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<div class="moz-cite-prefix">On 29/03/17 16:18, Olivier wrote:<br>
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<div>Hello,<br>
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After reading [1] (in french), I would be very happy if
I could get answers to:<br>
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1. Does this 13.7+20161113-3 package version has any
relation with asterisk's version it complements ? Current
asterisk version in repo is 13.14.0. Does this 13.7
complies with it ?<br>
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Debian's versioning scheme is all their own. And I would not expect
it to work with anything but a Debian-packaged Asterisk.<br>
<br>
Stretch is currently the "testing" distribution. This means that
new versions of packages could appear at any time; but if a
newly-introduced package breaks any other packages, they will be
removed from "testing" (and replaced as soon as possible with
newer, compatible versions) rather than allow packages to exist in
the repository that cannot be co-installed.<br>
<br>
If you really want to use a newer Asterisk version, the Debian
source will contain a file called "rules", which is really a
Makefile "in disguise". This should give you a good clue as to how
to hand-build an equivalent based on more up-to-date Source Code
(if the compile-time options have not changed too much, then you
might even get away with using it directly, but consider this a
bodge).<br>
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<div>2. From package description, is this package enough or
not to allow transcoding with G711 ?<br>
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For instance, in the following situation:<br>
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SIP Phone <---- Opus ----> Asterisk <---- G111 --->
ITSP<br>
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All codecs can input and output raw, uncompressed PCM; so as long as
you build all the relevant modules, your Asterisk will be able to
transcode between any two codecs it supports.<br>
<br>
(Is "G111" a typo for "G711" ?)<br>
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<div>3. Can you share here any personal field experience
with this codec, for home worker use case ?<br>
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<div>Is there a better user experience with Opus than
with G729 or G711 ?<br>
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Opus is, to the best of my knowledge, fully Open Source. G729 was
encumbered by patents in some jurisdictions, though it's now
patent-free. <br>
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G.711 A-Law is what the PSTN uses natively, and that is unlikely to
change anytime soon; though some VoIP providers are bringing Opus
online already. If you have many phones connected to your Asterisk,
then you may run into CPU limitations transcoding incoming and
outgoing calls between G711 and Opus. But that depends on your
Asterisk server. If you are recording calls, Asterisk will already
have to convert both the incoming and outgoing legs to raw PCM
anyway. In any case, if your provider supports Opus, you can
offload the donkey work to them .....<br>
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<div>4. Does it work on ARM boxes (Raspberry, ...) ?<br>
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The only thing that would prevent any software from working on ARM /
Raspberry Pi would be if it<br>
contained any architecture-specific binary code without Source Code
(which you could just about get away with, if you released it under
LGPL plus exceptions or an Apache licence). And I suspect if any
such code existed, it would be rewritten in fairly short order
anyway.<br>
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Also, it's Debian; and they really, really don't like binary blobs,
only grudgingly banishing them to a special "non-free" section which
is not even enabled by default. And that package was in the main
repository, suggesting full Source Code availability. In any case,
I see builds for armhf (R.Pi 1 and 2) and arm64 (R.Pi 3); so
even if there is some sneaky binary-only component, you will be able
to get it to work.<br>
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[1] <a moz-do-not-send="true"
href="https://packages.debian.org/stretch/asterisk-opus">https://packages.debian.org/stretch/asterisk-opus</a><br>
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<div>Regards<br>
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<p>--</p>
<p>JM or AJS<br>
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