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Bug 472430 - >=net-misc/asterisk-11.1 - add opus patch
Summary: >=net-misc/asterisk-11.1 - add opus patch
Status: RESOLVED FIXED
Alias: None
Product: Gentoo Linux
Classification: Unclassified
Component: [OLD] Server (show other bugs)
Hardware: All Linux
: Normal normal (vote)
Assignee: Tony Vroon (RETIRED)
URL: http://downloads.asterisk.org/pub/tel...
Whiteboard:
Keywords: PATCH
Depends on:
Blocks:
 
Reported: 2013-06-05 18:11 UTC by James Cloos
Modified: 2014-03-13 11:38 UTC (History)
1 user (show)

See Also:
Package list:
Runtime testing required: ---


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Description James Cloos 2013-06-05 18:11:06 UTC
Gentoo should add the opus patch to asterisk-11.

It patches with offset to each of 11.4.0, 11.3.0 and 11.2.2, as found in portage.

This is needed properly to support sip/webrtc, since opus is the mandatory to implement codec for sip/webrtc.

Reproducible: Always
Comment 1 Tony Vroon (RETIRED) gentoo-dev 2013-06-06 13:32:34 UTC
I have asked you before where the upstream bug is. Since there is none, and there is word of "legal problems", that suggests a license issue. That same license issue will preclude me from including it.
Comment 2 Tony Vroon (RETIRED) gentoo-dev 2013-06-06 13:34:15 UTC
It is GPL, okay. Continue.
Comment 3 Jaco Kroon 2013-06-07 13:50:22 UTC
Tony,

https://github.com/meetecho/asterisk-opus/blob/master/asterisk_opus%2Bvp8.diff

It touches chan_sip.c - please be careful.  The changes does look sane (mostly rtp changes, but there are updates to video handling too), and some stuff touches engine_rtp.c as well.

Kind Regards,
Jaco
Comment 4 Tony Vroon (RETIRED) gentoo-dev 2014-02-25 17:00:47 UTC
Asterisk trunk now has pass-through support, and this may well make it into version 12. It is too late for 11, but the patch you link to is not of a quality that I am comfortable carrying.
Evidence of VP8 pass-through:
https://reviewboard.asterisk.org/r/2723/
http://svnview.digium.com/svn/asterisk?view=revision&revision=397563
Comment 5 James Cloos 2014-02-26 08:30:16 UTC
Yes, it is too late for 11; I’m on 12 now anyway. ☺

(In fact, I’d forgotten about this bugz.)

It would be nice to have a backport of the trunk passthru code even if upstream doesn’t do it, though.
Comment 6 Tony Vroon (RETIRED) gentoo-dev 2014-02-26 12:36:56 UTC
The architectural differences between 11 & 12 far exceed the differences between 1.8 & 11. And even then the T38 backport frequently needed love. It seems unlikely, but if someone is willing to put in the work I will carry it.
Comment 7 James Cloos 2014-02-26 18:55:30 UTC
Sorry for the ambiguity;  I meant a backport from trunk to 12, if digium doesn’t do it upstream.
Comment 8 Tony Vroon (RETIRED) gentoo-dev 2014-02-26 19:06:01 UTC
Right. I am rather hoping they will, it seems silly not to.
Comment 9 Tony Vroon (RETIRED) gentoo-dev 2014-03-13 11:38:17 UTC
I am pleased to confirm passthrough support in Asterisk 12. As such, I will now close this report.

2013-08-23 15:49 +0000 [r397524-397527]  Matthew Jordan <mjordan@digium.com>

	* CHANGES: Update CHANGES file to reflect pass through support for
	  Opus/VP8

	* channels/chan_sip.c, res/res_pjsip_sdp_rtp.c,
	  include/asterisk/opus.h (added), include/asterisk/format.h,
	  channels/chan_pjsip.c, res/res_format_attr_opus.c (added),
	  main/channel.c, main/format.c, res/res_rtp_asterisk.c,
	  main/frame.c, main/rtp_engine.c: Add pass through support for
	  Opus and VP8; Opus format attribute negotiation This patch adds
	  pass through support for Opus and VP8. That includes: * Format
	  attribute negotiation for Opus. Note that unlike some other
	  codecs, the draft RFC specifies having spaces delimiting the
	  attributes in addition to ';', so you have "attra=X; attrb=Y".
	  This broke the attribute parsing in chan_sip, so a small tweak
	  was also included in this patch for that. * A format attribute
	  negotiation module for Opus, res_format_attr_opus * Fast picture
	  update for VP8. Since VP8 uses a different RTCP packet number
	  than FIR, this really is specific to VP8 at this time. Note that
	  the format attribute negotiation in res_pjsip_sdp_rtp was written
	  by mjordan. The rest of this patch was written completely by
	  Lorenzo Miniero. Review: https://reviewboard.asterisk.org/r/2723/
	  (closes issue ASTERISK-21981) Reported by: Tzafrir Cohen patches:
	  asterisk_opus+vp8_passthrough_20130718.patch uploaded by lminiero
	  (License 6518)