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[infinite3219 at gmail.com]

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.


<param name="use-vbr" value="1"/>
<param name="use-dtx" value="1"/>
<param name="complexity" value="10"/>
<param name="packet-loss-percent" value="30"/>
<param name="keep-fec-enabled" value="1"/>
<param name="use-jb-lookahead" value="1"/>
<param name="maxaveragebitrate" value="40000"/>
<param name="maxplaybackrate" value="16000"/>
<param name="sprop-maxcapturerate" value="16000"/>
<param name="adjust-bitrate" value="1"/>
</settings>

[brian at freeswitch.com]

CPU and bandwidth.

/b




On Fri, Mar 12, 2021 at 12:40 PM Terry C <infinite3219@gmail.com (infinite3219@gmail.com)> wrote:

[dragos at freeswitch.org]

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 (brian@freeswitch.com)> wrote:

[mario_fs at mgtech.com]

Could it be related to this which is outstanding?, I can duplicate it every time:https://github.com/signalwire/freeswitch/issues/963
Opus warbly, drops sections or unintelligible audio due to gaps in RTP timestamps