[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:comment-tabpanel&focusedCommentId=222750#comment-222750 ] 

Rusty Newton commented on ASTERISK-24314:
-----------------------------------------

All of your config files have windows newline characters mixed in. The dos2unix tool is not converting them properly (not cleaning them up completely), for whatever reason. However, that scares me from using them for reproduction. 

Can you please attach the actual files used on your server (in unix file format) to avoid any strangeness with differing newline characters?



> 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: Rusty Newton
>         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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