No, thats the codec rate. This will be in your conference configuration.
Perhaps 48000, from uuid_dump:
variable_rtp_use_codec_rate: 48000
Regards.
What rate is your conference running at?
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
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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