[asterisk-users] ConfBridge audio issues
dan at amtelco.com
Mon Aug 5 12:54:50 CDT 2019
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?
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