<html><head><meta http-equiv="Content-Type" content="text/html; charset=utf-8"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class="">We hadn’t configure this part, perhaps default? Sorry I’m newcomer to mod_conference.<div class=""><br class=""></div><div class="">Regards.<br class=""><div><br class=""><blockquote type="cite" class=""><div class="">在 2019年11月15日，03:26，Mike Jerris <<a href="mailto:mike@freeswitch.org" class="">mike@freeswitch.org</a>> 写道：</div><br class="Apple-interchange-newline"><div class=""><meta http-equiv="Content-Type" content="text/html; charset=utf-8" class=""><div style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class="">No, thats the codec rate. This will be in your conference configuration.<br class=""><div class=""><br class=""><blockquote type="cite" class=""><div class="">On Nov 14, 2019, at 1:42 AM, 王聡 <<a href="mailto:cong.wang.itsherpa@gmail.com" class="">cong.wang.itsherpa@gmail.com</a>> wrote:</div><br class="Apple-interchange-newline"><div class=""><meta http-equiv="Content-Type" content="text/html; charset=utf-8" class=""><div style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class="">Perhaps 48000, from uuid_dump:<div class=""><br class=""></div><blockquote style="margin: 0 0 0 40px; border: none; padding: 0px;" class=""><div class=""><div style="margin: 0px; font-stretch: normal; line-height: normal;" class="">variable_rtp_use_codec_rate: 48000</div></div></blockquote><div class=""><div style="margin: 0px; font-stretch: normal; line-height: normal;" class=""><br class=""></div><div style="margin: 0px; font-stretch: normal; line-height: normal;" class="">Regards.</div><div class=""><br class=""><blockquote type="cite" class=""><div class="">在 2019年11月14日，04:25，Mike Jerris <<a href="mailto:mike@freeswitch.org" class="">mike@freeswitch.org</a>> 写道：</div><br class="Apple-interchange-newline"><div class=""><span style="caret-color: rgb(0, 0, 0); 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; -webkit-text-stroke-width: 0px; text-decoration: none; float: none; display: inline !important;" class="">What rate is your conference running at?</span><br class="" style="caret-color: rgb(0, 0, 0); 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; -webkit-text-stroke-width: 0px; text-decoration: none;"><div style="caret-color: rgb(0, 0, 0); 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; -webkit-text-stroke-width: 0px; text-decoration: none;" class=""><br class=""><blockquote type="cite" class=""><div class="">On Nov 12, 2019, at 6:49 PM, 王聡 <<a href="mailto:cong.wang.itsherpa@gmail.com" class="">cong.wang.itsherpa@gmail.com</a>> wrote:</div><br class="Apple-interchange-newline"><div class=""><div class="" style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;">Hey all,<div class=""><br class=""></div><div class="">I’m trying to transfer my ivr system from bridge to conference for more function. The simple conference ivr likes:</div><div class=""><br class=""></div><blockquote class="" style="margin: 0px 0px 0px 40px; border: none; padding: 0px;"><div class=""><div class="">session:execute("pre_answer")</div></div><div class=""><div class="">session:execute("conference_set_auto_outcall", "user/" .. args.call_user)</div></div><div class=""><div class="">session:execute("conference", "testroom@default")</div></div></blockquote><div class=""><br class=""></div><div class="">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 class=""><br class=""></div><blockquote class="" style="margin: 0px 0px 0px 40px; border: none; padding: 0px;"><div class=""><div class="" style="margin: 0px; font-stretch: normal; line-height: normal;">variable_rtp_use_codec_string: OPUS,H264,H263-1998,VP8</div></div><div class=""><div class="" style="margin: 0px; font-stretch: normal; line-height: normal;">variable_rtp_use_codec_name: opus</div></div><div class=""><div class="" style="margin: 0px; font-stretch: normal; line-height: normal;">variable_rtp_use_codec_fmtp: useinbandfec%3D1</div></div><div class=""><div class="" style="margin: 0px; font-stretch: normal; line-height: normal;">variable_rtp_use_codec_rate: 48000</div></div><div class=""><div class="" style="margin: 0px; font-stretch: normal; line-height: normal;">variable_rtp_use_codec_ptime: 20</div></div><div class=""><div class="" style="margin: 0px; font-stretch: normal; line-height: normal;">variable_rtp_use_codec_channels: 1</div></div><div class=""><div class="" style="margin: 0px; font-stretch: normal; line-height: normal;">variable_rtp_last_audio_codec_string: opus%4048000h%4020i%401c</div></div><div class=""><div class="" style="margin: 0px; font-stretch: normal; line-height: normal;">variable_original_read_codec: opus</div></div><div class=""><div class="" style="margin: 0px; font-stretch: normal; line-height: normal;">variable_write_codec: opus</div></div><div class=""><div class="" style="margin: 0px; font-stretch: normal; line-height: normal;">variable_read_codec: opus</div></div></blockquote><div class="" style="margin: 0px; font-stretch: normal; line-height: normal;"><br class=""></div><div class="" 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 class="" style="margin: 0px; font-stretch: normal; line-height: normal;"><br class=""></div><blockquote class="" style="margin: 0px 0px 0px 40px; border: none; padding: 0px;"><div class="" style="margin: 0px; font-stretch: normal; line-height: normal;">variable_rtp_use_codec_string: OPUS,H264,H263-1998,VP8</div><div class="" style="margin: 0px; font-stretch: normal; line-height: normal;"><div class="" style="margin: 0px; font-stretch: normal; line-height: normal;">variable_rtp_use_codec_name: opus</div></div><div class="" style="margin: 0px; font-stretch: normal; line-height: normal;"><div class="" style="margin: 0px; font-stretch: normal; line-height: normal;">variable_rtp_use_codec_fmtp: useinbandfec%3D1</div></div><div class="" style="margin: 0px; font-stretch: normal; line-height: normal;"><div class="" style="margin: 0px; font-stretch: normal; line-height: normal;">variable_rtp_use_codec_rate: 48000</div></div><div class="" style="margin: 0px; font-stretch: normal; line-height: normal;"><div class="" style="margin: 0px; font-stretch: normal; line-height: normal;">variable_rtp_use_codec_ptime: 20</div></div><div class="" style="margin: 0px; font-stretch: normal; line-height: normal;"><div class="" style="margin: 0px; font-stretch: normal; line-height: normal;">variable_rtp_use_codec_channels: 1</div></div><div class="" style="margin: 0px; font-stretch: normal; line-height: normal;"><div class="" style="margin: 0px; font-stretch: normal; line-height: normal;">variable_rtp_last_audio_codec_string: opus%4048000h%4020i%401c</div></div><div class="" style="margin: 0px; font-stretch: normal; line-height: normal;"><div class="" style="margin: 0px; font-stretch: normal; line-height: normal;">variable_original_read_codec: opus</div></div><div class="" style="margin: 0px; font-stretch: normal; line-height: normal;"><div class="" style="margin: 0px; font-stretch: normal; line-height: normal;">variable_write_codec: opus</div></div><div class="" style="margin: 0px; font-stretch: normal; line-height: normal;"><div class="" style="margin: 0px; font-stretch: normal; line-height: normal;">variable_read_codec:<span class="Apple-converted-space"> </span><font color="#ff2600" class="">L16</font></div></div></blockquote><div class="" style="margin: 0px; font-stretch: normal; line-height: normal;"><div class="" style="margin: 0px; font-stretch: normal; line-height: normal;"><br class=""></div><div class="" style="margin: 0px; font-stretch: normal; line-height: normal;">Both tests are based on Linphone offical app. </div><div class="" 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 class="" 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 class="" style="margin: 0px; font-stretch: normal; line-height: normal;">Any suggestion would be appreciated.</div></div></div></div></blockquote></div><br class="" style="caret-color: rgb(0, 0, 0); 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; -webkit-text-stroke-width: 0px; text-decoration: none;"></div></blockquote></div></div></div></div></blockquote></div><br class=""></div>_________________________________________________________________________<br class=""><br class="">The FreeSWITCH project is sponsored by SignalWire <a href="https://signalwire.com" class="">https://signalwire.com</a><br class="">Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.<br class="">Build your next product on our scalable cloud platform.<br class=""><br class="">Join our online community to chat in real time <a href="https://signalwire.community" class="">https://signalwire.community</a><br class=""><br class="">Professional FreeSWITCH Services<br class=""><a href="mailto:sales@freeswitch.com" class="">sales@freeswitch.com</a><br class="">https://freeswitch.com<br class=""><br class="">Official FreeSWITCH Sites<br class="">https://freeswitch.com/oss<br class="">https://freeswitch.org/confluence<br class="">https://cluecon.com<br class=""><br class="">FreeSWITCH-users mailing list<br class="">FreeSWITCH-users@lists.freeswitch.org<br class="">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br class="">UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br class="">https://freeswitch.com</div></blockquote></div><br class=""></div></body></html>