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:[email protected]] On Behalf Of Alex Balashov Sent: Monday, June 09, 2014 3:14 PM To: [email protected] 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 _______________________________________________ VoiceOps mailing list [email protected] https://puck.nether.net/mailman/listinfo/voiceops _______________________________________________ VoiceOps mailing list [email protected] https://puck.nether.net/mailman/listinfo/voiceops
