<aside> ⚠️
The instructions below will break FS-based dial-in for the whole BBB server. To interop both LK and FS, the call gateway needs to be aware of the audio bridge being used.
Example:
Your call gateway manager knows that a server has audioBridge=livekit
or that /create has audioBridge=livekit
.
You'd keep FS in 5060 and livekit-sip in 5062 in BBB servers.
LB then creates the call gateway's dialplan redirecting to the correct port (5060
if audioBridge=bbb-webrtc-sfu
, 5062
if audioBridge=livekit
).
If that approach is implemented, adjust the instructions below accordingly.
</aside>
Call gateway (the external FS) should behave roughly the same as it does today in, e.g., Scalelite’s scenario: https://medium.com/@JesusFederico/scalelite-and-dial-in-numbers-f070fe0059b0
In /opt/freeswitch/conf/vars.xml
: move BBB's FS SIP bind port to 5063
(to prevent collision with livekit-sip)
<!--/opt/freeswitch/conf/vars.xml-->
<X-PRE-PROCESS cmd="set" data="external_sip_port=5063"/>
In /etc/bigbluebutton/livekit-sip.yaml
: set sip_port
and sip_port_listen
to 5060
In /etc/bigbluebutton/livekit-sip.yaml
: set rtp_port.port_range_start
and
rtp_port.port_range_end
to a valid UDP range that has connectivity to
the call gateway
In /etc/bigbluebutton/bbb-webrtc-sfu/production.yml
:
# Merge with pre-existing configuration
livekit:
rtcAgent:
enabled: true
sip:
enabled: true
requirePin: false
dispatch:
options:
hidePhoneNumber: true
Restart BBB: $ sudo bbb-conf --restart
Restart livekit-sip
: $ sudo systemctl restart livekit-sip
livekit-sip
is not yet part of bigbluebutton.target
, so it needs to be restarted directly.Verify all components are running via bbb-conf --status
and sudo systemctl status livekit-sip
List of known issues. They’ll be checked when resolved.
*
isn’t muting/unmuting the endpointlivekit.sip.dispatch.options.hidePhoneNumber
, but it’s looking for livekit.sip.dispatch.hidePhoneNumber
as of 3.0.13)