----------------------------------------------------------- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/2723/#review10605 -----------------------------------------------------------
/trunk/channels/chan_sip.c <https://reviewboard.asterisk.org/r/2723/#comment20096> Shouldn't this be &a_audio instead of &a_text, or does it matter? - Sean Bright On Aug. 23, 2013, 3:42 p.m., Matt Jordan wrote: > > ----------------------------------------------------------- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/2723/ > ----------------------------------------------------------- > > (Updated Aug. 23, 2013, 3:42 p.m.) > > > Review request for Asterisk Developers, Joshua Colp and Mark Michelson. > > > Bugs: ASTERISK-21981 > https://issues.asterisk.org/jira/browse/ASTERISK-21981 > > > Repository: Asterisk > > > Description > ------- > > Note: This patch was written by Lorenzo Miniero. I know he's at the IETF this > week, but I figured we could get the formal code review going for him :-) > > 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 > * Fast picture update for VP8. Since VP8 uses a different RTCP packet number > than FIR, this really is specific to VP8 at this time. Ideally this would be > more generic and flexible for user preferences and other video codecs, but > that could be done at a latter date. > > The only part of this work that I did was port over the fast picture update > code to chan_pjsip. I *think* that chan_pjsip will still suck out the > attributes in res_pjsip_sdp_rtp, but I could be mistaken (Josh?) > > > Diffs > ----- > > /trunk/res/res_rtp_asterisk.c 397424 > /trunk/res/res_pjsip_sdp_rtp.c 397424 > /trunk/res/res_format_attr_opus.c PRE-CREATION > /trunk/main/rtp_engine.c 397424 > /trunk/main/frame.c 397424 > /trunk/main/format.c 397424 > /trunk/main/channel.c 397424 > /trunk/include/asterisk/opus.h PRE-CREATION > /trunk/include/asterisk/format.h 397424 > /trunk/channels/chan_sip.c 397424 > /trunk/channels/chan_pjsip.c 397424 > > Diff: https://reviewboard.asterisk.org/r/2723/diff/ > > > Testing > ------- > > > Thanks, > > Matt Jordan > >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev