Exploring Call-in / Dial-in

Services like Jit.si, zoom, and the likes they have facilities to call into meetings over normal phone lines. BigBlueButton also can provide that. We haven’t set that up yet, for a number of reasons, but we’d like to explore that with you. We’re likely to make a few first baby steps to test the waters and see which members really need it. Did I say that making an extra donation will help us speed up? :wink: Seriously, we are aware that for some people with audio problems, having a call-in option would really make a difference.

Most importantly: BBB has the key facilities to make this happen. Basically, the audio switching system (called freeswitch) is making it possible to call in and out over a SIP connection. SIP is the open standard for audio connections, many office phones use that standard as do many of the phone providers nowadays.

First steps
So the first step would be to expose the SIP account to each of our BBB servers. If you have a SIP phone device (hardware or software), you can simply call into that SIP address and select the room you want to join. This of course will require some configuration work for the tech.circle, but shouldn’t be too hard.

The second step would be to point a dial-in number from the Plain Old Telephony System (POTS) to the desired SIP account of our servers in Germany, Canada or the demo server. We will need to select a proper dial-in provider. Consumer grade dial-in numbers costs as little as 75 cents/month at for ex localphone.com, but you can only call in with one person at a time. We’ll be needing a business service.

Then the third step would be to have a routing server, that allows you us to put many dial-in lines to one BBB server. It is called a PBX, the famous one in the free world is called Asterisk. This will allow us many different dial-in numbers and route them. Then there’ll be the challenge to route to different BBB servers. Probably we’ll end up with different Asterisk PBXes?

Who needs it now?
We’ll be interested to explore this for members who really need this, otherwise it wouldn’t make sense. Be prepared that at first it’ll be experimental, taking one step at a time and see how that works for your people.

Detailed questions:

  • how many people of your userbase would need to call in at the same time in a certain continent?
  • We could start by hooking up a US or Canadian number to the CA server and a German number to the DE server. Would that cover (most) of your needs in the short term?
  • Did you explore SIP phone software like SIPdroid on Android? That could be a solution for some people, and would cost zero money to the collective and to the end user (considering that the enduser has Internet connection on their mobile)

Please feel free to add your thoughts, usecase and attractive SIP service providers.

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I am asking GEO folks about how many people use dial in. In my experience in various groups, it is usually on the order of 10% or fewer of the people in a given call. A US/Canadian number would be great for my purposes. SIP sounds like a good option, would it make sense to run a trial?

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We would probably need 3 callers at the most. One is usually enough, so we could start with that.

meet.coop tech circle are looking at a trial. Might need to be on the demo server, the main server must stay stable for development in support of Amsterdam Cities programme right now.

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Any news about Dial-in? It is preventing some people we refer to the service from switching.

It’s on the roadmap, but not within the next ¿2? quarters. @wouter @dvdjaco @Yurko

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This month with tech.circle we’re doing several improvements for the City of Amsterdam / Cities for Change programme (live streaming, simplified joining of meeting rooms, server cluster).

As far as I know the Dial-In experiments would start after that. That is not yet a full roll-out, but once @yurko gets some time to work on it, we might want to start with people calling in over a SIP connection, then rent a Digital phone number in some country and allow people to call in over that number. When people want to pay for it we might want to add numbers in more countries and with more capacity (for concurrent calls). And ultimately we need want to work on routing algorithms so people can call in to a virtual PABX and then choose to what server cluster (in what continent) they’d like to call to what room.
So in sum, we can start simple but end up more complex and more costly on the way :slight_smile:

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I think we can maybe look into doing a pilot project for adding dial in support soon. The hard part of this tasks is the scale required especially since it requires to purchase resources out to another network (the PSTN or Telephone System).

For starters getting some data would help us scale the pilot property. If we can collect some data such as:

  • who is interested in this feature (and who would be willing to help beta test it)
  • where they are located and/or where (country) they are willing to call into
  • what kinda usage would they have (how many simultaneous calls)

This would be a great start to start looking into an implementation!

@mnoyes @wolcen @b5baxter

I am interested and would help beta test it.

Our co-op is located in Canada (Vancouver, BC).

We would probably need 3 callers at the most. One is usually enough, so we could start with that.

Typically, only we only need a single dial-in line, if at all.

From our perspective:

  • We see that others are unwilling to switch to this service due to the safety-net of this feature not being available (we recommend the service a lot).
  • For us, it is also a requirement to dropping a backup service.

There’s only one group we’ve recommended to the service that refused due to a hard requirement for multiple connections (e.g. 5-10).

Our COOP is based in the US with partners in Nicaragua and Germany. We’re also willing/available to help with testing.

Thank you!

The use cases I am thinking of involve one to three people dialing in at a time. We could beta test with GEO, I think. (USA)

I’m one of the sysadmin in the Tor Project, and we’ve been using meet.coop’s BBB instance with great success for all sorts of things:

  • all hands, ad-hoc, or weekly team meetings
  • training sessions
  • “office hours”
  • “demo days”
  • hack week

It is working great! Sometimes we have people in South America that have trouble getting through on audio, but I’m not sure it’s due to BBB or the link… Hard to tell…

Now to answer your questions directly…

i’d say half a dozen to a dozen or so. Our largest meetings are between 30 to 50 people, I think, and I’d say we’d typically have 5-6 people in the largest meeting on phone. But who knows?

Absolutely, that would work great for us.

I have not tested SIPdroid itself. I have used the built-in SIP dialer that’s part of the stock Android “Phone” app (which, amazingly, is often missing from commercial Android deployments: it’s part of LineageOS and CalyxOS for example)… It works fine for my purposes.

I’d be happy to test a SIP endpoint if that’s useful. I use baresip on my desktop…

For business hosting, particularly in Canada, I recommend https://voip.ms, which has good service and pricing. I don’t think you’d need to setup Asterisk, to be honest: in theory freeswitch is a complete replacement and can do routing and all that stuff. I’d advise against it, actually: having two PBX systems to learn will make training your staff needlessly hard.

For me, the killer feature is to be able to bridge audio from my computer into BBB without firing up a web browser that eats all those precious computer resources and power. During office hours, I have a browser window open all day waiting for people to jump in for questions, and it’s sometimes spinning up my CPU fan, just doing nothing. If I could just sit there with my SIP client instead, it would be much more lightweight, and I could still attend.

This could also allow for some cool features like piping music on hold, which I’m currently doing with Pipewire hacks. :wink:

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This is so important. If meet.coop is serving people in the global-South - or just any people who don’t have lots of credit or bandwidth on their phones - we need to be very careful about the amount of resource that our platform demands on the user’s device. This is basic Design Justice.

Do we in fact have a policy or stance on just what kinds of end-user devices we mean to support? (Defining bands of capability? like we define bands of ‘fair use’?) A spectrum, of course. But how many points are there on that spectrum? what are their respective priorities for our provisioning? and how different does the platform need to look, from each of these different device standpoints?

as an aside, people should look at this talk before digging too deep into FreeSwitch land. I’m not exactly sure (i skipped through a lot of the talk), but i suspect they might support other tools for SIP, for example they are testing Janus as an alternative backend which would replace both the current Kurento and Freeswitch backends, from what I understand. This may come in the upcoming 2.4, as experimental. See also this design document.

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Hello all.

I’m happy to say we have been working on the pilot project for dial in participants to our Big Blue Button instance!

There are many things we would still need to figure operationally before this can be a permanent fixture, but its a challenge we will approach head on and this pilot project will give us some information to make those decision.

The metrics of this pilot project will help us understand the demand on the system and how to structure it to create a viable and sustainable model moving forward.

The First Phase of the pilot is now up and running! This phase consisted of:

:white_check_mark: Acquiring a Canadian phone number
:white_check_mark: Configuring ca.meet.coop to accept calls from the assigned number
:white_check_mark: Configure a SIP dial-in number for use with a SIP Client. I have tested with Linphone (F-Droid, Google Play)

The second phase is now up and running. This phase consisted of

:white_check_mark: Acquiring a European phone number
:white_check_mark: Configuring de.meet.coop to accept calls from the assigned number
:white_check_mark: Configure a SIP dial-in number for use with a SIP Client. I have tested with Linphone (F-Droid, Google Play)

How It Works

Conference PIN will be generated when you start a meeting on both CA and DE servers. It will be displayed at the top of the chat. You must provide this number along with the dial-in number to the individuals that wish to join the conference via the phone.

The two dial in options. Please use the correct numbers based on the geographic location

CA Server

  • SIP at 309817101@sip.ca.meet.coop
  • Dial-in to Canada (Ontario) at 807.788.MEET(807.788.6338)

DE Server

  • SIP at 309817102@sip.de.meet.coop
  • Dial-in to Spain (Barcelona) at (34)932208649

When you dial in you will be asked for the Conference PIN Number. Once entered you will be placed into the room.

During the call the following touch-tone options are available

0 - Mute/UnMute

  • This is a normal mute, just your outgoing audio. You can still hear what is going on.

* - Mute/UnMute

  • This Mute is for outgoing AND incoming audio. You will not hear anything, and no one will hear you.

Increase YOUR volume
3 Talk Volume Up
2 Talk Volume Zero
1 Talk Volume Down

Increase everyone else’s volume
6 Listen Volume Up
5 Listen Volume Zero
4 Listen Volume Down

Energy level is a threshold that dictates the level at which a person is determined to be speaking versus the background noise received.
9 Energy Up
8 Energy Zero
7 Energy Down

Things to keep in mind:

  • Conference PIN is randomly generated every time you start a new meeting. There is no way to predict what it will be until you create the meeting.
  • You must have at least ONE user in BBB with voice connected in the BBB client otherwise the PIN will not work.
  • If your room is set to “mute on join” users will need to press 0 (unmute) first to be able to talk.

As always please provide any feedback in this thread :slight_smile:

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Thank you for all your expertise anarcat! I pretty much agree with you on all the points!

VOIP.MS if a little-known gem in the sip industry and they are great. My biggest issue is how to expand this internationally. The international DIDs are limited to 2 channels. I’ve contacted them to see if there is a feasible solution to this problem

I agree, if we can figure out a way to keep all the sip stuff on the voip.ms platform, we can leverage it to do routing, such as leverage our international phone numbers to connect to any of our servers. But there is quite a bit of operational work that would need to get done!

Thank for this very interesting.

We are still in the process of moving to 2.3, i dont think 2.4 will be something that will be available in the near future. So with that, we spent the effort on free switch to get these features now :slight_smile:

Guess we just have to port them when 2.4 comes along =)

thanks @Yurko for brining in your great expertise with SIP Dial-in! I’ve tried it a few weeks ago and it worked great. It is nice to see a different icon for users dialing in to distinguish them from web users.

I’m curious to hear from the user members who have asked for this. How does it work for you, @b5baxter @wolcen @mnoyes @anarcat ?

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This is terrific @Yurko thanks so much for picking this up and running with it. Time to develop some user documentation. This calls for a thread under Product strategy category? Perhaps new sub-thread, Documentation/FAQ? Who’s to open this? @Yurko maybe, you’re currently the wizard of dial-in :wink:

2 posts were merged into an existing topic: BBB on mobile operating systems