[asterisk-bugs] [JIRA] (ASTERISK-24423) High pitched constant sound in ConfBridge

[Thomas Frederiksen (JIRA)]

Thomas Frederiksen created ASTERISK-24423:
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             Summary: High pitched constant sound in ConfBridge
                 Key: ASTERISK-24423
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24423
             Project: Asterisk
          Issue Type: Bug
      Security Level: None
          Components: Applications/app_confbridge
    Affects Versions: 11.11.0
         Environment: CentOS 6
            Reporter: Thomas Frederiksen


For a audio broadcasting we have Confbridge configured like below.
ConfTransmit is the sip-client that will be transmitting audio to the listeners.
ConfMuted is all the listeners that can listen to the audio from ConfTransmit:

[Conf]
type=bridge
internal_sample_rate=16000

[ConfTransmit]
type=user
marked=yes
dsp_drop_silence=yes

[ConfMuted]
type=user
startmuted=yes
wait_marked=yes
dsp_drop_silence=yes
quiet=yes

When using these settings together with G.722 SIP-clients in both ends, the "ConfTransmit" user sometimes hears a high pitched (sounds like around 8kHz) constant sound from the conference.
The issue always happens if the ConfMuted user is the first to dialin to the conference. Then when the ConfTransmit user dials in after the ConfMuted user the high pitched sound is heard in the ConfTransmits phone constantly.

The problem only exists if both SIP-users are running G.722 codec.
The high pitched sound disappears if i disable dsp_drop_silence, but then a burst of white noise is heard every 5 seconds or so.
The problem also exists if "internal_sample_rate=auto" but disappears completely if i set the conference to "internal_sample_rate=8000" and dsp_drop_silence is on.

The very strange thing is that i cannot hear any audio quality difference when ConfBridge i set to 8000 instead of 16000. It still sounds like 16kHz G.722

To my understanding if the internal_sample_rate in ConfBridge is set to 8000 the audio should sound like 8kHz audio or G.711, is this correct?
Why doesn't this have any effect on the audio quality?

I have tried with both X-Lite and Linphone SIP-phones to eliminate if the problem was with one of the SIP-clients.

I was also wondering if this problem still exists in Asterisk 12 or 13, or it has been fixed on these versions?



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