[Freeswitch-users] Improving voice quality

[Adam Ben-Ayoun]

Oleg, we did try Chrome on PC in the past, indeed it gave better results. I
will retry again soon.

On 12 May 2015 at 21:04, Oleg Stolyar <olegstolyar at 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 at 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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>
>
> _________________________________________________________________________
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> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
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> http://confluence.freeswitch.org
> http://www.cluecon.com
>
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