Transcoding is something that’s not been mentioned here yet. Especially with 
the growth of Microsoft Teams using Silk audio codec driving wideband telephony 
and all voice arriving from a wireless network needs to be transcoded before 
going to the PSTN which is clamped at G.711 narrowband.

Packet loss & jitter (and latency which is not part of normal voice clarity 
measurements) cause poor voice quality as perceived by the user. Maybe they 
don’t have QoS set for voice on their LAN/ customer premises, or likely gets 
stripped off as it comes in over ISP/access link/last Mile.

Congestion, CPU utilization, audio set up (microphone etc.)  and other 
impairments introduced by the desktop/PC/Softphone (transmitting & receiving) 
will also impact call quality

+ anything processing audio in the network such as a transcoder.

The “MOS” value you typically see from a packet monitoring system is derived 
from R factor and only takes into account packet loss and jitter. But the user 
experience is based on the AUDIO they receive. the only MOS measurement to 
quantify this is AUDIO MOS or PESQ/POLQA MOS and involves transmitting an audio 
file across the network and comparing it with its reference.

if you're concerned about customs complaining poor UX, record a small sample of 
their audio coming from them (with their permission of course, usual waivers 
etc.] and send it back to them as a pcap, so they can listen to it for 
themselves. If you need any help decoding anything other than G.711 in 
Wireshark, let us know.

Many Thanks & Best Regards,

Richard Jobson
Teraquant Corporation
ph: 719 488 1003
d/l: (719) 766-8523
www.teraquant.com<http://www.teraquant.com/>
[email protected]
https://www.linkedin.com/in/uc-expert-monitoring/

Network Monitoring and Service Assurance - Speech Quality Experts (PESQ/POLQA) 
and Active Testing - Reporting – HPBX - Session Border Controllers – SDN and 
SD-WAN - Big Data Analytics and fraud detection and protection.

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From: VoiceOps <[email protected]> on behalf of Tim Bray via 
VoiceOps <[email protected]>
Reply-To: Tim Bray <[email protected]>
Date: Monday, June 14, 2021 at 3:23 AM
To: Mike Hammett <[email protected]>, <[email protected]>
Subject: Re: [VoiceOps] Call Quality


Hi,



It depends on your definition of pipes not too overloaded.   And I'm presuming 
from mention of pipes that you mean network induced call quality problems.



Usually bufferbloat.   Routers with too much memory cause a lot of latency at 
the point of a fast to slow transition in the network.





And this can be caused by anything from a crappy DSL router on upstream, and 
somebody emails a large attachment during a call.   Or it can be something like 
a unsupported 100meg optic on the customer side of a juniper edge router on a 
10gig core.    Customer does a download, latency goes nuts and all the phone 
calls sound naff.



Tools to test.

fast.com and press the `Show more info` button.   Forget the bandwidth figures, 
and look at the difference between the loaded and unloaded latency.  If a big 
difference, you have a problem.





http://www.dslreports.com/speedtest  This will give a bufferbloat score.  Then 
look at the results, and scroll down a bit and it will show you upload and 
download latency figures for idle, downloading and uploading.    (This is one 
of the most amazing tools, and I'd love a way to pay them some money each month 
to support the service.  They were struggling a bit at some stage.)







Tim




On 13/06/2021 19:11, Mike Hammett wrote:
I've heard a variety of complaints and concerns over the years about call 
quality. How are these quality issues introduced? As long as pipes and 
equipment aren't overloaded, where is a quality issue to come from?


Obviously, the closer you are to the handsets, the less opportunity there is 
for issues. What else is there to take into account?


-----
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com



Midwest Internet Exchange
http://www.midwest-ix.com






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