<html><head><meta http-equiv="Content-Type" content="text/html; charset=gb2312"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class="">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 style="margin: 0 0 0 40px; border: none; padding: 0px;" class=""><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 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_string: OPUS,H264,H263-1998,VP8</div></div><div class=""><div style="margin: 0px; font-stretch: normal; line-height: normal;" class="">variable_rtp_use_codec_name: opus</div></div><div class=""><div style="margin: 0px; font-stretch: normal; line-height: normal;" class="">variable_rtp_use_codec_fmtp: useinbandfec%3D1</div></div><div class=""><div style="margin: 0px; font-stretch: normal; line-height: normal;" class="">variable_rtp_use_codec_rate: 48000</div></div><div class=""><div style="margin: 0px; font-stretch: normal; line-height: normal;" class="">variable_rtp_use_codec_ptime: 20</div></div><div class=""><div style="margin: 0px; font-stretch: normal; line-height: normal;" class="">variable_rtp_use_codec_channels: 1</div></div><div class=""><div style="margin: 0px; font-stretch: normal; line-height: normal;" class="">variable_rtp_last_audio_codec_string: opus%4048000h%4020i%401c</div></div><div class=""><div style="margin: 0px; font-stretch: normal; line-height: normal;" class="">variable_original_read_codec: opus</div></div><div class=""><div style="margin: 0px; font-stretch: normal; line-height: normal;" class="">variable_write_codec: opus</div></div><div class=""><div style="margin: 0px; font-stretch: normal; line-height: normal;" class="">variable_read_codec: opus</div></div></blockquote><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="">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;" class=""><br class=""></div><blockquote style="margin: 0 0 0 40px; border: none; padding: 0px;" class=""><div style="margin: 0px; font-stretch: normal; line-height: normal;" class="">variable_rtp_use_codec_string: OPUS,H264,H263-1998,VP8</div><div style="margin: 0px; font-stretch: normal; line-height: normal;" class=""><div style="margin: 0px; font-stretch: normal; line-height: normal;" class="">variable_rtp_use_codec_name: opus</div></div><div style="margin: 0px; font-stretch: normal; line-height: normal;" class=""><div style="margin: 0px; font-stretch: normal; line-height: normal;" class="">variable_rtp_use_codec_fmtp: useinbandfec%3D1</div></div><div style="margin: 0px; font-stretch: normal; line-height: normal;" class=""><div style="margin: 0px; font-stretch: normal; line-height: normal;" class="">variable_rtp_use_codec_rate: 48000</div></div><div style="margin: 0px; font-stretch: normal; line-height: normal;" class=""><div style="margin: 0px; font-stretch: normal; line-height: normal;" class="">variable_rtp_use_codec_ptime: 20</div></div><div style="margin: 0px; font-stretch: normal; line-height: normal;" class=""><div style="margin: 0px; font-stretch: normal; line-height: normal;" class="">variable_rtp_use_codec_channels: 1</div></div><div style="margin: 0px; font-stretch: normal; line-height: normal;" class=""><div style="margin: 0px; font-stretch: normal; line-height: normal;" class="">variable_rtp_last_audio_codec_string: opus%4048000h%4020i%401c</div></div><div style="margin: 0px; font-stretch: normal; line-height: normal;" class=""><div style="margin: 0px; font-stretch: normal; line-height: normal;" class="">variable_original_read_codec: opus</div></div><div style="margin: 0px; font-stretch: normal; line-height: normal;" class=""><div style="margin: 0px; font-stretch: normal; line-height: normal;" class="">variable_write_codec: opus</div></div><div style="margin: 0px; font-stretch: normal; line-height: normal;" class=""><div style="margin: 0px; font-stretch: normal; line-height: normal;" class="">variable_read_codec: <font color="#ff2600" class="">L16</font></div></div></blockquote><div style="margin: 0px; font-stretch: normal; line-height: normal;" 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="">Both tests are based on Linphone offical app. </div><div style="margin: 0px; font-stretch: normal; line-height: normal;" class="">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;" class="">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;" class="">Any suggestion would be appreciated.</div><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 style="margin: 0px; font-stretch: normal; line-height: normal;" class="">C.Wang</div></div></body></html>