste Systems LLC
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

Please be kind to the English language:

http://www.entrepreneur.com/article/232906
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