FS-7471 improve configs for video
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@ -226,6 +226,41 @@
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</profile>
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<profile name="video-mcu-stereo">
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<param name="domain" value="$${domain}"/>
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<param name="rate" value="48000"/>
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<param name="channels" value="2"/>
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<param name="interval" value="20"/>
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<param name="energy-level" value="200"/>
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<!-- <param name="tts-engine" value="flite"/> -->
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<!-- <param name="tts-voice" value="kal16"/> -->
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<param name="muted-sound" value="conference/conf-muted.wav"/>
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<param name="unmuted-sound" value="conference/conf-unmuted.wav"/>
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<param name="alone-sound" value="conference/conf-alone.wav"/>
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<param name="moh-sound" value="local_stream://stereo"/>
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<param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/>
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<param name="exit-sound" value="tone_stream://%(500,0,300,200,100,50,25)"/>
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<param name="kicked-sound" value="conference/conf-kicked.wav"/>
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<param name="locked-sound" value="conference/conf-locked.wav"/>
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<param name="is-locked-sound" value="conference/conf-is-locked.wav"/>
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<param name="is-unlocked-sound" value="conference/conf-is-unlocked.wav"/>
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<param name="pin-sound" value="conference/conf-pin.wav"/>
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<param name="bad-pin-sound" value="conference/conf-bad-pin.wav"/>
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<param name="caller-id-name" value="$${outbound_caller_name}"/>
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<param name="caller-id-number" value="$${outbound_caller_id}"/>
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<param name="comfort-noise" value="false"/>
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<param name="conference-flags" value="video-floor-only|rfc-4579|livearray-sync|minimize-video-encoding"/>
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<param name="video-mode" value="mux"/>
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<param name="video-layout-name" value="3x3"/>
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<param name="video-layout-name" value="group:grid"/>
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<param name="video-canvas-size" value="1920x1080"/>
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<param name="video-canvas-bgcolor" value="#333333"/>
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<param name="video-layout-bgcolor" value="#000000"/>
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<param name="video-codec-bandwidth" value="1mb"/>
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<param name="video-fps" value="15"/>
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</profile>
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<profile name="sla">
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<param name="domain" value="$${domain}"/>
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<param name="rate" value="16000"/>
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@ -43,7 +43,9 @@
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<!-- <load module="mod_unicall"/> -->
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<!-- <load module="mod_skinny"/> -->
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<!-- <load module="mod_khomp"/> -->
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<load module="mod_rtc"/>
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<!-- <load module="mod_rtmp"/> -->
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<load module="mod_verto"/>
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<!-- Applications -->
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<load module="mod_commands"/>
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@ -8,6 +8,7 @@
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<profile name="mine">
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<param name="bind-local" value="0.0.0.0:8081"/>
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<param name="bind-local" value="0.0.0.0:8082" secure="true"/>
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<param name="force-register-domain" value="$${domain}"/>
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<param name="secure-combined" value="$${certs_dir}/wss.pem"/>
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<param name="secure-chain" value="$${certs_dir}/wss.pem"/>
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<param name="userauth" value="true"/>
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@ -380,7 +380,15 @@
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<action application="conference" data="$1-${domain_name}@cdquality"/>
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</condition>
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</extension>
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<!-- STEREO 48kHz conferences / Video MCU -->
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<extension name="cdquality_conferences">
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<condition field="destination_number" expression="^(35\d{2})$">
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<action application="answer"/>
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<action application="conference" data="$1-${domain_name}@video-mcu-stereo"/>
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</condition>
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</extension>
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<!-- dial the FreeSWITCH conference via SIP-->
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<extension name="freeswitch_public_conf_via_sip">
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<condition field="destination_number" expression="^9(888|8888|1616|3232)$">
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@ -23,7 +23,7 @@
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<params>
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<param name="dial-string" value="{^^:sip_invite_domain=${dialed_domain}:presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(*/${dialed_user}@${dialed_domain})}"/>
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<!-- These are required for Verto to function properly -->
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<!-- <param name="jsonrpc-allowed-methods" value="verto"/> -->
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<param name="jsonrpc-allowed-methods" value="verto"/>
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<!-- <param name="jsonrpc-allowed-event-channels" value="demo,conference,presence"/> -->
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</params>
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@ -305,13 +305,12 @@
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<param name="force-register-db-domain" value="$${domain}"/>
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<!-- uncomment for sip over websocket support -->
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<!--<param name="ws-binding" value=":5066"/>-->
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<!-- for sip over websocket support -->
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<param name="ws-binding" value=":5066"/>
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<!-- uncomment for sip over secure websocket support -->
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<!-- You need wss.pem in $${certs_dir} for wss -->
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<!--<param name="wss-binding" value=":7443"/>-->
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<!-- for sip over secure websocket support -->
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<!-- You need wss.pem in $${certs_dir} for wss or one will be created for you -->
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<param name="wss-binding" value=":7443"/>
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<!--<param name="delete-subs-on-register" value="false"/>-->
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@ -255,8 +255,8 @@
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127 - BV32
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-->
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<X-PRE-PROCESS cmd="set" data="global_codec_prefs=OPUS,G722,PCMU,PCMA,GSM"/>
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<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
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<X-PRE-PROCESS cmd="set" data="global_codec_prefs=OPUS,G722,PCMU,PCMA,VP8"/>
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<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=OPUS,G722,PCMU,PCMA,VP8"/>
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<!--
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xmpp_client_profile and xmpp_server_profile
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@ -424,11 +424,23 @@
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<X-PRE-PROCESS cmd="set" data="internal_tls_port=5061"/>
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<X-PRE-PROCESS cmd="set" data="internal_ssl_enable=false"/>
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<!-- External SIP Profile -->
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<X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/>
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<X-PRE-PROCESS cmd="set" data="external_sip_port=5080"/>
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<X-PRE-PROCESS cmd="set" data="external_tls_port=5081"/>
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<X-PRE-PROCESS cmd="set" data="external_ssl_enable=false"/>
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<!-- Video Settings -->
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<!-- Setting the max bandwdith -->
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<X-PRE-PROCESS cmd="set" data="rtp_video_max_bandwidth_in=1mb"/>
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<X-PRE-PROCESS cmd="set" data="rtp_video_max_bandwidth_out=1mb"/>
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<!-- WebRTC Video -->
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<!-- Suppress CNG for WebRTC Audio -->
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<X-PRE-PROCESS cmd="set" data="suppress_cng=true"/>
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<!-- Enable liberal DTMF for those that can't get it right -->
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<X-PRE-PROCESS cmd="set" data="rtp_liberal_dtmf=true"/>
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<!-- Helps with WebRTC Audio -->
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<X-PRE-PROCESS cmd="set" data="answer_delay=4000"/>
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</include>
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