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:
Acquiring a Canadian phone number
Configuring ca.meet.coop to accept calls from the assigned number
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
Acquiring a European phone number
Configuring de.meet.coop to accept calls from the assigned number
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 
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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 
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 
Thank you! Haven’t tried it yet, but will soon.
This all sounds great! I have not tested it and to be honest am not sure how to do that? Is there a how-to that I am missing?
TL;DR:
- Call the dial in number
- Enter the conference pin number provided at the top of the room
- Press 0 to mute/unmute
The closest thing to docs we have right now is this post above.
I will work on better documentation soon.
Currently it is only on the meet.coop ca server. (dail-in for the de server will be active within the next week, just waiting for the phone number to be provisioned)
As of last night the call in details are listed at the top of the room in the CA server. Will do the same for DE when its ready.
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Yep, that’s my sense of things also. A lot may be down to ‘the last mile’ of the journey? Can we systematically engage with that (eg firm guidance on OK browser/OS combinations) or is it haphazard?
Thanks so much for doing this! So, my hope is that the dial-in number(s) will appear in the invite box. Will it require a separate sign-in/app for the SIP?
After a week of testing, last night we have just added the text CA server
We will add similar to the DE server as soon as the new number is provisioned
The sip option has two methods
Option one - Plain old telephone number that can be dialed with any phone (long distance charges apply see your carrier for details)
Option two - a sip URI that requires a sip phone like such as linphone.
current CA welcome message
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This just in.
Dial-in to Spain (Barcelona) at (34)932208649
Preliminary tests show it working. Waiting for members living in Europe to confirm!
I have updated the info in the original post with the European phone numbers. Once we confirm its working i will update the meeting message in BBB!
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@yurko, I just did a test session calling in through the BCN phone number. Works perfectly. I didn’t try to call in from a SIP account directly, that’d be the best obviously, as it would cost us a penny in dial-in calling minutes.
So now we have the instructions in @Yurko’s post above, linked also from the conf rooms. At a later stage we’ll need to move this to our documentation space, but for now it should do.
We should write a brief post on the OpenCollective Updates section to inform all members, and have them try Dial-in through a SIP application like Linphone or just over the normal telephone network (the latter costs us a cent per so many minutes per participants calling in).
This should be a solution for people on low bandwidth internet connections. Hope it really helps them.
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Awesome! thank you!
Not to worried about the SIP URI as i did my testing with that already. It shouldnt matter were you are in the world for that to work.
Also the dial in number for Barcelona is limited to 2 incoming calls. This was the easiest and fastest way to get it up and running. The SIP URI does not have this restriction
If we get people telling us they get busy signals we can get an additional number to offset that need. But this is a good start.
A post was merged into an existing topic: BigBlueButton 2.3-dev for Ubuntu 18.04 is reading for testing
Can we set it up so that admins can disable this feature under “organization settings”? In particular, we don’t want to have dial in support for our space, since dial in makes sessions less secure.
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Hmm interesting issue! Thanks for bringing this up.
There is no out-of-the-box solution for this. Let me think on this.
I think the easiest way would be to find a way to change the welcome message for your rooms, that way the conference pin number (that is randomly generated) will not appear.
Currently green light does not support this.
I don’t know if that’s sufficient, unless BBB has some bruteforce-detection mechanism that would keep attackers from guessing this (easy to guess) PIN…
We just tested this here and it works well for the dial-in part (on the phone number), so congratulations on that. The call quality was described as “excellent” by the caller (I haven’t tried it myself yet, from the phone side), but from the web side, they did sound a bit “old”, as if they were calling from an old landlines.
I hadn’t realized this, but 4G has this VoLTE thing that makes phone call sound much better, and it’s possible to leverage this on the VoIP side as well. You can get better audio by switching from the (possibly default) G711/PCMU codec to G722, for example. G.729a is also available at VoIP.ms but maybe not in your PBX.
I have also tried to call the SIP URI, and I’m getting a 403 Forbidden so that doesn’t seem to work. Having that working might enable some users to dial in without an actual phone or VoIP account, and could possible reduce your VoIP fees, but it could expose you to more attacks. Actually, this was a configuration problem on my end: I needed a “p2p” account in my sip client (regint=0 in baresip) for the calling to work…
Update: One more thing I noticed is that part of the phone number shows up in the UI. I’m not sure I like this, at all. Phone numbers are rather private information and divulging even a part of it seems like a really bad idea. The last 4 digits are particularly sensitive, as they are the most user-specific ones. Assuming you know a bit about me, you can certainly guess what my regional code is, so, combined with the last four digits, you have only 999 (and probably less) combinations to run through before you find my real phone number. Not great.
A better solution would be to show a random username or, even better, allow users to associate their phone call with their existing account. For example, when joining through the web, you could have a “join by phone” option next to the “listen only” and “join with audio” buttons, which would associate the call with your phone call normally, not routing it through the web browser, but still providing other functionality like video streaming and chat. I understand that this is an upstream feature change and do not expect you folks to fix that directly, but I thought I would mention it anyways.
Interestingly, it seems like the SIP interface has lower latency than the web one (in Chromium). In a sound check with a SIP phone and Chromium started, i see the phone icon light up before the web one. Not sure what the implications of that are, but I guess it is a good sign on the SIP implementation.
Otherwise, congratulations on the launch.
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