[asterisk-users] ConfBridge audio issues

[Dan Cropp]

We have a system where two calls are in a ConfBridge with recording.  This is Asterisk 16.3.0

Channel A seems to work perfectly.  Wireshark is showing the RTP to/from working fine and having no jitter/lag issues.  This call hears everything from channel B.

Channel B we have more issues capturing a wireshark trace because their channel can be in the system for hours.
When the two calls are in the ConfBridge, Channel B is the first to speak.  Everything seems perfectly fine.  Channel A hears it well and ConfBridge recording sounds good.
Then, channel B replies.  Audio from channel B seems fine in wireshark (no jitter/lag).  However, the ConfBrdge recording and channel B indicate garbled audio.

This only happens for the first couple seconds channel B talks.
After that, everything seems to be perfectly fine.


For each channel added to the ConfBridge, the user profile has...
jitterbuffer = yes
denoise = no
dsp_drop_silence = yes
dsp_silence_threshold = 2500
dsp_talking_threshold = 160

On the bridge profile.
internal_sample_rate = 0
mixing_interval = 20
jitterbuffer is not being set.  According to the wiki, this defaults to no
binaural_active is not being set.  According to the wiki, this defaults to no


One other possible coincidence in the samples I have received, channel B seems to always start talking roughly 2500 ms into the ConfBrdge.  Could this static audio be occurring due to the dsp_drop_silence and the dsp_silence_threshold hitting at 2500 ms?

Does anyone have any suggestions?

Dan
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