We are using the OPUS codec with Chrome and WebRTC to make SIP calls into FS conferences. We have packet captures at the edge of network that sound clear, but when they get to FreeSWITCH they sound distorted via delays and clipping. I've tried several combinations of jitterbuffer settings, disabled/enabling RTP timers, conference. I have a mechanism I use in which I replay the RTP packets using SIPP and try them out. Is there a way to dynamically adjust the jitterbuffer so that the right size is being used? Right now I have it set at 180:540:540. Maybe there is some other setting I could try that would help? Perhaps there's a way to force OPUS to use less bandwidth at the expense of quality? I'm willing to share the PCAPs some examples. Thanks in advance. --Danny Gershman