| Summary: | >=net-misc/asterisk-11.1 - add opus patch | ||
|---|---|---|---|
| Product: | Gentoo Linux | Reporter: | James Cloos <cloos> |
| Component: | [OLD] Server | Assignee: | Tony Vroon (RETIRED) <chainsaw> |
| Status: | RESOLVED FIXED | ||
| Severity: | normal | CC: | voip+disabled |
| Priority: | Normal | Keywords: | PATCH |
| Version: | unspecified | ||
| Hardware: | All | ||
| OS: | Linux | ||
| URL: | http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-12-current | ||
| Whiteboard: | |||
| Package list: | Runtime testing required: | --- | |
|
Description
James Cloos
2013-06-05 18:11:06 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. It is GPL, okay. Continue. 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 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 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. 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. Sorry for the ambiguity; I meant a backport from trunk to 12, if digium doesn’t do it upstream. Right. I am rather hoping they will, it seems silly not to. 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) |
