[Freeswitch-users] L16 Codec in mod_conference

王聡 cong.wang.itsherpa at gmail.com
Thu Nov 14 09:42:23 UTC 2019


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> 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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