<div dir="ltr">Your issue does not have anything to do with L16 per se, and you cannot disable it for mod_conference. Everything is decoded into L16 / encoded from L16, like all audio codecs do. <div>A conference is more CPU intensive than a bridge.</div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Thu, Nov 14, 2019 at 9:43 AM 王聡 <<a href="mailto:cong.wang.itsherpa@gmail.com">cong.wang.itsherpa@gmail.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div style="overflow-wrap: break-word;">Perhaps 48000, from uuid_dump:<div><br></div><blockquote style="margin:0px 0px 0px 40px;border:none;padding:0px"><div><div style="margin:0px;font-stretch:normal;line-height:normal">variable_rtp_use_codec_rate: 48000</div></div></blockquote><div><div style="margin:0px;font-stretch:normal;line-height:normal"><br></div><div style="margin:0px;font-stretch:normal;line-height:normal">Regards.</div><div><br><blockquote type="cite"><div>在 2019年11月14日，04:25，Mike Jerris <<a href="mailto:mike@freeswitch.org" target="_blank">mike@freeswitch.org</a>> 写道：</div><br><div><span style="font-family:Helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;text-decoration:none;float:none;display:inline">What rate is your conference running at?</span><br style="font-family:Helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;text-decoration:none"><div style="font-family:Helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;text-decoration:none"><br><blockquote type="cite"><div>On Nov 12, 2019, at 6:49 PM, 王聡 <<a href="mailto:cong.wang.itsherpa@gmail.com" target="_blank">cong.wang.itsherpa@gmail.com</a>> wrote:</div><br><div><div style="overflow-wrap: break-word;">Hey all,<div><br></div><div>I’m trying to transfer my ivr system from bridge to conference for more function. The simple conference ivr likes:</div><div><br></div><blockquote style="margin:0px 0px 0px 40px;border:none;padding:0px"><div><div>session:execute("pre_answer")</div></div><div><div>session:execute("conference_set_auto_outcall", "user/" .. args.call_user)</div></div><div><div>session:execute("conference", "testroom@default")</div></div></blockquote><div><br></div><div>My FreeSWITCH server is configured to accept only opus for audio codec, and it worked well on bridge mode. The uuid_dump showed:</div><div><br></div><blockquote style="margin:0px 0px 0px 40px;border:none;padding:0px"><div><div style="margin:0px;font-stretch:normal;line-height:normal">variable_rtp_use_codec_string: OPUS,H264,H263-1998,VP8</div></div><div><div style="margin:0px;font-stretch:normal;line-height:normal">variable_rtp_use_codec_name: opus</div></div><div><div style="margin:0px;font-stretch:normal;line-height:normal">variable_rtp_use_codec_fmtp: useinbandfec%3D1</div></div><div><div style="margin:0px;font-stretch:normal;line-height:normal">variable_rtp_use_codec_rate: 48000</div></div><div><div style="margin:0px;font-stretch:normal;line-height:normal">variable_rtp_use_codec_ptime: 20</div></div><div><div style="margin:0px;font-stretch:normal;line-height:normal">variable_rtp_use_codec_channels: 1</div></div><div><div style="margin:0px;font-stretch:normal;line-height:normal">variable_rtp_last_audio_codec_string: opus%4048000h%4020i%401c</div></div><div><div style="margin:0px;font-stretch:normal;line-height:normal">variable_original_read_codec: opus</div></div><div><div style="margin:0px;font-stretch:normal;line-height:normal">variable_write_codec: opus</div></div><div><div style="margin:0px;font-stretch:normal;line-height:normal">variable_read_codec: opus</div></div></blockquote><div style="margin:0px;font-stretch:normal;line-height:normal"><br></div><div style="margin:0px;font-stretch:normal;line-height:normal">But after I modified my ivr into conference, the read_codec turned into L16. Uuid_dump showed:</div><div style="margin:0px;font-stretch:normal;line-height:normal"><br></div><blockquote style="margin:0px 0px 0px 40px;border:none;padding:0px"><div style="margin:0px;font-stretch:normal;line-height:normal">variable_rtp_use_codec_string: OPUS,H264,H263-1998,VP8</div><div style="margin:0px;font-stretch:normal;line-height:normal"><div style="margin:0px;font-stretch:normal;line-height:normal">variable_rtp_use_codec_name: opus</div></div><div style="margin:0px;font-stretch:normal;line-height:normal"><div style="margin:0px;font-stretch:normal;line-height:normal">variable_rtp_use_codec_fmtp: useinbandfec%3D1</div></div><div style="margin:0px;font-stretch:normal;line-height:normal"><div style="margin:0px;font-stretch:normal;line-height:normal">variable_rtp_use_codec_rate: 48000</div></div><div style="margin:0px;font-stretch:normal;line-height:normal"><div style="margin:0px;font-stretch:normal;line-height:normal">variable_rtp_use_codec_ptime: 20</div></div><div style="margin:0px;font-stretch:normal;line-height:normal"><div style="margin:0px;font-stretch:normal;line-height:normal">variable_rtp_use_codec_channels: 1</div></div><div style="margin:0px;font-stretch:normal;line-height:normal"><div style="margin:0px;font-stretch:normal;line-height:normal">variable_rtp_last_audio_codec_string: opus%4048000h%4020i%401c</div></div><div style="margin:0px;font-stretch:normal;line-height:normal"><div style="margin:0px;font-stretch:normal;line-height:normal">variable_original_read_codec: opus</div></div><div style="margin:0px;font-stretch:normal;line-height:normal"><div style="margin:0px;font-stretch:normal;line-height:normal">variable_write_codec: opus</div></div><div style="margin:0px;font-stretch:normal;line-height:normal"><div style="margin:0px;font-stretch:normal;line-height:normal">variable_read_codec:<span> </span><font color="#ff2600">L16</font></div></div></blockquote><div style="margin:0px;font-stretch:normal;line-height:normal"><div style="margin:0px;font-stretch:normal;line-height:normal"><br></div><div style="margin:0px;font-stretch:normal;line-height:normal">Both tests are based on Linphone offical app. </div><div style="margin:0px;font-stretch:normal;line-height:normal">During test, the audio quality has a significant loss on conference mode compared with bridge, perhaps due to L16 codec.</div><div style="margin:0px;font-stretch:normal;line-height:normal">Is there any solutions for this situation? I wonder if I can disable L16 on my server, or force the codec to opus in conference.</div><div style="margin:0px;font-stretch:normal;line-height:normal">Any suggestion would be appreciated.</div></div></div></div></blockquote></div><br style="font-family:Helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;text-decoration:none"><span style="font-family:Helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;text-decoration:none;float:none;display:inline">_________________________________________________________________________</span><br style="font-family:Helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;text-decoration:none"><br style="font-family:Helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;text-decoration:none"><span style="font-family:Helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;text-decoration:none;float:none;display:inline">The FreeSWITCH project is sponsored by SignalWire<span> </span></span><a href="https://signalwire.com/" style="font-family:Helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px" target="_blank">https://signalwire.com</a><br style="font-family:Helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;text-decoration:none"><span style="font-family:Helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;text-decoration:none;float:none;display:inline">Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.</span><br style="font-family:Helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;text-decoration:none"><span style="font-family:Helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;text-decoration:none;float:none;display:inline">Build your next product on our scalable cloud platform.</span><br style="font-family:Helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;lett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