I suggest that your past experience was based upon wholly avoiding and kind of
digital anything. A wholly analog path placed no arbitrary constraints on the
performance.
Add any kind of traditional telecom digital device and you will not see that
same performance. The A/D process must bandwidth limit the signal to avoid
aliasing, which is both audible and objectionable.
I've had a look at a Grandstream ATAs that implement Opus. I've not been able
to get it to interop using Opus with anything I have on hand. I've tried a
Polycom VVX-600, Grandstream GXV phones and various soft phones.
I admit that I haven't tried very hard. Once I saw the web UI of the ATA I
somewhat lost interest in tinkering with it. It looks like it's crafted by
grade school kids.
I still have the ATA. I even bought a Trimline phone to have a suitable analog
end-point. I could expose it via SIP URI of someone would like to make a couple
of test call. Maybe even assign it a DID.
Michael Graves
mgra...@mstvp.com
http://www.mgraves.org
o(713) 861-4005
c(713) 201-1262
sip:mgra...@mjg.onsip.com
skype mjgraves
--------- Original Message --------- Subject: Re: [Astlinux-users] Opus Codec
Support
From: "Josh" <jma...@hotmail.com>
Date: 4/16/17 9:17 pm
To: "AstLinux Users Mailing List" <astlinux-users@lists.sourceforge.net>
I have old Western Electric analog phones that don't have any sort of filters
or hard limiting. I'd assume most cordless phones have ~8KHz sample rates,
however, but all modern analog phones I've used aren't hard limited. With a
cordless phone, there's a reason to do that (to make efficient use of spectrum)
but with an analog phone, it would only add unnecessary complexity to the
design.
See this frequency response from a recording I made from a Western Electric
analog telephone. The call was analog end to end through old electromechanical
telephone switching equipment that I mess with, and I sampled at 22050Hz. A
higher sample rate would have illustrated my point better, but this what I had
on hand.
http://imgur.com/a/FYTsO
I'm sure an IP phone will produce higher fidelity audio over any codec than an
analog phone, but improvement will definitely be noticed, at least by my ears,
using an "HD" codec with an analog phone such as a Western Electric 2500 or 500
set.
Josh
From: David Kerr <da...@kerr.net>
Sent: Sunday, April 16, 2017 8:08 PM
To: AstLinux Users Mailing List
Subject: Re: [Astlinux-users] Opus Codec Support
I was about to reply to Josh with much the same comments... if the ultimate
device is an analog attached telephone then it is unnecessary and a complete
waste to bother with HD codecs.
But in saying that, in the last year I have upgraded the equipment in my house
from ATA connected devices to pure VoIP (I went with one Yealink T46G and five
W52P's) which I connect with g722 codec and the difference in sound quality is
noticeable... especially if you can keep everything in the chain HD. But even
going out to a VoIP trunk that uses uLaw/aLaw ultimately to e.g. my iPhone
using VoLTE the quality is noticeable better... presumably because there is no
analog signal anywhere in the chain even though the digital signal is
downsampled as it goes through the trunk.
Anyone know of a VoIP trunk vendor that will support HD voice codecs... and
keep HD through its paths if the other end also supports HD?
David
On Sun, Apr 16, 2017 at 4:13 PM, Kristian Kielhofner <k...@kriskinc.com>
wrote:
Hi Josh,
The availability of Opus on an ATA is a little strange because
virtually any analog POTS device you connect to it will undoubtedly
have filters and/or other hardware limitations in place for the 300 -
3300Hz frequency range supported by the PSTN.
To support frequencies beyond the usual ~3 KHz frequency range of
the PSTN you need different audio hardware to support the increased
frequency range represented in sample rates greater than 8KHz (8-48KHz
being supported by the Opus codec itself).
Polycom and other handset manufacturers that support "wideband
audio" made very noticeable changes to the audio components (the
speakers and microphones of the handsets, speakerphones, etc) used
with their devices. Hardware like mobile devices, desktops, etc using
Opus with Chrome/WebRTC/etc tend to best support frequency ranges up
to 48KHz even though it's well beyond the "standard" 20-20,000Hz range
of most audio components and human hearing.
In short, it's unlikely you'll gain any noticeable improvement in
audio quality by using Opus with an ATA and depending on the analog
device itself you may actually have worse audio quality due to
clipping and all kinds of other strange things that could happen when
an analog device designed for the PSTN of the last 100 years is
connected to something that can sample audio well beyond anything the
audio hardware itself was designed for.
Here's a fun site that tests the range your audio hardware can
produce (and what your ears can hear):
http://onlinetonegenerator.com/hearingtest.html
I'm 33 years old and my hearing tops out at about 15 KHz but for now
I'll make myself feel better by blaming my cheap computer speakers ;)!
On Sun, Apr 16, 2017 at 2:51 PM, Josh <jma...@hotmail.com> wrote:
> I had read that the Asterisk 13 config changes were likely to affect many
> users, but that wasn't on my mind. Thanks for mentioning it!
>
>
> Yeah, I'm not wild about the details of Opus either but it's the only way I
> can use "HD" voice with the hardware I've got, so I figure I'll give it a
> whirl pretty soon. I just bought some unlockable $10 Vonage-branded
> Grandstream HT-802 ATAs which were too enticing to pass up! If you're
> interested in messing with that, the details of the unlocking method are
> available on page 2 of this thread:
>
>
>
> https://www.dslreports.com/forum/r30988922-Unlock-New-Grandstream-ATA-used-by-Vonage-Unlocked
>
>
> It requires a serial cable with 3.3V logic and actually didn't even require
> soldering since you can easily stick individual pin headers in the three
> appropriate holes (TX RX and GND). I'm comfortable with those methods so
> the actual process took me about 5 minutes, but reading the entire thread
> and digesting all of the information (which I've learned to do when
> unlocking a device I don't want to accidentally brick) took considerably
> longer.
>
>
> Josh
>
> ________________________________
> From: Lonnie Abelbeck <li...@lonnie.abelbeck.com>
> Sent: Sunday, April 16, 2017 11:42 AM
>
> To: AstLinux Users Mailing List
> Subject: Re: [Astlinux-users] Opus Codec Support
>
> Josh,
>
> To be clear, upgrading the base system to Asterisk 13 is easy, but you will
> have to tweak your Asterisk configuration.
>
> This URL, at the end look for "Upgrade Note for Asterisk 11 -> 12" and
> Upgrade Note for Asterisk 12 -> 13 ...
>
> Asterisk LTS Series Version
> https://doc.astlinux-project.org/userdoc:tt_asterisk_upgrade_version
>
> If you built your own custom AstLinux image with Asterisk 13, you could add
> to a finished build the proper downloaded binary blobs from Digium
> "format_ogg_opus.so" and "codec_opus.so" to their proper location at
> output/target/usr/lib/asterisk/modules/ and quickly rebuild with
> ./scripts/build . Never tried it, those Opus binary blobs send shivers down
> my spine. :-)
>
> Lonnie
>
>
> On Apr 16, 2017, at 10:08 AM, Josh <jma...@hotmail.com> wrote:
>
>> Lonnie,
>>
>> Oops, I missed the previous discussion about Opus. Thanks for pointing me
>> to it! I would really like to use Opus with Astlinux now that I have an
>> HT802 ATA that supports it but not SILK (as well as people I call via SIP
>> who use Opus). However, I was aware of the legal issues surrounding it so I
>> had a hunch it might be a hassle to redistribute it with Astlinux. I guess
>> I could roll my own Astlinux build with Opus support once you adopt 13.13
>> (which I believe is the first build that includes Opus in menuselect),
>> right?
>>
>> The upgrade to 13 looks easy. Thanks for confirming!
>>
>> Josh
>>
>> From: Lonnie Abelbeck <li...@lonnie.abelbeck.com>
>> Sent: Sunday, April 16, 2017 9:05 AM
>> To: AstLinux Users Mailing List
>> Subject: Re: [Astlinux-users] Opus Codec Support
>>
>> Hi Josh, (comments inline)
>>
>> On Apr 15, 2017, at 9:46 PM, Josh <jma...@hotmail.com> wrote:
>>
>> > Hello,
>> >
>> > Does the current Asterisk 13 version of Astlinux support the Opus codec?
>>
>> No, Digium's Opus binary blob is not included, but the SILK CODEC is
>> included.
>>
>> Here is a related mailing-list thread on the subject ...
>>
>> [Astlinux-users] Opus CODEC
>>
>>
>> https://www.mail-archive.com/astlinux-users@lists.sourceforge.net/msg08638.html
>>
>>
>> > Also, I'm currently running astlinux-1.2.8 i686 - Asterisk 11.23.1. If
>> > I want to upgrade to version 13, can I just change my repository URL and
>> > do
>> > a regular upgrade through the web GUI?
>>
>> Yes, exactly.
>>
>> Later, should you want to use Asterisk 11.25.1 instead, change your
>> repository URL back to ast11, then System tab -> System Firmware Upgrade: [
>> Revert to Previous ] then [ Upgrade with New ] and lastly Reboot/Restart
>> System.
>>
>> Lonnie
>>
>>
>>
>> ------------------------------------------------------------------------------
>> Check out the vibrant tech community on one of the world's most
>> engaging tech sites, Slashdot.org! http://sdm.link/slashdot
>> _______________________________________________
>> Astlinux-users mailing list
>> Astlinux-users@lists.sourceforge.net
>> https://lists.sourceforge.net/lists/listinfo/astlinux-users
> Astlinux-users Info Page - SourceForge
> lists.sourceforge.net
> Welcome to the AstLinux users mailing list! To see the collection of prior
> postings to the list, visit the Astlinux-users Archives. Using Astlinux ...
>
>> Astlinux-users Info Page - SourceForge
>> lists.sourceforge.net
>> Welcome to the AstLinux users mailing list! To see the collection of prior
>> postings to the list, visit the Astlinux-users Archives. Using Astlinux ...
>>
>>
>> Donations to support AstLinux are graciously accepted via PayPal to
>> pay...@krisk.org.
>>
>> ------------------------------------------------------------------------------
>> Check out the vibrant tech community on one of the world's most
>> engaging tech sites, Slashdot.org!
>> http://sdm.link/slashdot_______________________________________________
>> Astlinux-users mailing list
>> Astlinux-users@lists.sourceforge.net
>> https://lists.sourceforge.net/lists/listinfo/astlinux-users
> Astlinux-users Info Page - SourceForge
> lists.sourceforge.net
> Welcome to the AstLinux users mailing list! To see the collection of prior
> postings to the list, visit the Astlinux-users Archives. Using Astlinux ...
>
>>
>> Donations to support AstLinux are graciously accepted via PayPal to
>> pay...@krisk.org.
>
>
> ------------------------------------------------------------------------------
> Check out the vibrant tech community on one of the world's most
> engaging tech sites, Slashdot.org! http://sdm.link/slashdot
> _______________________________________________
> Astlinux-users mailing list
> Astlinux-users@lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
> Astlinux-users Info Page - SourceForge
> lists.sourceforge.net
> Welcome to the AstLinux users mailing list! To see the collection of prior
> postings to the list, visit the Astlinux-users Archives. Using Astlinux ...
>
>
> Donations to support AstLinux are graciously accepted via PayPal to
> pay...@krisk.org.
>
> ------------------------------------------------------------------------------
> Check out the vibrant tech community on one of the world's most
> engaging tech sites, Slashdot.org! http://sdm.link/slashdot
> _______________________________________________
> Astlinux-users mailing list
> Astlinux-users@lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to
> pay...@krisk.org.
--
Kristian Kielhofner
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http://sdm.link/slashdot_______________________________________________
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support AstLinux are graciously accepted via PayPal to pay...@krisk.org.
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