[asterisk-dev] Opus and VP8

Lorenzo Miniero lminiero at gmail.com
Mon May 27 09:20:04 CDT 2013


I may still have a patch for Asterisk 1.8 as well: I started working on
this a few months ago, but at the time there were problem with hooks when
using Opus (no audio when any audio hook was in place) and so I gave up
when I moved to Asterisk 11. If you think it might be of help I'll try and
retrieve that one too.

Lorenzo


2013/5/27 Olle E. Johansson <oej at edvina.net>

>
> 27 maj 2013 kl. 16:09 skrev Lorenzo Miniero <lminiero at gmail.com>:
>
> Dear all,
>
> I've just published the patch on github:
>
> https://github.com/meetecho/asterisk-opus
>
> The README should be quite self explainatory, but if you need any
> additional info feel free to ask me.
> Any feedback will be more than welcome!
>
>
> And if anyone wants to create patches for 1.8 and trunk - feel free to
> help!
>
> /O
>
>
> Lorenzo
>
>
> 2013/5/26 Olle E. Johansson <oej at edvina.net>
>
>>
>> 26 maj 2013 kl. 12:08 skrev Hans Witvliet <asterisk at a-domani.nl>:
>>
>> > Seems my mesage didn't reach the list...
>> > (could me my end of the list that's failing)
>> This is not about software logic, it's about legal issues and we can't
>> take that discussion here, it's for
>> Digium to make by themselves and we just have to respect their decision,
>> whatever we think
>> about it.
>>
>> We can create a community distribution based on GPL and we in the
>> community can
>> encourage as many as possible to use it - but it's their decision about
>> the type of legal
>> risc they consider it to be.
>>
>> Hopefully the issues that stop Digium from supporting it today will
>> diminish so that
>> Opus and VP8 can be included in the "official" distribution at some point
>> - and then
>> everything should be ready for some fast action.
>>
>> Digium has done everything they can for Asterisk - and more.
>> This time it's the community that has to prove that we can handle the
>> situation ;-)
>>
>> /O
>>
>> > Hans
>> >
>> > -----Original Message-----
>> > From: Hans Witvliet <asterisk at a-domani.nl>
>> > To: asterisk-dev at lists.digium.com
>> > Subject: Re: [asterisk-dev] Opus and VP8
>> > Date: Sat, 25 May 2013 12:19:13 +0200
>> >
>> > -----Original Message-----
>> > From: Olle E. Johansson <oej at edvina.net>
>> > Reply-to: Asterisk Developers Mailing List
>> > <asterisk-dev at lists.digium.com>
>> > To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
>> > Cc: Olle E. Johansson <oej at edvina.net>
>> > Subject: Re: [asterisk-dev] Opus and VP8
>> > Date: Fri, 24 May 2013 13:26:29 +0200
>> >
>> >
>> > 24 maj 2013 kl. 12:51 skrev Lorenzo Miniero <lminiero at gmail.com>:
>> >
>> >> PS: a few months ago I also talked, on the #asterisk-dev IRC, about
>> >> the support I added for both Opus (transcoding) and VP8 (passthrough)
>> >> in Asterisk, codecs that are currently the default ones used in
>> >> WebRTC. I checked whether there was an interest in a patch for them,
>> >> but at the time there were some concerns about the copyright status of
>> >> Opus that prevented it to be considered for integration in Asterisk.
>> >> Has this situation changed in the meanwhile? I can open a separate
>> >> thread for this if needed.
>> >>
>> > Lorenzo,
>> >
>> >
>> > Good seeing you here!
>> >
>> >
>> > Due to legal issues I don't think Digium can accept a contribution of
>> > Opus and VP8 in the svn repositories today.
>> >
>> >
>> > I would encourage you, if you have these patches, to publish them on a
>> > web site like github or sourceforge so w all can help you test it. I
>> > really would like for these to be available for the community in an easy
>> > form.
>> >
>> >
>> > Some things can be done in Asterisk though and that's the code points
>> > for pass through media. I don't think that would cause any legal
>> > issues.
>> >
>> >
>> > Hi Olle,
>> >
>> > I understand that companies like Digium are very carefully with regards
>> > to legal aspects, but how come that another USA-based company can
>> > use/ship vp8 freely (linphone). The European based company that
>> > builds/distribute Jitsi also ships it in their latest version:
>> >
>> > Linphone:
>> > Audio with the following codecs: speex (narrow band and wideband), G711
>> > (ulaw,alaw), GSM, G722. Through additionals plugins, it also supports
>> > AMR-NB, SILK, G729 and iLBC.
>> > Video with codecs: VP8 (WebM), H263, H263-1998, MPEG4, theora and H264
>> > (thanks to a plugin based on x264), with resolutions from QCIF(176x144)
>> > to SVGA(800x600) provided that network
>> >
>> > Jitsi:
>> > "Among the most prominent new features you will find quality multi-party
>> > video conferences for XMPP, audio device hot-plugging, support for
>> > Outlook presence and calls, an overhauled user interface and support for
>> > the Opus and VP8 audio/video codec. You can download the new version at
>> > the following location: https://download.jitsi.org/"
>> >
>> >
>> >
>> >
>>
>>
>
>
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