On Thu, Oct 14, 2021 at 2:37 PM Pascal Cadotte <p...@wazo.io> wrote: > > Hello everyone, > > We've been trying to improve the quality of our video conferences using > confbridge. We've been able to figure out how to get the video usable for all > participants even for users using bad internet connections using the REMB > configuration options. > > However, we are still having problems getting decent audio when there's > packet loss. We think this is because PLC is not used for channels in > confbridge. We found that information in this page > https://wiki.asterisk.org/wiki/display/AST/PLC+Restrictions+and+Caveats "In > addition, MeetMe and ConfBridge calls will not use PLC." > > We are looking into ways to improve this situation and would like to gather > some information about this restriction before working on it. > > 1. Does anyone know why PLC has not been implemented for meetme and > confbridge? > 2. Is there a known workaround to allow a channel to have PLC enabled while > being in confbridge? > 3. Is it a problem many people have? I've only seen this question that seems > to be related > https://community.asterisk.org/t/jitterbuffer-plc-fec-etc-how-do-i-know-they-are-working/85146 > 4. Is there something in progress that we could contribute to? > > The tests have been made using the opus codec, PJSIP on a WSS transport on > Asterisk 18.6.0. > > Given two users Alice and Bob > Given a 10% paquet loss on Alice > When Alice and Bob are talking through confbridge Bob hears a lot of cracking > When Alice and Bob are talking to each other using the Dial application to > call Bob's phone, the call quality is almost flawless
I'll take a quick stab at an answer here. I went looking at translate.c, plc.c, and channel.c, and here are some notes that I wrote as I was trying to remember how this works: The generic plc code in Asterisk was written a long time ago and can be only used for 8 KHz SLINEAR, not for other sample rates. I'd presume that this means a codec such as OPUS needs to output to 8KHz SLINEAR somehow to even start using the generic PLC code. It appears that generic plc is actually calling the concealment functions when making a write to an ast_channel (so after a frame is produced from a mixing bridge presumably and goes out to a channel driver) - not on the reception side, reading a frame from a channel. (channel.c) Unless someone else wants to go look at and correct me, I would assume that that means concealment comes after a bridge mixes frames together and presumably all input information about packet loss is wiped clean (meaning no generic PLC is happening with a mixing bridge). Looking at the codecs that support native PLC (such iLBC, speex, etc) it looks like they implement this on the conversion to SLINEAR (and correspondingly on an ast_read on the channel in quesiton), which should work for a case where you have a confbridge or meetme involved. For Sangoma's codec_opus, we have support for its native PLC and FEC functionality also, so I'd make sure that you have fec=yes enabled for its entry in codecs.conf. I'd also make sure you have jitterbuffer enabled on any channels/conferences in question for good measure. So, to sum it up: If you're using OPUS, use the opus native fec option (I think PLC should be happening also automatically as well) and make sure that the jitterbuffer is enabled on the channels/conferences in question. If you'd like to use a codec with non-native PLC, you're going to need to figure out how to get the generic PLC code working on the ast_read() instead of the ast_write() to the channel in question. (hoping my assumptions about some of this is correct too :-) ) Hope that helps, Matthew Fredrickson -- _____________________________________________________________________ -- 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
