[asterisk-bugs] [JIRA] (ASTERISK-24314) ConfBridge Doesn't Deliver 48 kHz Audio

[Rusty Newton (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-24314?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Rusty Newton updated ASTERISK-24314:
------------------------------------

    Assignee: Frankie Chin  (was: Rusty Newton)
      Status: Waiting for Feedback  (was: Triage)

> ConfBridge Doesn't Deliver 48 kHz Audio
> ---------------------------------------
>
>                 Key: ASTERISK-24314
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24314
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Applications/app_confbridge
>    Affects Versions: 12.4.0
>            Reporter: Frankie Chin
>            Assignee: Frankie Chin
>         Attachments: confbridge_serverA.conf, extensions_serverA.conf, extensions_serverB.conf, full_serverA, sip_serverA.conf, sip_serverB.conf
>
>
> I have two servers registered to each other via SIP. I only enabled "slin48" codec in sip.conf of both servers. 
> Scenario 1 (Happy):
> I use AMI to originate a call to Server B. Once Server B answers the call, Server A will start playing a 48 kHz speech audio from a *.sln48 file. At Server B, the audio is recorded into another *.sln48 file. The recorded audio quality at Server B is basically the same as the original source.
> Scenario 2:
> Using another AMI application, it originates a call to Server B and puts it into a conference hosted in Server A. It then originates another call to a local channel in Server A and puts the local channel into the conference as well. The local channel then starts playing the same speech audio from the source *.sln48 file into the conference. The audio is also recorded at Server B. But this time, the recorded audio quality is much worse than the source audio.
> The following are the settings in my confbridge.conf which I think relevant:
> - internal_sample_rate = 48000
> - mixing interval = 20
> - dsp_drop_silence = yes
> - dsp_talking_threshold = 128
> - dsp_silence_threshold = 2000
> I have even tried setting the internal_sample_rate to 192000 but it didn't improve the recorded audio quality. My final objective is to be able to put multiple servers into a conference and get a local channel in one server to play the 48 kHz audio out to all the other servers.



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