[Freeswitch-users] L16 Codec in mod_conference

[Dragos Oancea]

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