Could someone post a 1.5-2 minutes recording of them talking (or playing an audiobook) via BBB conference?

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Alex

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Oct 24, 2019, 12:06:44 PM10/24/19
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I can't get rid of these sounds(@ ~30+sec)




1) It's on any device, any browser, any network

2) It's on any BBB installation (the official demo.bigbluebutton.org, distancelearning.cloud/demo-bigbluebutton-for-yourself/, my server).

3) There is no jitter or packet loss accorrding to

sudo tshark -q -f 'udp portrange 16384-32768' -o rtp.heuristic_rtp:TRUE -z rtp,streams

4) there is no issue with the internal clock or anything

5) I uploaded a wav file and played it DIRECTLY VIA FREESWITCH into a conference to all participants  using

fs_cli ->conference list-> conference XXXX play /tmp/test.wav

NO issues. The sound was very clear.

6) both manual and auto-installation script was tested


At this point unless I hear a clear recording of some of you guys I won't believe it's on my side (and what "side"? see #1 and 2)  especially taking into account all the things I posted above.

Thank you.

sd...@distancelearning.cloud

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Oct 25, 2019, 9:42:45 AM10/25/19
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Is it in the READ file of presenter?

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Alex

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Oct 29, 2019, 7:23:26 AM10/29/19
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Hello.

IWhen I enable this

"

Add this to bbb_conference.xml right before <action application="conference" data="$1@cdquality"/>

 

<action application="set" data="RECORD_WRITE_ONLY=true"/>

<action application="record_session" data="/var/freeswitch/meetings/${destination_number}_${strftime(%Y-%m-%d-%H-%M-%S)}_WRITE_${caller_id_name}.wav"/>

<action application="set" data="RECORD_READ_ONLY=true"/>

<action application="record_session" data="/var/freeswitch/meetings/${destination_number}_${strftime(%Y-%m-%d-%H-%M-%S)}_READ_${caller_id_name}.wav"/>"


And there is only one user (presenter) in a conference , it creates only one file with 874384883268324-something name


When  a user joins another two files get created. READ and WRITE. READ file is silent. WRITE and 874384883268324-something audio files contain actual audio. Both have these sound issues.

sd...@distancelearning.cloud

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Oct 29, 2019, 11:48:38 AM10/29/19
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Sounds like you presenter is entering through the dialplan somewhere else if I had to guess.

 

Regards,

Stephen

 

Sent: Tuesday, October 29, 2019 7:22 AM
To: BigBlueButton-Setup <bigbluebu...@googlegroups.com>
Subject: Re: [bigbluebutton-setup] Could someone post a 1.5-2 minutes recording of them talking (or playing an audiobook) via BBB conference?

 

Hello.

 

IWhen I enable this

 

"

Add this to bbb_conference.xml right before <action application="conference" data="$1@cdquality"/>

 

<action application="set" data="RECORD_WRITE_ONLY=true"/>

<action application="record_session" data="/var/freeswitch/meetings/${destination_number}_${strftime(%Y-%m-%d-%H-%M-%S)}_WRITE_${caller_id_name}.wav"/>

<action application="set" data="RECORD_READ_ONLY=true"/>

<action application="record_session" data="/var/freeswitch/meetings/${destination_number}_${strftime(%Y-%m-%d-%H-%M-%S)}_READ_${caller_id_name}.wav"/>"

 

And there is only one user (presenter) in a conference , it creates only one file with 874384883268324-something name

 

When  a user joins another two files get created. READ and WRITE. READ file is silent.

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Alex

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Oct 29, 2019, 12:09:54 PM10/29/19
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Well, it's the default manual\and auto-installation...



And I'm still gonna ask everyone here... Can anyone actually produce a clear audio recording without these sounds?

On Tuesday, October 29, 2019 at 6:48:38 PM UTC+3, HostBBB.com wrote:

Sounds like you presenter is entering through the dialplan somewhere else if I had to guess.

 

Regards,

Stephen

 

From: bigbluebu...@googlegroups.com <bigbluebu...@googlegroups.com> On Behalf Of Alex
Sent: Tuesday, October 29, 2019 7:22 AM
To: BigBlueButton-Setup <bigbluebu...@googlegroups.com>
Subject: Re: [bigbluebutton-setup] Could someone post a 1.5-2 minutes recording of them talking (or playing an audiobook) via BBB conference?

 

Hello.

 

IWhen I enable this

 

"

Add this to bbb_conference.xml right before <action application="conference" data="$1@cdquality"/>

 

<action application="set" data="RECORD_WRITE_ONLY=true"/>

<action application="record_session" data="/var/freeswitch/meetings/${destination_number}_${strftime(%Y-%m-%d-%H-%M-%S)}_WRITE_${caller_id_name}.wav"/>

<action application="set" data="RECORD_READ_ONLY=true"/>

<action application="record_session" data="/var/freeswitch/meetings/${destination_number}_${strftime(%Y-%m-%d-%H-%M-%S)}_READ_${caller_id_name}.wav"/>"

 

And there is only one user (presenter) in a conference , it creates only one file with 874384883268324-something name

 

When  a user joins another two files get created. READ and WRITE. READ file is silent.

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Richard Alam

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Oct 29, 2019, 2:26:40 PM10/29/19
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Hi Alex,

On Tue, Oct 29, 2019 at 12:09 PM Alex <fedotov...@gmail.com> wrote:
Well, it's the default manual\and auto-installation...



And I'm still gonna ask everyone here... Can anyone actually produce a clear audio recording without these sounds?

I created a 2 minute recording on demo and the sounds are there. It is not perfect but clear enough.

Richard
 

On Tuesday, October 29, 2019 at 6:48:38 PM UTC+3, HostBBB.com wrote:

Sounds like you presenter is entering through the dialplan somewhere else if I had to guess.

 

Regards,

Stephen

 

From: bigbluebu...@googlegroups.com <bigbluebu...@googlegroups.com> On Behalf Of Alex
Sent: Tuesday, October 29, 2019 7:22 AM
To: BigBlueButton-Setup <bigbluebu...@googlegroups.com>
Subject: Re: [bigbluebutton-setup] Could someone post a 1.5-2 minutes recording of them talking (or playing an audiobook) via BBB conference?

 

Hello.

 

IWhen I enable this

 

"

Add this to bbb_conference.xml right before <action application="conference" data="$1@cdquality"/>

 

<action application="set" data="RECORD_WRITE_ONLY=true"/>

<action application="record_session" data="/var/freeswitch/meetings/${destination_number}_${strftime(%Y-%m-%d-%H-%M-%S)}_WRITE_${caller_id_name}.wav"/>

<action application="set" data="RECORD_READ_ONLY=true"/>

<action application="record_session" data="/var/freeswitch/meetings/${destination_number}_${strftime(%Y-%m-%d-%H-%M-%S)}_READ_${caller_id_name}.wav"/>"

 

And there is only one user (presenter) in a conference , it creates only one file with 874384883268324-something name

 

When  a user joins another two files get created. READ and WRITE. READ file is silent.

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Alex

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Oct 30, 2019, 7:30:13 AM10/30/19
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Thank you Richard!



Hey Fred,

Is it possible that there might be something wrong with the way BBB uses Freeswitch? Something in configs?

I've  ruled out jitter, packet loss and freeswitch itself (transmitting an audio file from the server directly to a conference via freeswitch produces no sound issues.

For now we have to stream via youtube , start our webinar in a listen only mode and just share the streaming video..

Cheers!
Hi Alex,

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Alex

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Oct 30, 2019, 8:42:06 AM10/30/19
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And another question..

Is there any chance this will be tested and looked into?

It looks like I'm not the only one experiencing this issue.

Fred Dixon

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Oct 30, 2019, 9:31:30 AM10/30/19
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Hi Alex,

> It looks like I'm not the only one experiencing this issue.

Everyone is experiencing having minor gaps in the audio in the recordings.  Just as you found earlier, it's on our demo server as well: you can record audio and there will be minor gaps in the audio (sometimes they are very brief, but audible).  Sometimes you'll have minutes of clear audio, and other times there will be minor gaps between words, sometimes at the very beginning of when a speaker talks. .

The questions you are asking is (a) why are they occurring and (b) how to prevent them?

The degree of these disruptions will differ depending on a one (or more) -- the user's computer, the user's bandwidth, the existence of a firewall between the user and the BigBlueButton server, the network latency/dropped packets between the user's network and the BigBlueButton server, the BigBlueButton server bandwidth's, the CPU load on the BigBlueButton server.

To narrow down (a), one idea would be to try setting up a BigBlueButton server local on your network and do a recording -- this will remove all possibilities of network latency/dropped packets.  We would be interested to hear if you get an audio recording with less gaps.


Regards,... Fred

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Richard Alam

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Oct 30, 2019, 10:00:36 AM10/30/19
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Hi Alex,

Perhaps you are experiencing this.


In your server, try editing  /opt/freeswitch/conf/autoload_configs/conference.conf.xml and set energy-level to a low value.

    <profile name="cdquality">
      <param name="domain" value="$${domain}"/>
      <param name="rate" value="48000"/>
      <param name="interval" value="20"/>
      <param name="energy-level" value="100"/>

HTH.

Richard


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Alex

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Oct 30, 2019, 10:03:37 AM10/30/19
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Not only recording. Listening live too.

>the user's computer, the user's bandwidth, the existence of a firewall between the user and the BigBlueButton server, the network latency/dropped packets between the user's network and the BigBlueButton server, the BigBlueButton server bandwidth's, the CPU load on the BigBlueButton server.

>the user's computer,

Any user computer and device including smartphones

>the user's bandwidth

100 mbit and 750Mbit aren't enough?

>the existence of a firewall between the user and the BigBlueButton server

Isn't turn server supposed to deal with it?

> the network latency/dropped packets between the user's network and the BigBlueButton server,

Minimal if any at all. No dropped packers or jitter observed when conducting these tests

>the BigBlueButton server bandwidth's

100Mbit and 1Gbit should be enough for 2 users (presenter and a guest)

>the CPU load on the BigBlueButton server.

Minimal for 2 users on an 8 cores CPU(dedicated), 8GB ram

We are not having issues with Zoom.us or with youtube streaming,

What;s even more important.

As I've said we __directly_played an audio file from our server into a conference to all participants, No issues. If any of the things you've mentioned were in play it would have affected it  and it would have had the same sound issues, wouldn't it?

Alex

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Oct 30, 2019, 10:10:26 AM10/30/19
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Hello Richard.

Thank you for the suggestion.

This is the first I do after I install BBB on any server.  We always set energy level to 0 to disable it completely and reduce any gaps. Needless to say it doesn't help much. 0 might be a bit better than 100 but the difference is minimal.
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Fred Dixon

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Oct 30, 2019, 11:57:48 AM10/30/19
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Hi Alex,

> I can't get rid of these sounds(@ ~30+sec) https://drive.google.com/file/d/1XOIsofU2nD5vfZC8F7khGeIXV4KVIDBI/view

I can hear gaps in the audio.

> As I've said we __directly_played an audio file from our server into a conference to all participants, No issues. If any of the things you've mentioned were in play it would have affected it  and it would have had the same sound issues, wouldn't it?

It should, but anytime you think that everything is as it should, but it still is not working, then there is at least one assumption that is invalid.  The question is which one?

Is it an option for you to try recording on a local BigBlueButton server (to your LAN).  It should remove any potential issues of the network (one of the assumptions).

If you you still hear the gaps --> it is not the network (we haven't figure out yet, but we've narrowed it down)
If you don't hear the gaps -> it is very likely the network (same as above, no solution yet -- but we can focus on the differences in networking)

Regards,.. Fred



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Alex

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Oct 30, 2019, 1:00:51 PM10/30/19
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1) Why are there no gaps on other services(zoom, youtube, discord, steam,skype you name it). Even Jitsi (a far inferior to BBB system)that  uses webrtc didn't have them. it had other issues , but not this one.


2) >any potential issues of the network

Which network? 

- The server's network? We tried it on..3 different networks\data centers. Each server had minimal ping, little (less than 1%)  jitter and packet loss

Don't forget we tried the official bbb demo server and the Stephen's server.  


- The presenter's network?  I think we tried doing it using 4 or more different ISPs.


Let's assume it's not the network.  What's next?

PS Your server is in Canada, right?  And you live in Canada. Can you, personally, get an audio without these sounds? Very easy to heck it out. Run a webinar as a presenter from one laptop, join from another one., record the live audio stream with OBS or whatever.

Sincerely yours,
Alex
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Richard Alam

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Oct 31, 2019, 11:07:02 AM10/31/19
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On Wed, Oct 30, 2019 at 1:00 PM Alex <fedotov...@gmail.com> wrote:
1) Why are there no gaps on other services(zoom, youtube, discord, steam,skype you name it). Even Jitsi (a far inferior to BBB system)that  uses webrtc didn't have them. it had other issues , but not this one.


2) >any potential issues of the network

Which network? 

- The server's network? We tried it on..3 different networks\data centers. Each server had minimal ping, little (less than 1%)  jitter and packet loss

Don't forget we tried the official bbb demo server and the Stephen's server.  


- The presenter's network?  I think we tried doing it using 4 or more different ISPs.


Let's assume it's not the network.  What's next?

Audio packets are sent directly from the browser to FreeSWITCH. If it's not the network, then it's something inside FreeSWITCH. We'll have to get the FS folks involved in that. We just used FS out-of-box and modify configs.

You mentioned that in your setup there is no or very minimal jitter. Can you try again and collect call stats from FS to see what it thinks of the call quality?

To get the uuids of the channels

/opt/freeswitch/bin/fs_cli', '-x', 'show channels as json

To get the stats, change the uuid to your channel.

/opt/freeswitch/bin/fs_cli -x 'json {"command": "mediaStats", "data": {"uuid": "bd3b2d9f-7cf8-4e9e-86dd-8f88305f2b48" }}'
 
Richard

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Alex

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Nov 1, 2019, 4:22:38 AM11/1/19
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then it's something inside FreeSWITCH

We directly played a wav filestored on the server  through Freeswitch into a conference. No issues.

Alex

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Nov 1, 2019, 4:33:04 AM11/1/19
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2 people in a  conference. A presenter and a guest


/opt/freeswitch/bin# /opt/freeswitch/bin/fs_cli -x 'json {"command": "mediaStats", "data": {"uuid": "ece1b196-7b9a-48e3-9313-d7928b169f4d" }}'

{"command":"mediaStats","data":{"uuid":"ece1b196-7b9a-48e3-9313-d7928b169f4d"},"status":"success","response":{"audio":{"in_raw_bytes":6733826,"in_media_bytes":6733727,"in_packet_count":73677,"in_media_packet_count":73676,"in_skip_packet_count":1037,"in_jitter_packet_count":0,"in_dtmf_packet_count":0,"in_cng_packet_count":0,"in_flush_packet_count":1,"in_largest_jb_size":0,"in_jitter_min_variance":7.890340,"in_jitter_max_variance":110.500000,"in_jitter_loss_rate":0,"in_jitter_burst_rate":0,"in_mean_interval":20.000810,"in_flaw_total":3,"in_quality_percentage":99,"in_mos":4.492027,"out_raw_bytes":1987017,"out_media_bytes":1987017,"out_packet_count":74107,"out_media_packet_count":74107,"out_skip_packet_count":0,"out_dtmf_packet_count":0,"out_cng_packet_count":0,"rtcp_packet_count":74033,"rtcp_octet_count":5878200}}}


/opt/freeswitch/bin# /opt/freeswitch/bin/fs_cli -x 'json {"command": "mediaStats", "data": {"uuid": "d033a87e-2851-44de-a6cc-afcbdb987a17" }}'
{"command":"mediaStats","data":{"uuid":"d033a87e-2851-44de-a6cc-afcbdb987a17"},"status":"success","response":{"audio":{"in_raw_bytes":0,"in_media_bytes":0,"in_packet_count":0,"in_media_packet_count":0,"in_skip_packet_count":64121,"in_jitter_packet_count":0,"in_dtmf_packet_count":0,"in_cng_packet_count":0,"in_flush_packet_count":0,"in_largest_jb_size":0,"in_jitter_min_variance":0,"in_jitter_max_variance":0,"in_jitter_loss_rate":0,"in_jitter_burst_rate":0,"in_mean_interval":0,"in_flaw_total":0,"in_quality_percentage":100,"in_mos":4.500000,"out_raw_bytes":8914051,"out_media_bytes":8914051,"out_packet_count":64121,"out_media_packet_count":64121,"out_skip_packet_count":0,"out_dtmf_packet_count":0,"out_cng_packet_count":0,"rtcp_packet_count":0,"rtcp_octet_count":0}}}

Alex

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Nov 1, 2019, 5:00:00 AM11/1/19
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Src IP addr  Port    Dest IP addr  Port       SSRC          Payload  Pkts         Lost   Max Delta(ms)  Max Jitter(ms) Mean Jitter(ms) Problems?
   217.23.XX.XX 28744  85.143.164.218 55054 0x3936C535      Unknown(96)  2055     0 (0.0%)            0.00            0.00            0.00
 85.143.XXX.XXX 60314    217.23.XX.XX 25208 0x272CB6C1     Unknown(120)  2339     0 (0.0%)            0.00            0.00            0.00
   217.23.XX.XX 22350  85.143.XXX.XXX 25942 0x85C09081     Unknown(109) 10175     0 (0.0%)            0.00            0.00            0.00
 85.143.XXX.XXX 25942    217.23.XX.XX 22350 0xA29E117E     Unknown(109) 10176     0 (0.0%)            0.00     


RTP analyzed by tshark

0 packets lost, 0 jitter.
 

Alex

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Nov 1, 2019, 1:15:57 PM11/1/19
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We've tested today Nextcloud Talk.

It also uses WebRTC.

The sound was clear.

I'm sorry, but at this point,considering everything posted in this thread and audio clipping (or whatever we can call it) reproduced on the official BBB demo server by different parties, we should acknowledge that there is undeniably an issue with Freeswitch\BBB config. 

Where exactly the problem lies is for the devs to find out. Unfortunately I can't do it.

I believe that this should be on a priority list. Web conference is about audio quality first and foremost. Polling, chat, video sharing, whiteboard etc are all good and fine, but audio quality comes first.

I guess I won't be posting anymore\spamming forum with my ramblings on the matter =)  coz I believe enough was said.

Thank you all guys for taking your time and trying to help me out.

PS  My friend and I tested the official bbb demo server and the issue was much more noticeable there ( coz of energy level =100?)


Richard Alam

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Nov 1, 2019, 3:31:41 PM11/1/19
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On Fri, Nov 1, 2019 at 1:16 PM Alex <fedotov...@gmail.com> wrote:
We've tested today Nextcloud Talk.

It also uses WebRTC.

The sound was clear.

I'm sorry, but at this point,considering everything posted in this thread and audio clipping (or whatever we can call it) reproduced on the official BBB demo server by different parties, we should acknowledge that there is undeniably an issue with Freeswitch\BBB config. 

Where exactly the problem lies is for the devs to find out. Unfortunately I can't do it.

I've been spending the day investigating this issue.

I can reproduce the issue as you described on my local VM.

I play an audio book wav file on the server directly into the conference and the recorded wav file is clear.

I use playback the same audio book wav using VLC -> VoiceMeter and use VoiceMeter as input (instead of mic) to call into the conference and confirm that the recorded wav file has clippings.

I also captured pcap on the server but unfortunately cannot extract the audio out as it is encrypted.

We are trying to narrow down where these clippings are introduced but so far no success.

We can't play an opus file directly into the conference to see if the transcoding from opus to wav introduces the issue. There's also the question of the bitrate of the audio sent by the browser to the server.

We'll continue to work on it and hopefully will have some luck.

Richard


 

I believe that this should be on a priority list. Web conference is about audio quality first and foremost. Polling, chat, video sharing, whiteboard etc are all good and fine, but audio quality comes first.

I guess I won't be posting anymore\spamming forum with my ramblings on the matter =)  coz I believe enough was said.

Thank you all guys for taking your time and trying to help me out.

PS  My friend and I tested the official bbb demo server and the issue was much more noticeable there ( coz of energy level =100?)


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Alex

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Nov 5, 2019, 4:46:11 AM11/5/19
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Hey Richard.

Thanx a lot for looking into this.

on my local VM

VMs have their own issues when it comes to freeswitch tbh. This one though can be reproduced even on bare metal servers.

I really hope in time this will get fixed.

For now we have decided to stream audio\video via youtube and share a stream link right into a webinar.

Regards

Richard Alam

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Nov 6, 2019, 3:46:09 PM11/6/19
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Hi Alex,

We got some help from the freeswitch folks.

It looks like the opus.conf and conference.conf just needed some tuning. I did several recordings and they sounded good.

Can you please make the following modifications on your server?

Just two file need to be modified. 
/opt/freeswitch/conf/autoload_configs/conference.conf.xml
/opt/freeswitch/conf/autoload_configs/opus.conf.xml

For /opt/freeswitch/conf/autoload_configs/opus.conf.xml

<configuration name="opus.conf">
      <settings>
        <param name="use-vbr" value="1"/>
        <param name="use-dtx" value="0"/>
        <param name="complexity" value="5"/>
        <param name="maxaveragebitrate" value="32000"/>
        <param name="maxplaybackrate" value="8000"/>
        <param name="packet-loss-percent" value="15"/>
        <param name="keep-fec-enabled" value="1"/>
        <param name="use-jb-lookahead" value="1"/>
        <param name="advertise-useinbandfec" value="1"/>
      </settings>
</configuration>

For /opt/freeswitch/conf/autoload_configs/conference.conf.xml
The  energy-level and  comfort-noise values were changed.

    <profile name="cdquality">
      <param name="domain" value="$${domain}"/>
      <param name="rate" value="48000"/>
      <param name="interval" value="20"/>
      <param name="energy-level" value="200"/>
      <!-- <param name="sound-prefix" value="$${sounds_dir}/en/us/callie"/> -->
      <param name="muted-sound" value="conference/conf-muted.wav"/>
      <param name="unmuted-sound" value="conference/conf-unmuted.wav"/>
      <param name="alone-sound" value="conference/conf-alone.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="1400"/>
    </profile>

Please tell me if it is good too on your end so we can modify the default configs and include in the next release.

Thanks.

Richard




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Alex

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Nov 7, 2019, 11:32:26 AM11/7/19
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Hi Richard!

I'll try it tomorrow when I get to the office.

Regards,
Alex

Fred Dixon

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Nov 7, 2019, 11:42:22 AM11/7/19
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Hi Alex,

We're still doing some testing.  Please use the following values for opus.conf.xml

<configuration name="opus.conf">
      <settings>
        <param name="use-vbr" value="1"/>
        <param name="use-dtx" value="0"/>
        <param name="complexity" value="5"/>

        <param name="packet-loss-percent" value="15"/>
        <param name="keep-fec-enabled" value="1"/>
        <param name="use-jb-lookahead" value="1"/>
        <param name="advertise-useinbandfec" value="1"/>
      </settings>
</configuration>


Regards,... Fred

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Alex

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Nov 8, 2019, 7:55:57 AM11/8/19
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Hey Fred (and K)

So I've tested a bit with these settings:

<param name="comfort-noise" value="1400"/>
<param name="energy-level" value="200"/>

<configuration name="opus.conf">
      <settings>
        <param name="use-vbr" value="1"/>
        <param name="use-dtx" value="0"/>
        <param name="complexity" value="5"/>
        <param name="packet-loss-percent" value="15"/>
        <param name="keep-fec-enabled" value="1"/>
        <param name="use-jb-lookahead" value="1"/>
        <param name="advertise-useinbandfec" value="1"/>
      </settings>
</configuration>

And I dunno.. it might be a bit better, but i still hear these "popping' sounds


Regards,
Alex

Richard Alam

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Nov 8, 2019, 9:00:12 AM11/8/19
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Can you please try with the first setting with the maxaveragebitrate? We rmoved it because when we did not use audibook but just regular speech the audio was telephone quality. I think is is the bitrate 8k settings. Will have to do more tests today

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Richard Alam

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Nov 8, 2019, 5:35:17 PM11/8/19
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Hi,

I did more config tweaking and did some tests.

Here are recordings I did with the config below. 


Can you please try with the following config changes?

root@droplet-4689:/opt/freeswitch# cat conf/autoload_configs/opus.conf.xml

<configuration name="opus.conf">
      <settings>
        <param name="use-vbr" value="1"/>
        <param name="use-dtx" value="0"/>
        <param name="complexity" value="5"/>
        <param name="packet-loss-percent" value="10"/>

        <param name="keep-fec-enabled" value="1"/>
        <param name="use-jb-lookahead" value="1"/>
        <param name="maxaveragebitrate" value="30000"/>
        <param name="maxplaybackrate" value="48000"/>
        <param name="sprop-maxcapturerate" value="48000"/>
        <param name="adjust-bitrate" value="1"/>
      </settings>
</configuration>


root@droplet-4689:/opt/freeswitch# cat conf/autoload_configs/conference.conf.xml

      <param name="energy-level" value="200"/>
      <param name="comfort-noise" value="1400"/>

root@droplet-4689:/opt/freeswitch# cat conf/dialplan/default/bbb_echo_to_conference.xml
<include>
  <extension name="ECHO_TO_CONFERENCE">
    <condition field="${bbb_from_echo}" expression="true" break="on-false"/>
    <condition field="destination_number" expression="^(ECHO_TO_CONFERENCE)$">
        <action application="set" data="rtp_jitter_buffer_plc=true"/>
        <action application="set" data="suppress_cng=true"/>
        <action application="set" data="jitterbuffer_msec=20:400"/>
        <action application="set" data="dtmf_type=info"/>
        <action application="send_dtmf" data="7"/>
        <action application="conference" data="${vbridge}@cdquality"/>
    </condition>
  </extension>
</include>

Please let me know if this is good enough for your use case. Tried to find opus settings from Jitsi so I could try it out but could not find them.

Richard

Alex

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Nov 9, 2019, 8:17:06 AM11/9/19
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Hey Richard,

May I ask why u have changed echo conference settings? Don't they affect only self-test?.

I listened to the files in your archive.

1) The recorded files 183f0bf3a0982a127bdb8161e0c44eb696b3e75c-1573248389295-xxxx have these "sounds"

2) The 3min recording actually sounds pretty good


Have you tried checking out how it sounds when you join as a listener (and recording it as a listener using,say, obs(audio only capture)?

Unfortunately I can't test how it sounds for listeners until Monday, but here is a file from var/freeswitch

I might be mistaken but it seems to sound better with the new settings. The "sounds" are still there though, maybe not as noticeable as before..

PS I don't think Jitsi uses freeswitch hence no opus.conf.xml. I'll ask on their forums where does it keep webrtc\opus settings.Hopefully someone will respond.

Regards,
Alex
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Richard Alam

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Nov 11, 2019, 10:42:31 AM11/11/19
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Hi Alex,



On Sat, Nov 9, 2019 at 8:17 AM Alex <fedotov...@gmail.com> wrote:
Hey Richard,

May I ask why u have changed echo conference settings? Don't they affect only self-test?.

The conf/dialplan/default/bbb_echo_to_conference.xml dialplan is where the user is put into the conference after echo test.

I tried to make changes to the settings based on this document.

 

I listened to the files in your archive.

1) The recorded files 183f0bf3a0982a127bdb8161e0c44eb696b3e75c-1573248389295-xxxx have these "sounds"

Agreed. Everything I tried doesn't remove these sounds completely.
 

2) The 3min recording actually sounds pretty good

These are the source audio files I play so they don't have the sounds.
 


Have you tried checking out how it sounds when you join as a listener (and recording it as a listener using,say, obs(audio only capture)?

No, I haven't. What I tried is to bridge the call to another FS server so I can to packet capture as the packets leave the first FS. I then extracted
the raw audio and can hear the clipping sounds there.

Unfortunately, we can't decode the packets when they leave the browser and into the server as they are encrypted. That is why I had to bridge the call
to get unencrypted packets.

Richard
 
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Alex

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Nov 11, 2019, 1:35:08 PM11/11/19
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Hi.

> we can't decode the packets when they leave the browser and into the server as they are encrypted

I'm not sure if it is relevant but you can decrypt SSL traffic in Wireshark



>Everything I tried doesn't remove these sounds completely.

 ;(

So,I guess this is it then? The end of the road so to speak....

PS I've tried listening live as a guest today and it was all the same as the saved audio file in var/freeswitch(presenter only)

Sincerely yours,
Alex
Hi Alex,



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Fred Dixon

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Nov 11, 2019, 2:00:13 PM11/11/19
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Hi Alex,

> ;(
> So,I guess this is it then? The end of the road so to speak....

Sorry we couldn't remove all the small gaps in the audio.  But it's not the end of the road.  FreeSWITCH, the internet, the browsers, and our knowledge of how it all works is always improving. 

Thanks for pushing a bit on this, as it brought us back to take a closer look at the configuration files we were using in FreeSWITCH.  If the audio quality is not good enough for you in BigBlueButton, that's OK too.  You can always check back with us periodically to see how the project has progressed. 

Regards,... Fred


Hi Alex,



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Alex

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Nov 12, 2019, 3:45:34 AM11/12/19
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Hello Fred.

May I ask what freeswitch config changes will make it into the upcoming release of BBB (if it is planned) ? The ones Richard posted a bit earlier or something different? 

Regards,
Alex
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Fred Dixon

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Nov 12, 2019, 7:37:21 AM11/12/19
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Hi Alex,

Here's the default settings in BigBlueButton 2.2-rc-1

# cat /opt/freeswitch/conf/autoload_configs/opus.conf.xml

<configuration name="opus.conf">
      <settings>
        <param name="use-vbr" value="1"/>
        <param name="use-dtx" value="0"/>
        <param name="complexity" value="5"/>

        <param name="packet-loss-percent" value="15"/>
        <param name="keep-fec-enabled" value="1"/>
        <param name="use-jb-lookahead" value="1"/>
        <param name="advertise-useinbandfec" value="1"/>
      </settings>
</configuration>

Regards,... Fred

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Alex

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Nov 13, 2019, 9:11:43 AM11/13/19
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Hi

We've restored back these settings and made a test run with some students who were not aware of the issue and they said it got kinda irritating in 1 hour or so, and they still had study for 3-4  hours , so we switched back to adding youtube stream into the webinar.

In short we are essentially left with 3 choices:

1) use BBB + youtube(OBS) which is really inconvenient and we'll have to assign a person who would  babysit every teacher coz this setup is too complicated for them.(sometimes we have to change youtube's live streaming channels for different lectures, so it could get messy really fast if someone screws up)

2) Look for other self-hosted solutions and unfortunately  there is simply no alternative to BBB in terms of what it offers. OpenMeeting is just too dated and heavy on resources for no reason at all.And it's interface is atrocious.  Nextcloud talk works fine but it's not a webinar..It's more of a skype alternative and lacks lots of features. Jitsi's sound is fine, but it is MUCH more CPU heavy/per user and for some reason webcams get laggy and sometimes we get plagued by "user x is experiencing a connectivity issue" and judging by their forum we are not alone. Lots of threads on this problem.

And..this is it.. I don't think there is something else available. If there is, I'd be obliged if you could point me in the right direction.

3) Get a paid service (zoom etc).BBB's interface is better in terms of being user friendly(everything in one place and visible at once- chat, user li\its etc. + unfortunately we have found out that  the upper management is not that happy with the  zoom and K's  pricing considering all their limitations in regard to the number of presenters and viewers.We work for a non-private federal company and they are very, VERY stingy.

Anyways, I hope this issue will not be forgotten and, perhaps, in time, you'll find out a fix.

Peace out.

Alex

Fred Dixon

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Nov 13, 2019, 9:57:43 AM11/13/19
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Hi Alex,

> BBB's interface is better in terms of being user friendly(everything in one place and visible at once- chat, user li\its etc. +

Thanks for the good words.  Our focus is on education (online learning), and our focus is to give the instructor lots of tools to engage students and have the interface "fade into the background" as the instructor and students get on with the job of learning.

We're not giving up on improving the audio.  We'll be revisiting this issue in the coming months and working with yourself and others in the forum to better understand where these random clicks/pops are coming from in the audio.

For the moment, we're primarily focused on finishing BigBlueButton 2.2.

As an aside, the FreeSWITCH project recently raised over 11m in investment, so they (SignalWire) are pushing hard to accelerate development as well.  We use FreeSWITCH as the core for handling all audio in BigBlueButton.

Regards,... Fred

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Alex

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Nov 15, 2019, 2:54:17 PM11/15/19
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Hi Richard.

Try these settings

conference.conf.xml
<param name="interval" value="120"/>
<param name="energy-level" value="0"/>
<param name="comfort-noise" value="1400"/>

opus.conf..xml

<configuration name="opus.conf">
      <settings>
        <param name="use-vbr" value="1"/>
        <param name="use-dtx" value="0"/>
        <param name="complexity" value="5"/>
        <param name="packet-loss-percent" value="15"/>
        <param name="keep-fec-enabled" value="1"/>
        <param name="use-jb-lookahead" value="1"/>
        <param name="advertise-useinbandfec" value="1"/>
      </settings>
</configuration>


Do recordings\live audio have less clipping? I'm not sure about the interval setting though..120 is absolute maximum  and they say it could degrade audio quality.

Regards.

On Monday, November 11, 2019 at 6:42:31 PM UTC+3, Richard Alam wrote:
Hi Alex,



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Alex

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Nov 27, 2019, 1:12:23 PM11/27/19
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Hey Fred.

I was wondering whether this issue is posted on github so that it won't get lost on the forum ;)

If it is, could you share a link,please, so that I could bookmark it and check its status occasionally.

Cheers,

Alex
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Fred Dixon

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Nov 27, 2019, 1:38:48 PM11/27/19
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Hi Alex,

On the settings that you posted, did you hear an audible improvement in the audio capture on your setup?

Regards,... Fred

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Alex

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Nov 27, 2019, 3:53:54 PM11/27/19
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Hi.

It seems that with these settings audio clipping doesn't happen as often as it used to but it is still there.

Besides, I'm not sure it is "safe" to use such a high interval value (120 is maximum). The devs even warn that it might degrade audio quality.

I suspect it's just a positive side effect, nothing more.

Yours,

Alex
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Fred Dixon

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Nov 27, 2019, 4:11:57 PM11/27/19
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Hi Alex,

Looking at the docs


which states

  Higher numbers require less CPU but can degrade conversation quality, so experimentation with your setup is best.

But in your testing your finding this improves audio quality?  Did you do a test where you increased just that value and the audio quality improved?


Regards,... Fred


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Alex

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Nov 28, 2019, 4:37:20 AM11/28/19
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But in your testing your finding this improves audio quality?

hey Fred.

I think they are talking about the general audio quality. For instance, set <param name="complexity" value to 1 instead of 5(10). You will notice that the audio sounds noticeably worse than at higher values.

In our case, the problem is not audio quality but these strange audio noises\clipping sounds this whole thread is about. And considering that it was confirmed the issue is universal I would think it deserves to be posted on github so it won't get lost on the forums ;)

Did you do a test where you increased just that value and the audio quality improved?

I haven't done it. I can't test it today coz the server is in use.  I could try it this weekend.



Regards,

Alex

Fred Dixon

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Nov 28, 2019, 7:11:58 AM11/28/19
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Hi Alex,

Here's an existing issue where you can add your comments and track our progress


We had updated this issue about two weeks ago with a reference link to this discussion in bigbluebutton-setup.

Regards,... Fred


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Alex

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Mar 25, 2020, 6:55:16 AM3/25/20
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Good day.

Fred, any news on the issue??

Sincerely yours,

Alex

On Thursday, November 28, 2019 at 3:11:58 PM UTC+3, Fred Dixon wrote:
Hi Alex,

Here's an existing issue where you can add your comments and track our progress


We had updated this issue about two weeks ago with a reference link to this discussion in bigbluebutton-setup.

Regards,... Fred


On Thu, Nov 28, 2019 at 4:37 AM Alex <fedotov...@gmail.com> wrote:
But in your testing your finding this improves audio quality?

hey Fred.

I think they are talking about the general audio quality. For instance, set <param name="complexity" value to 1 instead of 5(10). You will notice that the audio sounds noticeably worse than at higher values.

In our case, the problem is not audio quality but these strange audio noises\clipping sounds this whole thread is about. And considering that it was confirmed the issue is universal I would think it deserves to be posted on github so it won't get lost on the forums ;)

Did you do a test where you increased just that value and the audio quality improved?

I haven't done it. I can't test it today coz the server is in use.  I could try it this weekend.



Regards,

Alex

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Richard Alam

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Mar 25, 2020, 7:57:08 AM3/25/20
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Hi Alex,

On Wed, Mar 25, 2020 at 6:55 AM Alex <fedotov...@gmail.com> wrote:
Good day.

Fred, any news on the issue??

No progress. 

We're very busy with keeping things running due to increase in load as result of covid.

Richard
 

Sincerely yours,

Alex

On Thursday, November 28, 2019 at 3:11:58 PM UTC+3, Fred Dixon wrote:
Hi Alex,

Here's an existing issue where you can add your comments and track our progress


We had updated this issue about two weeks ago with a reference link to this discussion in bigbluebutton-setup.

Regards,... Fred


On Thu, Nov 28, 2019 at 4:37 AM Alex <fedotov...@gmail.com> wrote:
But in your testing your finding this improves audio quality?

hey Fred.

I think they are talking about the general audio quality. For instance, set <param name="complexity" value to 1 instead of 5(10). You will notice that the audio sounds noticeably worse than at higher values.

In our case, the problem is not audio quality but these strange audio noises\clipping sounds this whole thread is about. And considering that it was confirmed the issue is universal I would think it deserves to be posted on github so it won't get lost on the forums ;)

Did you do a test where you increased just that value and the audio quality improved?

I haven't done it. I can't test it today coz the server is in use.  I could try it this weekend.



Regards,

Alex

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Alex

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Mar 29, 2020, 6:56:02 AM3/29/20
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Hey Richard.


We used almost the same config with small changes

https://pastebin.com/8Fs9qG0w

The sound is MUCH better and we can actually start using bbb coz I gotta be honest, it was hard to get through a lecture (not even a long one) with these constant crackly sounds.  You need to start looking into these settings  when you are done with whatever it is you lot are doing now ;)

Stay healthy ;)


Regards

Alex
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Richard Alam

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Mar 29, 2020, 8:04:04 AM3/29/20
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On Sun, Mar 29, 2020 at 6:56 AM Alex <fedotov...@gmail.com> wrote:
Hey Richard.


We used almost the same config with small changes

https://pastebin.com/8Fs9qG0w

The sound is MUCH better and we can actually start using bbb coz I gotta be honest, it was hard to get through a lecture (not even a long one) with these constant crackly sounds.  You need to start looking into these settings  when you are done with whatever it is you lot are doing now ;)

Stay healthy ;)

Thanks!

Please post the changes to an issue. If not, please create one. 

Will try to get into the next build if the changes are only config changes. 

Great that the community is helping each other out.

Richard

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Kenren Taisho

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Apr 1, 2020, 2:55:45 AM4/1/20
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Alex,
I tried the changes that you posted. And I feel it does make a difference in the audio quality.
Thanks for sharing this!

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Martin Thomas Schrott

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Apr 2, 2020, 9:40:49 AM4/2/20
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richard, Alex,


we tested your changes - DO NOT PUSH THIS INTO THE REPO!!!


the sound quality defenitely  is much better and I would love to get
this, because all crackling noises are gone.

But after 10 - 20 min video and audio gets out of sync!

We tested this on all our servers today.


We switched back to the original settings and everything is back in sync
again.


Do you have any idea which of your changes could have removed the
crackling noises? I tried to just use

* energy-level = 0

this is good - but does not remove noise.


* set complexity to 10 - does not improve audio and does also not remove
noise.


* disabled vbr - does not help and not remove noise.


So no idea which of your changes does the magic trick to get clear audio
without the crackling noises

maybe disabling fec ...

and no idea what is the cause for the sync problem.


Hope there is a solution to get the noises removed but still keep video
and audio in sync...


Cheers

Martin


Alex

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Apr 2, 2020, 10:11:23 AM4/2/20
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Hey Martin!

You are right, these settings don't seem to affect crackling noises.   I think it's jitter buffer settings in bbb_conference.conf.xml and bbb_echo_to_conference.conf.xml

And we also started experiencing out of sync issues, I've noticed it just today. But tbh.. I'd take a slight out of syn issue over constant crackling noises  any time. =)

Martin Thomas Schrott

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Apr 2, 2020, 10:17:24 AM4/2/20
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...the out of sync gets more and more

after 20 min there was a gap of 2 seconds


thanks for your quick response - we will try to just use the jb changes instead of all...

best

Martin

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Alex

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Apr 2, 2020, 10:21:45 AM4/2/20
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Wait, that's not what I meant!
 energy level complexity and vbr shouldn't case out of sync...  BTU they also don't affect crackly sounds!

I believe bbb_conference.xml and bb_echo_to_conference.xml settings affect crackly sounds AND might cause out of sync issue.
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Martin Thomas Schrott

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Apr 2, 2020, 10:24:44 AM4/2/20
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...hm...

so if the buffer settings remove crackling - but introduze out of sync ... we cannot fix anyway?

Or do you have any idea?

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Alex

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Apr 2, 2020, 10:26:00 AM4/2/20
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I have no idea. This needs to be looked at by someone who is pretty familiar with freeswitch ;((

Martin Thomas Schrott

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Apr 2, 2020, 10:32:39 AM4/2/20
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so, we tried to just change the buffer settings

and emedeately saw packet loss in the log... so sync also will get lost ...


Anyone an idea how to remove the crackling noise without getting the packet loss and getting out of sync?

:-(

Martin

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Richard Alam

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Apr 2, 2020, 10:54:30 AM4/2/20
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On Thu, Apr 2, 2020 at 10:32 AM Martin Thomas Schrott <li...@mtsonline.at> wrote:

so, we tried to just change the buffer settings

and emedeately saw packet loss in the log... so sync also will get lost ...


Anyone an idea how to remove the crackling noise without getting the packet loss and getting out of sync?

:-(

Martin


I spent more than a week on this a few months ago without solving the issue. It'll need FreeSWITCH developers help.

The challenge is figuring out where the clips are introduced. You can't play the packets when the browser sends them and you can't play the packets when the server receives them as they are encrypted.

I captured the audio when FreeSWITCH has decoded the audio and hear the clips there. So it might be the codec when decoding the packets introducing the clips. But it's difficult to prove. 

When getting the FreeSWITCH developers involved, we have to prove that the issue happens on the master branch of FreeSWITCH and on a supported OS. Unfortunately, BBB is not on master and not on supported OS.

If any of you can reproduce this issue on the later FreeSWITCH and on Debian, that would help a lot in trying to get the FS devs involved.

Richard


 
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Alex

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Apr 2, 2020, 1:13:49 PM4/2/20
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>. So it might be the codec when decoding the packets introducing the clips. But it's difficult to prove.

If it was the codec then why would changing  jitter buffer settings almost eliminate these sounds?

Regards,
ALex

Richard Alam

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Apr 2, 2020, 2:26:03 PM4/2/20
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On Thu, Apr 2, 2020 at 1:13 PM Alex <fedotov...@gmail.com> wrote:
>. So it might be the codec when decoding the packets introducing the clips. But it's difficult to prove.

If it was the codec then why would changing  jitter buffer settings almost eliminate these sounds?

Perhaps some packets are lost or delayed and the codec needs information from those packets to construct perfect audio?
And that jitterbuffer provides allowances for delayed packets helping in constructing better audio but introduces latency?

Just my guess.

Richard
 

Regards,
ALex

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Alex

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Apr 3, 2020, 2:54:18 AM4/3/20
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Hey Richard ..

about decrypting traffic


have you tried using wireshark?


Regards,
Alex
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Richard Alam

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Apr 3, 2020, 7:36:35 AM4/3/20
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On Fri, Apr 3, 2020 at 2:54 AM Alex <fedotov...@gmail.com> wrote:
Hey Richard ..

about decrypting traffic


have you tried using wireshark?


I looked into it but the keys are not available. So can't decrypt.
 
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Alex

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Apr 3, 2020, 2:00:55 PM4/3/20
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Hello Martin!


Could you try these settings https://pastebin.com/G0N8Dq7r  on one of your servers and see if there is still audio\video out of sync issue

Regards,
Alex





On Thursday, April 2, 2020 at 4:40:49 PM UTC+3, Martin Thomas Schrott wrote:

Alex

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Apr 3, 2020, 3:22:57 PM4/3/20
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Just wanted to clarify things:

Use the settings that don't produce crackly sounds BUT change bbb_conference and bbb_echo_to_conference settings to the ones I proposed in my previous post..


Regards

Martin Thomas Schrott

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Apr 4, 2020, 5:14:37 AM4/4/20
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Hi Alex,


thank you for not giving up ;-)

in the original post:

https://groups.google.com/forum/#!msg/bigbluebutton-dev/joJyywgYTTk/roFBwEq4EQAJ

sthephen wrote:

DistanceLearning.cloud To disable AEC/noise/agc just need to set constraints in getusermedia, can add this to bbb webrtc_bridge and sip.js code. Also there is a hack to get chrome stereo going with sdp. Se also can send 2ch binaural audio end to end. Regards,



1.  waht about the things to disable?

2. why don't we use the video-mcu-stereo settings? could you also provide me with that ones, so we could try them? the link of Roberto seem to be expired again.

3. would you mind to post all changes, that you made together... to make sure we do not forget something?


cheers

Martin

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Alex

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Apr 4, 2020, 7:00:17 AM4/4/20
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Hello Martin.


I haven't touched browser settings Roberto suggested. You should probably ask him how to do it and what changes specifically he made.

.
Here are the freeswitch settings I'm trying to test right now


I disabled jitter buffer on bridge in both bbb-conference and echo to conference. 

Regards...

On Saturday, April 4, 2020 at 12:14:37 PM UTC+3, Martin Thomas Schrott wrote:

Hi Alex,


thank you for not giving up ;-)

in the original post:

https://groups.google.com/forum/#!msg/bigbluebutton-dev/joJyywgYTTk/roFBwEq4EQAJ

sthephen wrote:

DistanceLearning.cloud To disable AEC/noise/agc just need to set constraints in getusermedia, can add this to bbb webrtc_bridge and sip.js code. Also there is a hack to get chrome stereo going with sdp. Se also can send 2ch binaural audio end to end. Regards,



1.  waht about the things to disable?

2. why don't we use the video-mcu-stereo settings? could you also provide me with that ones, so we could try them? the link of Roberto seem to be expired again.

3. would you mind to post all changes, that you made together... to make sure we do not forget something?


cheers

Martin



Am 03.04.2020 um 21:22 schrieb Alex:
Just wanted to clarify things:

Use the settings that don't produce crackly sounds BUT change bbb_conference and bbb_echo_to_conference settings to the ones I proposed in my previous post..


Regards
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Alex

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Apr 4, 2020, 7:04:29 AM4/4/20
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>hy don't we use the video-mcu-stereo settings

I don't need it. it's up to you what profile to use.

> waht about the things to disable?

mmm what things?   I'm testing right now with the settings I posted on pastebin (see my previous post

The only thing I've disabled there is jitter buffer on bridge

Regards.

Martin Thomas Schrott

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Apr 4, 2020, 8:45:24 AM4/4/20
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Hi Alex, Stephen and Roberto,

I am not in the dev list ...will have to contact him off list or join the other list too, if he does not read here (or if you do not mind, could you forward this to him?).




thank you for the config - I will look into that too



>why don't we use the video-mcu-stereo settings

I don't need it. it's up to you what profile to use.


does the mcu also use the opus codec or does it use the v8 vor video and audio? ... or to ask the other way: would it be the same to add "channels: 2" to the cdquality profile when only using audio?




> waht about the things to disable?

mmm what things?   I'm testing right now with the settings I posted on pastebin (see my previous post


:-) I refered to the browser (or not browser) settings that roberto noted as important:

"important for us was to manually modify flags/config in browsers to disable AGC and AEC, and noise suppression. "

As stephen said:

"To disable AEC/noise/agc just need to set constraints in getusermedia, can add this to bbb webrtc_bridge and sip.js code. Also there is a hack to get chrome stereo going with sdp. Se also can send 2ch binaural audio end to end. "

But I have no idea what that in fact means to me ... how would I do that (STephen?)?



thank you all - maybe together we will be able to find the perfekt settings for BBB :-)


cheers

Martin


Roberto Becerra

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Apr 4, 2020, 4:19:35 PM4/4/20
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Hi Martin, Alex, 

So, I went silent for a few days while I was chasing those changes Stephen mentioned to disable AGC, AEC and NS, and finally managed to hack the SIP file in 

/usr/share/meteor/bundle/programs/web.browser/app/compatibility/sip.js


Thanks to Stephen!!


Basically what I did was to look for the instances of constraints or where the media stream is created, and inserted custom constraints and forced them in:


line 11941->.  mediaHint = Object.keys(mediaHint || {}).length ? mediaHint : this.mediaHint;

replaced with-> mediaHint = this.mediaHint


and 


line 11956-> 

var constraints = mediaHint.constraints ||
(this.mediaHint && this.mediaHint.constraints) ||
{audio: true, video: true};

inserted my own constraints:
var constraints = {
audio: {
autoGainControl: false,
echoCancellation: false,
noiseSuppression: false,
},
video:false
};

Same disclaimer as before, take these changes with a pinch of salt, it works for me now, but I am risking making these changes and possibly breaking something. 

As for the previous changes, using the mcu profile was an attempt to achieve stereo, butt as mentioned somewhere here, I think you can achieve that by specifying the chann count on the profile itself. Other than that, we also noticed the shift in sync during conferences and also noticed that the less jitter management there was on the configuration the less dropped frames we observed. 

Best

Roberto 

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sd...@distancelearning.cloud

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Apr 4, 2020, 6:23:49 PM4/4/20
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Neat Roberto

 

echoCancellation: false,

Only issue with this is everyone has to wear headphones.   No open mics in meetings.  Because speaker out is going back in the mic with nothing to cancel it out.

 

If have stereo mic with L/R inputs, and the conference is set to channel=2,  you can actually have this binaural audio end to end….     Can set a channel constraint, and make it optional or mandatory

 

Have fun… tweaking.

 

Regards,

Stephen

Martin Thomas Schrott

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Apr 4, 2020, 7:43:56 PM4/4/20
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Stephen, Roberto,


thank you both very much ... I will test these settings on Monday - maybe without echo cancelation - guess the noise suppression and autogain: False should be enough to remove the cracklings?

We will test if there arise any out of sync problems with these and alex jb settings combined...

hopefully this together will do it eventually.

cheers

Martin

Alex

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Apr 5, 2020, 4:00:50 AM4/5/20
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He Martin.

Tbh I don't think noise suppression and autogain have anything to do with crackling. I mean.. it practically disappeared after we changed jitter buffer setting following Roberto's guide... Besides, noise suppression and autogain are universal webrtc settings and other webrtc software I used didn't have this problem.,.

You could do two tests:

1) Use  this ...conf.xml config and don't make any changes to  noise suppression and autogain. Check whether you'll have out of sync issues and\or crackling sounds issue.

2) Keep BBB's default ...conf.xml settings and just disable  noise suppression and autogain.

This way you can determine whether  noise suppression and autogain cause crackling sounds.

I'm looking forward to hearing from you about the results of these tests.

Regards.

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Roberto Becerra

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Apr 5, 2020, 4:19:51 AM4/5/20
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True that Stephen! I was actually focused and testing with re-routing signals in my DAW, because we are aiming at music applications. But its true that for a normal conversation we would have to enable EC.

And agreeing with Alex, AGC and EC are not about the crackling but about preserving the quality of the audio, without gain imposed to it or artefacts (other that crackling, namely dropped samples) 

Also an additional note that I have noticed from the hours since the last reply. These AGC and EC settings seem to work for Firefox, but not for Chrome, argh.. since chrome seems to still be inducing some sort of gaining, even though I disabled its own AGC flags in chrome://flags.

Anyone has an idea on why is that?

Martin Thomas Schrott

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Apr 5, 2020, 8:02:00 AM4/5/20
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Hey Guys,


yeah, I also think that the gain and noise suppression do not cause the crackling, but I also am trying to encrease the sound quality - so I will try all different settings to compare, where crackling disappears and if the other things improve the sound quality.

These artefacts and humbling of voice drive me crazy ;-)

As reference I keep jitsy, because it also works without extensions in firefox and has clear voice all the time. maybe they also disabled the noise suppression...


Roberto, I may be wrong, but I think that somewhere in the docs or lists there had been talked about chrome using an extension and only firefox using the devices natively. Guess stephen knows it and can clear things.


cheers

Martin

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Alex

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Apr 5, 2020, 8:51:34 AM4/5/20
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> jitsy, because it also works without extensions in firefox

You should NOT use firefox with jitsi until they make it work properly. Use only chrome

"There is a known issue where Firefox does not support simulcast and always sends HD resolution which is then forwarded by the videobridge to each client. Therefore, the client’s network may become overloaded if several clients are using Firefox."

Martin Thomas Schrott

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Apr 6, 2020, 5:24:54 AM4/6/20
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Hi Alex,


not sure about the out of sync yet, we will have to test more, but as soon as we disable jitterbuffer_msec, we seem to have more packet loss entries in the logfile, is this possible, or is it not related?

And is the packet loss note in the logfile a problem, or is it okay?


cheers

Martin

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Martin Thomas Schrott

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Apr 6, 2020, 7:08:44 AM4/6/20
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> And is the packet loss note in the logfile a problem, or is it okay?
>
>

ok I figured out, that the packet loss occurs, if the "lost time" is
more than the set jitterbuffer ...so our connections have a loss higher
than 60ms - that is why it shows the packet los in the logfiles with
your jb settings.


I changed it to 70:300:100 and the packet loss disappears emedeately
from the logfiles.

But I have some crackling noises - but I think also with the 60ms I have
them from time to time.


as we turned off the look ahead in opus config and the keep vec enabled,
does the jb even work at all - and if so, what was the look ahead and
keep vec enabled for? ...

they seem to create more cracklings, as far as I could test.

but this all may always be due to connection problems at the testing
time ... :-/


Alex, what do you think causes the crackling mostly?


I now disabled the VBR and I set the interval in conference.xml to 10.
that both together improves quality a bit.

I also have noise suppression and autogain turned off in sip.js.


So quality is quite ok now.

Still would be nice to get the last cracklings removed though.


regarding the out of sync problems, currently we did not notice any, but
we will have to test with the users, that had that all the time. Should
be possible to do that this afternoon.

I will get back afterwards.


cheers

Martin




Martin Thomas Schrott

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Apr 6, 2020, 9:42:55 AM4/6/20
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ok guys,


I guess we have a quite good setting now ...

Alex, may you also try that:

in the dialplan I encreased your jb to 200

this prevents the packet loss and hopefully also the out of sync. We
culd not notice problems today - we will test tomorrow with more users.


<action application="set" data="rtp_jitter_buffer_plc=true"/>
<action application="set" data="rtp_jitter_buffer_during_bridge=false"/>
<action application="jitterbuffer" data="200"/>


* I set the interval in conference.xml to 10 instead of 20.

Energy-level has to be 0, allthough that makes problems with the display
of the active speaker, because most are always shown as active.

Here it would be great to have a separate "display-energy-level" so we
could set this to 300 without breaking the audio itself.


      <param name="interval" value="10"/>
      <param name="energy-level" value="0"/>


I also set stereo, not sure if that really works:

      <param name="channels" value="2"/>


* The opus settings are as alex  worked out, everything turned off, more
complexity and I gave more packet loss percentage and turned off the vbr:


        <param name="use-vbr" value="0"/>
    <param name="complexity" value="10"/>
        <param name="packet-loss-percent" value="15"/>


* as Roberto discovered but without turning off the echo cancelation:

/usr/share/meteor/bundle/programs/web.browser/app/compatibility/sip.js

line 11941->.  mediaHint = Object.keys(mediaHint || {}).length ?
mediaHint : this.mediaHint;
replaced with-> mediaHint = this.mediaHint
and
line 11956->
var constraints = mediaHint.constraints ||
(this.mediaHint && this.mediaHint.constraints) ||
{audio: true, video: true};
inserted my own constraints:
var constraints = {
audio: {
autoGainControl: false,
echoCancellation: true,
noiseSuppression: false,
},
video:false
};


(could we now set these auto gain / echo / ns settings somewhere in the
config files? How would that look like and in which config file?)


this seems to work very well, hope it is not temporary. ;-)

cheers

Martin


Roberto Becerra

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Apr 6, 2020, 9:49:23 AM4/6/20
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Martin, nice! thanks! Will have a test with your settings tomorrow and report back to everyone.

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Alex

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Apr 6, 2020, 11:36:52 AM4/6/20
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I've got a question though...Could high jitter buffer cause out of sync issue?

So, Martin, you tested with JB=200 and 15% packet loss in opus and you didn't experience any out of sync and crackly sound issues?

Anyways, I totally forgot how to check whether BBB session had packet loss problems  Could someone remind me how to do it?

Thanx.!

Alex

Martin Thomas Schrott

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Apr 6, 2020, 11:51:09 AM4/6/20
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Hi Alex,


yes exactly, 200 and 15% - no crackly noises and so far no out of sync problems, but as I said, we will be able to test this with more users tomorrow morning.

If this test also passes, this should be a solution... so let's hope it will. ;-)

The sound defenitely seems to be  okay, I had a few different tests now with a few users and different internet connections.

But I do not want to be happy before it really is proofed - also by others.


did you already have the chance to make a first test?

cheers

Martin

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Alex

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Apr 6, 2020, 12:05:28 PM4/6/20
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did you already have the chance to make a first test?




Nah I haven't yet. 

> 200 and 15% - no crackly noises and so far no out of sync problems

good;)

Alex

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Apr 6, 2020, 12:52:03 PM4/6/20
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Hey Martin

This is what out teacher experienced today

https://drive.google.com/file/d/1cDDxk0JiELOLtvfA1mRJaxh8hGfKnnk2/view?usp=sharing

It starts at ~ 5.40

I dunno what it was. She said she had this issue several times during her lecture.

It happens with 200ms jitterbuffer too.

I'll try to remove <action application="set" data="rtp_jitter_buffer_during_bridge=false"/> and see how it goes.

Regards

Alex

sd...@distancelearning.cloud

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Apr 6, 2020, 1:59:15 PM4/6/20
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Do you have turn?  Would be interested in seeing in logs if this user was relaying.

 

If relaying tcp you don’t ever drop packets..  since protocol just retries..  so this introduces different sound artifacts then UDP since data is delayed,

 

With normal UDP, and the standard settings with FEC…   missed packets <10% don’t matter the codec looks at some other packets, guesses and keeps on playing often with the “average” user not even noticing.

 

With FEC off..  you will introduce crappy audio any time packets are lost (cant control this in open internet)

 

So every tweak in audio configs has implications..    Chrome AEC Echo cancellation can get out of sync if the jitter buffers overflow and you hear echo all of a sudden.

 

And setting the interval in conference platform from 20 to 60, means only 1 out of 3 samples gets mixed in conference.

 

Regards,

Stephen

 

 

 

From: bigbluebu...@googlegroups.com <bigbluebu...@googlegroups.com> On Behalf Of Alex
Sent: Monday, April 6, 2020 12:52 PM
To: BigBlueButton-Setup <bigbluebu...@googlegroups.com>
Subject: Re: [bigbluebutton-setup] Audio quality was: (Could someone post a 1.5-2 minutes recording of them talking (or playing an audiobook) via BBB conference?)

 

Hey Martin

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Alex

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Apr 6, 2020, 2:26:31 PM4/6/20
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1) I do have a turn server running, but I doubt I could find this user or what was happening in the logs. I don't know her IP address.


2)  We don't have interval set to 60 in a conference.conf.xml. It is set to 20.  Jitter buffer is set to 60 (now testing 200 =)

The user in question was using mobile internet shared to her notebook.  Perhaps she got some packet loss (networks especially the mobile ones, are overloaded now coz of the whole coronavirus situation)

Regards,
Alex

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Martin Thomas Schrott

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Apr 6, 2020, 3:14:25 PM4/6/20
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Stephen, Alex,


so what does the jitterbuffer do exactly - if it also could cause this artifacts (on tcp)

and does the vec prevent theese tcp artifacts?


I had some packets lost  since we moved from jb 60:300:60 to jb set to 200.

So I tried jb with 60:300:60 but got back the crackling.

For me it seems, that the crackling occurs as soon as you split the jitterbuffer settings to three, instead of just one value. Could this be?

What does the buffer if only one value is set? means this, that the startvalue is the same as the max and the max drift?


I set the jb to 300 now, no idea if this will work without packet loss and even less idea if this makes out of sync problems again.


Any more thoughts Stephen? ;-)


cheers

Martin

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Martin Thomas Schrott

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Apr 7, 2020, 6:56:04 AM4/7/20
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hi guys



Am 06.04.2020 um 21:14 schrieb Martin Thomas Schrott:
>
> Stephen, Alex,
>
>
> so what does the jitterbuffer do exactly - if it also could cause this
> artifacts (on tcp)
>
> and does the vec prevent theese tcp artifacts?
>
>
> I had some packets lost  since we moved from jb 60:300:60 to jb set to
> 200.
>
> So I tried jb with 60:300:60 but got back the crackling.
>
> For me it seems, that the crackling occurs as soon as you split the
> jitterbuffer settings to three, instead of just one value. Could this be?
>
> What does the buffer if only one value is set? means this, that the
> startvalue is the same as the max and the max drift?
>
>
> I set the jb to 300 now, no idea if this will work without packet loss
> and even less idea if this makes out of sync problems again.
>
>

some people do not like the large jb setting, because they think audio
is too much behind the video. So I had to set it to 40.

I also set the packet loss percentage to 8

and activated (as in the default) vec and jb_lookahead.


This still seems to have no or very few cracklings.


What has to be disabled is: energy-level, this causes crackling all the
time.

And i disabled vbr because of bad audio quality.

Still just trying - not really knowing what I do ;-)


Maybe this setting can prevent the tcp artifacts Alex encountered. It
may be a compromise between good audio and no crackling noises.

I still have disabled autogain and noise supression.


unclear for me still are:


1. what the jitter buffer exactly does - and why setting all three
values instead of only one does crackle.


2. will we have a chance to fix the "user is speaking" display when the
energy-level is disabled - we really need a display-energy-level so
audio is not broken but the speakers are shown correctly.


3. why is audio all in all so bad encoded - I tryed to stream music
through bbb and it sounds horrible - where is the "cdquality"?? it
sounds like an old radio from the 1930s. As long as I read and
understand, opus should be able to stream hd quality - this is not even
phone quality. There has to be some big issue that we did not find yet.

Even if voice sounds quite okay now - with this settings, it does not
sound great as it does in jitsy or zoom. And music does not work at all,
but it should just work as well, when we pretend to have 48khz - so what
did we miss? where is the real problem... :-/


cheers

Martin


Alex

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Apr 7, 2020, 10:34:49 AM4/7/20
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What's your current dialplan configs ,Martin?


PS Have you by any chance backed up the original configs?   They are not on github.  I guess I'll have to install bbb on my vps tp get them lol

Martin Thomas Schrott

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Apr 7, 2020, 10:59:40 AM4/7/20
to bigbluebu...@googlegroups.com, Alex

Hi Alex,


I sent you the files in a mail off list some hours ago ;-)


I use your dialplan settings but instead of 60 I have 40 for the jitterbuffer as some users complained the audio too far behind video.


In opus I have turned on vec and jb_lookahead

this seems to work for voice as long as energy-level is turned off and autogain/noise suppression also are disabled in sip.js.


To reach cdquality, what it pretends to be, I have no idea, what we could try, as music does not work at all ... And music should be our reference, when this will be working, we will have reached good audio quality. :-)

cheers

Martin

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Alex

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Apr 7, 2020, 11:56:04 AM4/7/20
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haha I didn't notice. Thanx a lot.


Btw regarding the audio issue recording I posted a few posts above. Our teacher was using a mobile network, so I think this issue was related to some packet loss during peak hours. JB was set to 60 and to 200.

I had other 2 lectured today  and this time

1) teacher1 (the mobile network) had this problem again BUT with JB set to 60:200:20 it didn't last as long and the audio got fixed pretty fast. Very few if any crackly sounds.

2) teacher 2 (wired 1Gbit connection, JB was set to 60:200:20). There were very few if any crackly sounds but in time audio and video got slightly out of sync anyway and by the end of the lecture sometimes I could hear something similar to the recorded above issue but it was barely noticeable
.

I think simply setting JB=40 or 60 would work pretty badly on a slow mobile network prone to packet loss problems..but I might be mistaken

Please, do keep informed on how things will be working out for you with your settings.

Martin Thomas Schrott

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Apr 7, 2020, 2:16:48 PM4/7/20
to bigbluebu...@googlegroups.com, Alex

Hi guys,


it is so frustrating. When I set the jitterbuffer to 40 sound is quite okay (where okay still means bad when you compare with jitsi or zoom)

but it gets out of sync too often.

If I set the jb to three values e.g. 60:120:30  it keeps in sync, but seems to correct very often and on correcting has the crackling noise.

In our tests it seems, that the crackling mostly appears if webcams are turned on and only a few crackles occur when audio only conferences are used.


does only one value in the jitterbuffer mean, that there is no max_drift? or does it mean all three values are the same?


How can it be, that we do not reach a quality near jitsy? What do they do different? :-(


Could we add some offset so video is a bit delayed - then we could give more jitterbuffer and hope this tastes more in sync?


Or does anyone have any more suggestion / idea what could be the real/main problem so we can try to solve it?

Our goal should be to stream music and have the same quality as you would have when you listen to a shoutcast or youtube stream. Opus can hd quality and we do not even reach a cracklefree mono audio stream.


:-(

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sd...@distancelearning.cloud

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Apr 7, 2020, 3:34:02 PM4/7/20
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jitsi is just relaying everyone’s original audio captured in browser.

If you have 20 users.  20 audio streams just relayed to everyone.

SFU does nothing but relay unaltered media.

 

Freeswitch 1) decodes opus, 2) converts to L16, 3) mixes with all the other audio and comfort noise then,  4) encodes back to OPUS and 5) send 1 audio stream to everyone. 

 

With freeswitch you get single mixed recording of session.

 

Pros and cons of both

 

Regards,

Stephen

 

 

From: bigbluebu...@googlegroups.com <bigbluebu...@googlegroups.com> On Behalf Of Martin Thomas Schrott
Sent: Tuesday, April 7, 2020 2:16 PM
To: bigbluebu...@googlegroups.com; Alex <fedotov...@gmail.com>
Subject: Re: [bigbluebutton-setup] Audio quality was: (Could someone post a 1.5-2 minutes recording of them talking (or playing an audiobook) via BBB conference?)

 

Hi guys,

Martin Thomas Schrott

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Apr 7, 2020, 4:18:30 PM4/7/20
to bigbluebu...@googlegroups.com, sd...@distancelearning.cloud, Chad Pilkey, Frederick Dixon

Stephen, 

thanks so much, now I understand all that trouble eventually.


@Fred, Chad and the other core developers (excuse to include you here) but I think only with you this can be improved...


I found this thread:
which explains how and which technology was chosen.
Fred said:
">  Then I have a question, why use FreeSwitch to composite the audio? As usually the audio should consume small bandwidth (compared to video), it doesn't save much bandwidth using FreeSwitch instead of SFU?

Because FreeSWITCH rocks :-).  It does an excellent job mixing the audio (both from the browser and SIP trunking) and we've built upon FreeSWITCH for years . It also handles WebRTC calls directly with all the support for compression and forward error correction.  If you think the audio in BigBlueButton sounds good, it's because of the browsers support for WebRTC audio and FreeSWITCH.

We've been using FreeSWITCH for over six years in the project.   We are always inclined to choose the best tool for the job, and FreeSWITCH (IMHO) is the best tool for handling audio.

Regards,... Fred"


so let me tell: it does not rock. Sure, for very large setups and low bandwith the idea to mix the audiostream is great and maybe the only solution. But the audio really is not good and people do compare to jitsi or zoom and switch over to the other services frequently.

Would there be (or is there already) a way to route the audio via sfu instead of mixing in freeswitch?
Globally in a bbb server
or even better per room?
Could this be made available via a setting e.g. use_sfu_up_to_max_users_per_room: 20
so bbb would keep sfu until this max value is reached and beyond that switches to the mixing in freeswitch?
If this is not possible, so just make it as an option on creating rooms?
That would be so amazing, because we could choose good quality for videoconferences
and best quality with mixing for large rooms / teaching / webconferences with a lot of users.

I think this should be possible, if I understood it correctly - the recording still could be routed through freeswitch and also dial in users could still be routed through freeswitch and streamed to all other users ... so there should be no problem with that.
Only the other audio could be routed through sfu - if configured.

I am sure Alex / Richard / Roberto also would be happy to see this and may help where possible, to get that up and running.

Thoughts? Problems?

hopefully
Martin

Martin Thomas Schrott

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Apr 7, 2020, 7:40:10 PM4/7/20
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Hey guys,


in addition to my last message ...

first: i clearly would love to have the choice with sfu - as stated and explained in my last message - and hope someone can help to impliment this...

am I right, that this file would be one of the places where this happens?

https://raw.githubusercontent.com/bigbluebutton/bbb-webrtc-sfu/master/lib/audio/audio.js

...really - I have no idea but it looks as if this would do the inbound routing ... but to additionally route via sfu would not be enough, the mcu also would have to be changed a bit to prevent doublicate streams... Hope someone - who knows what to do - will help here.


second,  stephen explained, that freeswitch "2) converts to L16, "

I guess here is where the audio problems are introduzed. Would it be possible to just let the streams be in opus and mix the opus streams with eachother? this would remove decoding and also not need reencoding. Would safe a lot cpu  and - I am quite sure - would prevent audio problems, because mixing should not make crackles or other artefacts at all.

(yes, I guess only other opus files could be mixed, so chat and moh and comfort noise will have to be opus right?)

But this should be a quite easy way to get better audio, shouldn't it?

And just to test that, only some de- and encoding would have to be removed and a mix of the opus files added.

Or am I missing something here? :-/


cheers

Martin

Alex

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Apr 8, 2020, 2:32:29 AM4/8/20
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Tbh I would be happy if we could just deal with crackly sounds first ;)

Maybe jitter buffer needs some tweaking we don't know about? Considering that changing it affects crackling.


PS As for jitsi and zoom.. Don't forget that..

1) zoom is running numerous powerful servers all over the world, you don't host anything, most if a not all processing happens on their side

2)  We tried using Jitsi and we really liked the sound but Jitsi has some serious problems:

-  Jitsi uses HUGE amount of bandwidth per stream

- It basically properly works only with chrome and chromium based browsers atm. (see simulcast issue)

- One session can hold about 35+ users at once, after that various problems might start  which was also confirmed by the devs. So to host more without problems you need to start running JVB2 bridges on separate servers (creating BBB's scalite-like infrastructure).

- When you reach 35+ users per session browser might get unresponsive on a client's side or slow down greatly.

If your team is really small and you need a good audio quality for making music and stuff then jitsi might be a good choice. But imho it has too many flaws to use it for anythign else.   Afaik they are trying to deal with some of thee flaws but they are not there yet.

Regards,

Alex

Martin Thomas Schrott

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Apr 8, 2020, 2:55:27 AM4/8/20
to bigbluebu...@googlegroups.com, Alex

Am 08.04.2020 um 08:32 schrieb Alex:
> Tbh I would be happy if we could just deal with crackly sounds first ;)
>
> Maybe jitter buffer needs some tweaking we don't know about?
> Considering that changing it affects crackling.
>

Hi Alex, I am always willing to keep on testing, but I have no more
ideas and audio quality does not get great, just a bit better - with
loosing sync.

As I said, if we just use one value in the jitterbuffer e.g. 60 as you
used, crackling is minimized but in my opinion the jitterbuffer then
does nothing, because sync gets lost on packet loss.

As soon as I set three values e.g. 60:240:30 sync is much better (I
think because of the max drift value?) but crackling is back.


I really would like to try without the conversion from opus to l16 and
back to opus, as in my opinion this could be the issue, where the jb
interfears. But this really just is a guess that should be proofed.

For really good quality this may not be enough - that's why I asked if
sfu could be used for smaller rooms - where bandwith would not be a problem.

I am aware of the bandwith problem - that's why this always only could
be optional in the settings. But it really would be a good feature for
high quality teaching - our language department already had to change to
zoom because they need better voice quality than we can reach with bbb.
This should not happen.


So what would you try next? ;-)




>
> PS As for jitsi and zoom.. Don't forget that..
>

all true, but why not have the best of every system? If sfu can be used
in bbb alternatively it would be interesting for jitsi users as
alternative. And we would not loose users that need high quality audio.

Zoom unlikely cannot be compared as they have their own client and much
more flexibility because of that. But we should not forget to always
have a look at their quality as a goal to reach.


Hope we can get further

and improve quality much more...

cheers

Martin


Alex

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Apr 8, 2020, 5:43:32 AM4/8/20
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Hey Richard

but..doesn't it seem like it's a configuration issue caused by misconfigured jitter buffer settings in bbb_conference and bbb_echo_to_conference?

Sincerely yours,
Alex

When getting the FreeSWITCH developers involved, we have to prove that the issue happens on the master branch of FreeSWITCH and on a supported OS. Unfortunately, BBB is not on master and not on supported OS.

If any of you can reproduce this issue on the later FreeSWITCH and on Debian, that would help a lot in trying to get the FS devs involved.

Richard


Martin Thomas Schrott

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Apr 8, 2020, 5:49:53 AM4/8/20
to bigbluebu...@googlegroups.com, Alex

> but..doesn't it seem like it's a configuration issue caused by
> misconfigured jitter buffer settings in bbb_conference and
> bbb_echo_to_conference?


by the way, could someone explain, what the three locations are, where
we set the jitterbuffer? In conference we have two extensions - why?

And was is echo_to_conference? Is this just the situation during the
echotest? Or what does echo mean here?

Just to have more knolledge where we do need the settings and where we
may not need them.

thanks

Martin


Richard Alam

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Apr 8, 2020, 7:05:19 AM4/8/20
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Hi,

On Wed, Apr 8, 2020 at 2:55 AM Martin Thomas Schrott <li...@mtsonline.at> wrote:

Am 08.04.2020 um 08:32 schrieb Alex:
> Tbh I would be happy if we could just deal with crackly sounds first ;)
>
> Maybe jitter buffer needs some tweaking we don't know about?
> Considering that changing it affects crackling.
>

Hi Alex, I am always willing to keep on testing, but I have no more
ideas and audio quality does not get great, just a bit better - with
loosing sync.

As I said, if we just use one value in the jitterbuffer e.g. 60 as you
used, crackling is minimized but in my opinion the jitterbuffer then
does nothing, because sync gets lost on packet loss.

As soon as I set three values e.g. 60:240:30 sync is much better (I
think because of the max drift value?) but crackling is back.


I really would like to try without the conversion from opus to l16 and
back to opus, as in my opinion this could be the issue, where the jb
interfears. But this really just is a guess that should be proofed.


Freeswitch need to convert the audio to a common codec to mix it. With freeswitch,
calls might come in from different sources with different codecs. I believe there is
no way to tell FS which codec to use.
 
For really good quality this may not be enough - that's why I asked if
sfu could be used for smaller rooms - where bandwith would not be a problem.

I am aware of the bandwith problem - that's why this always only could
be optional in the settings. But it really would be a good feature for
high quality teaching - our language department already had to change to
zoom because they need better voice quality than we can reach with bbb.
This should not happen.

We like to have great audio too. In fact we were testing the settings  you posted on the weekend.

Supporting Jitsi will take time and effort but we have very limited resources. Maybe we can look at
a feature to be able to use Kurento to just stream audio without mixing with the understanding that
it can be used only for limited streams. That's a Paulo feature as he's the Kurento expert :)

Richard
 


So what would you try next? ;-)




>
> PS As for jitsi and zoom.. Don't forget that..
>

all true, but why not have the best of every system? If sfu can be used
in bbb alternatively it would be interesting for jitsi users as
alternative. And we would not loose users that need high quality audio.

Zoom unlikely cannot be compared as they have their own client and much
more flexibility because of that. But we should not forget to always
have a look at their quality as a goal to reach.


Hope we can get further

and improve quality much more...

cheers

Martin


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Richard Alam

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Apr 8, 2020, 7:20:54 AM4/8/20
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/opt/freeswitch/conf/dialplan/default# cat bbb_conference.xml
<include>

This is the extension that handle calls from the browser.

    <extension name="bbb_conferences_ws">
      <condition field="${bbb_authorized}" expression="true" break="on-false"/>
      <condition field="${sip_via_protocol}" expression="^wss?$"/>
      <condition field="destination_number" expression="^(\d{5,6})$">
        <action application="set" data="jitterbuffer_msec=20:400"/>
        <action application="answer"/>
        <action application="conference" data="$1@cdquality"/>
      </condition>
    </extension>

This is for people calling in from phone and from Flash (when it was supported)
    <extension name="bbb_conferences">
      <condition field="${bbb_authorized}" expression="true" break="on-false"/>
      <condition field="destination_number" expression="^(\d{5,6})$">
        <action application="set" data="jitterbuffer_msec=20:400"/>
        <action application="answer"/>
        <action application="conference" data="$1@cdquality"/>
      </condition>
    </extension>
</include>

This is the echo test extension 
/opt/freeswitch/conf/dialplan/default# cat bbb_echo_test.xml
<include>
  <extension name="bbb_echo_test_direct">
    <condition field="${bbb_authorized}" expression="true" break="on-false"/>
    <condition field="destination_number" expression="^9196$|^9196(\d{5,6})$">
      <action application="set" data="vbridge=$1"/>
      <action application="answer"/>
      <action application="set" data="bbb_from_echo=true"/>
      <action application="bind_digit_action" data="direct_from_echo,1,exec:execute_extension,ECHO_TO_CONFERENCE XML default"/>
      <action application="sleep" data="500"/>
      <action application="echo"/>
    </condition>
  </extension>
</include>

This transfer the caller from echo test to the conference.
Try removing the jitterbuffer here and leave the two one ones above.  

/opt/freeswitch/conf/dialplan/default# cat bbb_echo_to_conference.xml
<include>
  <extension name="ECHO_TO_CONFERENCE">
    <condition field="${bbb_from_echo}" expression="true" break="on-false"/>
    <condition field="destination_number" expression="^(ECHO_TO_CONFERENCE)$">
      <action application="set" data="jitterbuffer_msec=20:400"/>
      <action application="answer"/>
      <action application="conference" data="${vbridge}@cdquality"/>
    </condition>
  </extension>
</include>


thanks


Martin


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