Improving voice quality
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
How about using physical hardware? 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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John Covici covici@ccs.covici.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Are you sure what you are hearing is not just the noise gate? (squelch in old school radio terms) As far as audio quality goes it should blow hangouts away.... Of course as with anything running on commodity shared virtual server hosting your mileage will vary. On 5/12/15, 2:44 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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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-- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH
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@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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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"for example, there are small cut-offs every now and then" Have you tried setting "conference-flags" to "audio-always" in conference.conf.xml?
Bob has a good point. Not sure about the flag but I do have energy-level set to 0 in my conference.conf.xml <param name="energy-level" value="0"/> On Tue, May 12, 2015 at 3:29 PM, Bob Hartwig <bobjectsfreeswitch@gmail.com> wrote:
"for example, there are small cut-offs every now and then"
Have you tried setting "conference-flags" to "audio-always" in conference.conf.xml?
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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Yes, it's a good point, energy level does produce choppiness, but we are using 0 for some time now, we also tried audio-always and it's pretty much the same. In addition, we are using iSAC@16000h since it gave the least choppiness compared to OPUS/others. On 13 May 2015 at 03:15, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Bob has a good point. Not sure about the flag but I do have energy-level set to 0 in my conference.conf.xml
<param name="energy-level" value="0"/>
On Tue, May 12, 2015 at 3:29 PM, Bob Hartwig <bobjectsfreeswitch@gmail.com
wrote:
"for example, there are small cut-offs every now and then"
Have you tried setting "conference-flags" to "audio-always" in conference.conf.xml?
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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Here is the conference config in case there are some low hanging fruits in this area: <profile name="test"> <param name="domain" value="$${domain}"/> <param name="rate" value="16000"/> <param name="interval" value="30"/> <param name="energy-level" value="0"/> <param name="muted-sound" value="conference/conf-muted.wav"/> <param name="unmuted-sound" value="conference/conf-unmuted.wav"/> <param name="moh-sound" value="$${hold_music}"/> <param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/> <param name="exit-sound" value="tone_stream://%(500,0,300,200,100,50,25)"/> <param name="kicked-sound" value="conference/conf-kicked.wav"/> <param name="locked-sound" value="conference/conf-locked.wav"/> <param name="is-locked-sound" value="conference/conf-is-locked.wav"/> <param name="is-unlocked-sound" value="conference/conf-is-unlocked.wav"/> <param name="pin-sound" value="conference/conf-pin.wav"/> <param name="bad-pin-sound" value="conference/conf-bad-pin.wav"/> <param name="caller-id-name" value="$${outbound_caller_name}"/> <param name="caller-id-number" value="$${outbound_caller_id}"/> <param name="comfort-noise" value="0"/> <param name="conference-flags" value="audio-always"/> </profile> On 13 May 2015 at 12:33, Adam Ben-Ayoun <adam.ben.ayoun1@gmail.com> wrote:
Yes, it's a good point, energy level does produce choppiness, but we are using 0 for some time now, we also tried audio-always and it's pretty much the same. In addition, we are using iSAC@16000h since it gave the least choppiness compared to OPUS/others.
On 13 May 2015 at 03:15, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Bob has a good point. Not sure about the flag but I do have energy-level set to 0 in my conference.conf.xml
<param name="energy-level" value="0"/>
On Tue, May 12, 2015 at 3:29 PM, Bob Hartwig < bobjectsfreeswitch@gmail.com> wrote:
"for example, there are small cut-offs every now and then"
Have you tried setting "conference-flags" to "audio-always" in conference.conf.xml?
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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participants (5)
-
Adam Ben-Ayoun
-
Bob Hartwig
-
covici＠ccs.covici.com
-
Ken Rice
-
Oleg Stolyar