Nothing special except maybe I am using PCMU instead of Opus. I am using Ubuntu 14 now but it also worked fine with CentOS 5.9. Just for my own education - are you seeing the same issues with non-mobile clients? On Tue, May 12, 2015 at 12:44 PM, Adam Ben-Ayoun <adam.ben.ayoun1@gmail.com> wrote:
We tried on c4.4xlarge with dedicated tenancy and it's pretty much the same. Any special config you applied to the machine? I am using CentOS 6.
On 12 May 2015 at 21:04, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Have you tried it with non-mobile WebRTC clients? Is there the same problem?
I am using mod_conference on FS hosted on AWS m3.xlarge instance and I have not noticed audio problem. But my WebRTC clients are not mobile but rather Chrome on PCs.
On Tue, May 12, 2015 at 9:59 AM, Adam Ben-Ayoun < adam.ben.ayoun1@gmail.com> wrote:
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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