Josh,

Thanks for sharing, and that will work as long as your image's Asterisk version 
is compatible.

Good to know that the Opus binary blobs work.

Lonnie



On May 6, 2017, at 4:18 PM, Josh <jma...@hotmail.com> wrote:

> In case anybody else might want to implement Opus, I wanted to let you guys 
> know that I didn't actually need to implement my own development build.  I 
> tried the lazy way, and it worked.  First I upgraded to Astlinux 13.  Then I 
> simply downloaded the appropriate codec from:
> 
> http://downloads.digium.com/pub/telephony/codec_opus/asterisk-13.0/
> 
> And then I copied the .so files to /usr/lib/asterisk/modules.  Then, I had to 
> define opus in codecs.conf.  Now I'm in business!
> 
> From: mgra...@mstvp.com <mgra...@mstvp.com>
> Sent: Monday, April 17, 2017 12:45 PM
> To: AstLinux Users Mailing List
> Subject: Re: [Astlinux-users] Opus Codec Support
>  
> Josh,
>  
> Nope. It does not support dialing by SIP URI. You need to map the SIP URI to 
> an extension on your PBX.
>  
> Honestly, my impression is that they support Opus as an approach to low 
> bit-rate calling, like G.729 but without the licensing headache.
> 
> 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: 4/17/17 10:48 am
> To: "AstLinux Users Mailing List" <astlinux-users@lists.sourceforge.net>
> 
> Michael,
>  
> I play around with a lot of analog stuff but us crazy people in the club 
> often hook our switches up to channel banks to provide connectivity to our 
> Asterisk systems.  So, I'm somewhat familiar with the necessary A to D that 
> goes on.
>  
> Yeah, the Grandstream UI is not very nice IMO.  This is the first time I've 
> used one of their products and I'm definitely not impressed.
>  
> It's not immediately clear to me if the Grandstream device I have (HT-802) 
> supports direct SIP URI dialing.  I'll look into it.  If it does, we could 
> call eachother to test things out if you'd like.  That would be the easiest 
> thing for me since I have not added Opus to my Astlinux build, and don't plan 
> to for a few weeks at least.
>  
> Josh
> 
> From: mgra...@mstvp.com <mgra...@mstvp.com>
> Sent: Monday, April 17, 2017 10:37 AM
> To: AstLinux Users Mailing List
> Subject: Re: [Astlinux-users] Opus Codec Support
>  
> 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
> >>
> >>
> >>
> >> ------------------------------------------------------------------------------
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> >
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> >>
> >>
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> >
> >>
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> >
> >
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> >
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> 
> 
> --
> Kristian Kielhofner
> 
> ------------------------------------------------------------------------------
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