[asterisk-dev] Opus and VP8

Mon May 27 09:53:27 CDT 2013

Hi Olle,

answering inline.

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

> Thank you, Lorenzo!
>
> A few comments:
>
> We need to find a better way to handle video update requests - your code
> change in chan-sip is not very
> modular. We should update the existing ast_rtcp_send_h261fur() function
> if that one doesn't work.
>
>
Actually ast_rtcp_send_h261fur() does not exist anywhere in the code: it
was already commented in chan_sip. What I added was simply a "hack" to send
a RTCP FIR message in the VP8 case, rather than a SIP INFO that was the
default behaviour (the transmit_info_with_vidupdate that is there). I did
it this way to minimize the impact on the existing code, but I'll welcome
any indication on better ways to procede!

> Also, your adding this:
> ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d/2\r\n", rtp_code, mime, rate);
>
> I don't think Asterisk supports parsing that on incoming SDP. Have you
> tested this?
>
>
The append you see is mandated by the current specification:

      http://tools.ietf.org/html/draft-ietf-payload-rtp-opus-00#section-6.2
      The RTP clock rate in "a=rtpmap" MUST be 48000 and the number of
channels MUST be 2.

Asterisk seems to parse this fine, though (both browsers do this, I can't
remember if PhonerLite does as well), as Opus is correctly negotiated. I
can't remember any issue when checking with "sip set debug on" either. My
guess is that in the "rtpmap: %30u %127[^/]/%30u" sscanf it is simply
ignored.

> Apologize for the colors, it's from cut/paste.
>
>
Long live colors! :-)
Lorenzo

> /O
>
> 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!
>
> 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/"
>> >
>> >
>> >
>> >
>>
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20130527/507a7008/attachment-0001.htm>