[asterisk-users] ConfBridge audio issues (chan_sip)

[Dan Cropp]

We have a customer using ConfBridges.
Party A is connected, audio is fine.
We originate a call to party B through an Avaya switch.  It forwards the call to IVR.
The two channels are added to the same ConfBridge.

Using a wireshark capture, I can listen to the audio for both channels.

Initially, everything on the audio sounds great.
Audio for the connection to the Avaya switch always sounds fine.
Audio that party A hears becomes garbled about 7-8 seconds into the prompts playing from the IVR.

I can definitely hear the audio from voice mail plays one prompt when it's fine.  Then, it changes to a different voice/prompt that is louder.  At that point, the audio party A hears is garbled.

This is using chan_sip for both channels.

We have no problems with audio calls to other numbers, only to the IVR

Only reason we are using chan_sip is because we need to use REFERs and are working on a patch submission (hoping asterisk 16.5.0) to resolve a PJSIP REFER issue.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20190605/e8051d15/attachment.html>