[Freeswitch-users] L16 Codec in mod_conference

[Mike Jerris]

What rate is your conference running at?

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