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. 
A conference is more CPU intensive than a bridge.

On Thu, Nov 14, 2019 at 9:43 AM 王聡 <cong.wang.itsherpa@gmail.com> wrote:
Perhaps 48000, from uuid_dump:

variable_rtp_use_codec_rate: 48000

Regards.

在 2019年11月14日，04:25，Mike Jerris <mike@freeswitch.org> 写道：

What rate is your conference running at?

On Nov 12, 2019, at 6:49 PM, 王聡 <cong.wang.itsherpa@gmail.com> wrote:

Hey all,

I’m trying to transfer my ivr system from bridge to conference for more function. The simple conference ivr likes:

session:execute("pre_answer")
session:execute("conference_set_auto_outcall", "user/" .. args.call_user)
session:execute("conference", "testroom@default")

My FreeSWITCH server is configured to accept only opus for audio codec, and it worked well on bridge mode. The uuid_dump showed:

variable_rtp_use_codec_string: OPUS,H264,H263-1998,VP8
variable_rtp_use_codec_name: opus
variable_rtp_use_codec_fmtp: useinbandfec%3D1
variable_rtp_use_codec_rate: 48000
variable_rtp_use_codec_ptime: 20
variable_rtp_use_codec_channels: 1
variable_rtp_last_audio_codec_string: opus%4048000h%4020i%401c
variable_original_read_codec: opus
variable_write_codec: opus
variable_read_codec: opus

But after I modified my ivr into conference, the read_codec turned into L16. Uuid_dump showed:

variable_rtp_use_codec_string: OPUS,H264,H263-1998,VP8
variable_rtp_use_codec_name: opus
variable_rtp_use_codec_fmtp: useinbandfec%3D1
variable_rtp_use_codec_rate: 48000
variable_rtp_use_codec_ptime: 20
variable_rtp_use_codec_channels: 1
variable_rtp_last_audio_codec_string: opus%4048000h%4020i%401c
variable_original_read_codec: opus
variable_write_codec: opus
variable_read_codec: L16

Both tests are based on Linphone offical app. 
During test, the audio quality has a significant loss on conference mode compared with bridge, perhaps due to L16 codec.
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.
Any suggestion would be appreciated.

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_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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