On Mar 13, 2021, at 4:26 AM, Dragos Oancea <dragos@freeswitch.org> wrote:When trying to understand how it works make sure you are not mixing decoding side settings with the encoding side settings.packet-loss-percent is encoding side and is the initial value for loss if you use adjust-bitrate (callback with SCC_AUDIO_PACKET_LOSS from the core) .The callback comes only if there is incoming RTCP - which for audio is typically at 5 seconds interval.So the adjust-bitrate feature will update the loss too, not only the bitrate.Otherwise, without adjust-birate, it's the default value for OPUS_SET_PACKET_LOSS_PERC() which will stay like this along the call and depending on the preset bitrate it will generate a certain amount of FEC on top of the payload of some of the packets (encoder decisions) . So no, you should not set that very high, since you can't know the real network conditions before the call is made. 15 or 20 are good values._________________________________________________________________________On Fri, Mar 12, 2021 at 9:05 PM Brian West <brian@freeswitch.com> wrote:CPU and bandwidth./bOn Fri, Mar 12, 2021 at 12:40 PM Terry C <infinite3219@gmail.com> wrote:Hi,_________________________________________________________________________We use FreeSwitch 1.10.5 with Opus audio for clients. We target voice quality to be wideband. The config settings are below. We also set a jitter buffer in the dialplan.Is there a downside to increasing the packet loss percent to 40-50% other than using more bandwidth? Is there a voice quality impact if you set it too high (what would be considered too high)?Thanks.<paramname="use-vbr"value="1"/><paramname="use-dtx"value="1"/><paramname="complexity"value="10"/><paramname="packet-loss-percent"value="30"/><paramname="keep-fec-enabled"value="1"/><paramname="use-jb-lookahead"value="1"/><paramname="maxaveragebitrate"value="40000"/><paramname="maxplaybackrate"value="16000"/><paramname="sprop-maxcapturerate"value="16000"/><paramname="adjust-bitrate"value="1"/></settings>
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