On 29/03/17 16:18, Olivier wrote:
Hello,
After reading [1] (in french), I would be very happy if I could get
answers to:
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 ?
Debian's versioning scheme is all their own. And I would not expect it
to work with anything but a Debian-packaged Asterisk.
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.
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).
2. From package description, is this package enough or not to allow
transcoding with G711 ?
For instance, in the following situation:
SIP Phone <---- Opus ----> Asterisk <---- G111 ---> ITSP
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.
(Is "G111" a typo for "G711" ?)
3. Can you share here any personal field experience with this codec,
for home worker use case ?
Is there a better user experience with Opus than with G729 or G711 ?
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.
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 .....
4. Does it work on ARM boxes (Raspberry, ...) ?
The only thing that would prevent any software from working on ARM /
Raspberry Pi would be if it
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.
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.
[1] https://packages.debian.org/stretch/asterisk-opus
Regards
--
JM or AJS
--
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