Olivier
2017-Mar-29 15:18 UTC
[asterisk-users] Questions regarding asterisk-opus package in Debian Stretch repo and Opus in general
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 ? 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 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 ? 4. Does it work on ARM boxes (Raspberry, ...) ? [1] https://packages.debian.org/stretch/asterisk-opus Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170329/bc79937a/attachment.html>
Tzafrir Cohen
2017-Mar-29 15:28 UTC
[asterisk-users] Questions regarding asterisk-opus package in Debian Stretch repo and Opus in general
On Wed, Mar 29, 2017 at 05:18:18PM +0200, 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 ?The opus codec was used as an external patch. It looked ugly and thus a separate package was preffered. Its version number is not directly related to Asterisk. It has originally been split from the Debian packaging of Asterisk, and starting from the same version number allowed easier upgrading. There is no version number for the upstream code (the patch).> > 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 ---> ITSPTechnically Asterisk codecs translate to/from (typically) linear and Asterisk combines codecs to do whatever transcoding needed. So the codec does not transcode directly to G.711. But Asterisk can transcode between opus and G.711.> > 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 ? > > 4. Does it work on ARM boxes (Raspberry, ...) ?Should work just the same.> > > [1] https://packages.debian.org/stretch/asterisk-opus-- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.cohen at xorcom.com http://www.xorcom.com
Olivier
2017-Mar-29 16:04 UTC
[asterisk-users] Questions regarding asterisk-opus package in Debian Stretch repo and Opus in general
2017-03-29 17:28 GMT+02:00 Tzafrir Cohen <tzafrir.cohen at xorcom.com>:> On Wed, Mar 29, 2017 at 05:18:18PM +0200, 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 ? > > The opus codec was used as an external patch. It looked ugly and thus a > separate package was preffered. >Thank you very much for this informative answer. Is there any relation between this external patch and the binary mentioned in [2] [2] http://blogs.digium.com/2016/09/30/opus-in-asterisk/ The later one mentions a binary-only distribution to comply with legal constraints. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170329/974c557d/attachment.html>
JM or AJS
2017-Mar-29 18:13 UTC
[asterisk-users] Questions regarding asterisk-opus package in Debian Stretch repo and Opus in general
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 ---> ITSPAll 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170329/ba9e6d67/attachment.html>
Maybe Matching Threads
- No audio after a reinvite changing codec
- SIP codec negotiation / manipulation
- Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)
- Asterisk 1.6 T38 to G711 transcoding is this possible?
- speex, ilbc and g729 codecs, GSM with IAX