325887e325
This is a rework of the audio join procedure whithout the explict listen
only separation in mind. It's supposed to be used in conjunction with
the transparent listen only feature so that the distinction between
modes is seamless with minimal server-side impact. An abridged list of
changes:
- Let the user pick no input device when joining microphone while
allowing them to set an input device on the fly later on
- Give the user the option to join audio with no input device whenever
we fail to obtain input devices, with the option to try re-enabling
them on the fly later on
- Add the option to open the audio settings modal (echo test et al)
via the in-call device selection chevron
- Rework the SFU audio bridge and its services to support
adding/removing tracks on the fly without renegotiation
- Rework the SFU audio bridge and its services to support a new peer
role called "passive-sendrecv". That role is used by dupled peers
that have no active input source on start, but might have one later
on.
- Remove stale PermissionsOverlay component from the audio modal
- Rework how permission errors are detected using the Permissions API
- Rework the local echo test so that it uses a separate media tag
rather than the remote
- Add new, separate dialplans that mute/hold FreeSWITCH channels on
hold based on UA strings. This is orchestrated server-side via
webrtc-sfu and akka-apps. The basic difference here is that channels
now join in their desired state rather than waiting for client side
observers to sync the state up. It also mitigates transparent listen
only performance edge cases on multiple audio channels joining at
the same time.
The old, decoupled listen only mode is still present in code while we
validate this new approach. To test this, transparentListenOnly
must be enabled and listen only mode must be disable on audio join so
that the user skips straight through microphone join.
35 lines
1.9 KiB
XML
35 lines
1.9 KiB
XML
<include>
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<extension name="bbb_conferences_ws">
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<condition field="${bbb_authorized}" expression="true" break="on-false" />
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<condition field="${sip_via_protocol}" expression="^wss?$" />
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<condition field="destination_number" expression="^(\d{5,11})$">
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<action application="set" data="jitterbuffer_msec=100:250" />
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<action application="set" data="rtp_jitter_buffer_plc=true" />
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<action application="set" data="rtp_jitter_buffer_during_bridge=true" />
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<action application="set" data="suppress_cng=true" />
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<action application="answer" />
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<!-- Special condition for BBB's "transparent listen only" mechanism - HOLD on creation -->
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<condition field="${sip_user_agent}" expression="^bbb-webrtc-sfu-mhos$" break="never">
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<action application="log" data="INFO Channel is going to be HELD and MUTED on creation ${uuid}" />
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<action application="set" data="api_result=${uuid_hold(${uuid})}" />
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</condition>
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<!-- Duplicate condition to guarantee line-order (not nested-order) execution of this extension -->
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<condition field="destination_number" expression="^(\d{5,11})$" require-nested="false">
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<action application="conference" data="$1@cdquality" />
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</condition>
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</condition>
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</extension>
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<extension name="bbb_conferences">
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<condition field="${bbb_authorized}" expression="true" break="on-false" />
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<condition field="destination_number" expression="^(\d{5,11})$">
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<action application="set" data="jitterbuffer_msec=100:250" />
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<action application="set" data="rtp_jitter_buffer_plc=true" />
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<action application="set" data="rtp_jitter_buffer_during_bridge=true" />
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<action application="set" data="suppress_cng=true" />
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<action application="answer" />
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<action application="conference" data="$1@cdquality" />
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</condition>
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</extension>
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</include>
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