[Freeswitch-users] L16 Codec in mod_conference

[王聡]

We hadn’t configure this part, perhaps default? Sorry I’m newcomer to mod_conference.

Regards.

> 在 2019年11月15日，03:26，Mike Jerris <mike at freeswitch.org> 写道：
> 
> No, thats the codec rate.  This will be in your conference configuration.
> 
>> On Nov 14, 2019, at 1:42 AM, 王聡 <cong.wang.itsherpa at gmail.com <mailto:cong.wang.itsherpa at 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 at freeswitch.org <mailto:mike at freeswitch.org>> 写道：
>>> 
>>> What rate is your conference running at?
>>> 
>>>> On Nov 12, 2019, at 6:49 PM, 王聡 <cong.wang.itsherpa at gmail.com <mailto: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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