Hi guys,

We are using Freeswitch as a audio "MCU" for WebRTC using mod_conference, we are currently using mobile clients on both Android and iOS. The voice quality is good but not as good as Hangouts for example, there are small cut-offs every now and then that affects the overall experience. We are trying to understand the reason for that, maybe it's Freeswitch mixing algorithm, our servers infrastructure, or something else. We did tried Janus audio conference demo and it was very good (we used chrome on Android which should use pretty much the same stack). We really need help figuring this out. I want to first make sure it's not our infrastructure (mainly network/CPU). We are using virtualized m3.large on AWS (2 vCPUs and 7.5GB RAM). Maybe there's a certain instance type on a certain provider that is known to give the best quality? Any ideas on how to go about this?

Thanks,

Adam

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