FS-7471 improve configs for video

This commit is contained in:
Brian 2015-04-24 11:18:21 -05:00 committed by Michael Jerris
parent 22a4a4dd7e
commit d3a5605ab6
7 changed files with 68 additions and 11 deletions

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@ -226,6 +226,41 @@
</profile>
<profile name="video-mcu-stereo">
<param name="domain" value="$${domain}"/>
<param name="rate" value="48000"/>
<param name="channels" value="2"/>
<param name="interval" value="20"/>
<param name="energy-level" value="200"/>
<!-- <param name="tts-engine" value="flite"/> -->
<!-- <param name="tts-voice" value="kal16"/> -->
<param name="muted-sound" value="conference/conf-muted.wav"/>
<param name="unmuted-sound" value="conference/conf-unmuted.wav"/>
<param name="alone-sound" value="conference/conf-alone.wav"/>
<param name="moh-sound" value="local_stream://stereo"/>
<param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/>
<param name="exit-sound" value="tone_stream://%(500,0,300,200,100,50,25)"/>
<param name="kicked-sound" value="conference/conf-kicked.wav"/>
<param name="locked-sound" value="conference/conf-locked.wav"/>
<param name="is-locked-sound" value="conference/conf-is-locked.wav"/>
<param name="is-unlocked-sound" value="conference/conf-is-unlocked.wav"/>
<param name="pin-sound" value="conference/conf-pin.wav"/>
<param name="bad-pin-sound" value="conference/conf-bad-pin.wav"/>
<param name="caller-id-name" value="$${outbound_caller_name}"/>
<param name="caller-id-number" value="$${outbound_caller_id}"/>
<param name="comfort-noise" value="false"/>
<param name="conference-flags" value="video-floor-only|rfc-4579|livearray-sync|minimize-video-encoding"/>
<param name="video-mode" value="mux"/>
<param name="video-layout-name" value="3x3"/>
<param name="video-layout-name" value="group:grid"/>
<param name="video-canvas-size" value="1920x1080"/>
<param name="video-canvas-bgcolor" value="#333333"/>
<param name="video-layout-bgcolor" value="#000000"/>
<param name="video-codec-bandwidth" value="1mb"/>
<param name="video-fps" value="15"/>
</profile>
<profile name="sla">
<param name="domain" value="$${domain}"/>
<param name="rate" value="16000"/>

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@ -43,7 +43,9 @@
<!-- <load module="mod_unicall"/> -->
<!-- <load module="mod_skinny"/> -->
<!-- <load module="mod_khomp"/> -->
<load module="mod_rtc"/>
<!-- <load module="mod_rtmp"/> -->
<load module="mod_verto"/>
<!-- Applications -->
<load module="mod_commands"/>

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@ -8,6 +8,7 @@
<profile name="mine">
<param name="bind-local" value="0.0.0.0:8081"/>
<param name="bind-local" value="0.0.0.0:8082" secure="true"/>
<param name="force-register-domain" value="$${domain}"/>
<param name="secure-combined" value="$${certs_dir}/wss.pem"/>
<param name="secure-chain" value="$${certs_dir}/wss.pem"/>
<param name="userauth" value="true"/>

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@ -380,7 +380,15 @@
<action application="conference" data="$1-${domain_name}@cdquality"/>
</condition>
</extension>
<!-- STEREO 48kHz conferences / Video MCU -->
<extension name="cdquality_conferences">
<condition field="destination_number" expression="^(35\d{2})$">
<action application="answer"/>
<action application="conference" data="$1-${domain_name}@video-mcu-stereo"/>
</condition>
</extension>
<!-- dial the FreeSWITCH conference via SIP-->
<extension name="freeswitch_public_conf_via_sip">
<condition field="destination_number" expression="^9(888|8888|1616|3232)$">

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@ -23,7 +23,7 @@
<params>
<param name="dial-string" value="{^^:sip_invite_domain=${dialed_domain}:presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(*/${dialed_user}@${dialed_domain})}"/>
<!-- These are required for Verto to function properly -->
<!-- <param name="jsonrpc-allowed-methods" value="verto"/> -->
<param name="jsonrpc-allowed-methods" value="verto"/>
<!-- <param name="jsonrpc-allowed-event-channels" value="demo,conference,presence"/> -->
</params>

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@ -305,13 +305,12 @@
<param name="force-register-db-domain" value="$${domain}"/>
<!-- uncomment for sip over websocket support -->
<!--<param name="ws-binding" value=":5066"/>-->
<!-- for sip over websocket support -->
<param name="ws-binding" value=":5066"/>
<!-- uncomment for sip over secure websocket support -->
<!-- You need wss.pem in $${certs_dir} for wss -->
<!--<param name="wss-binding" value=":7443"/>-->
<!-- for sip over secure websocket support -->
<!-- You need wss.pem in $${certs_dir} for wss or one will be created for you -->
<param name="wss-binding" value=":7443"/>
<!--<param name="delete-subs-on-register" value="false"/>-->

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@ -255,8 +255,8 @@
127 - BV32
-->
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=OPUS,G722,PCMU,PCMA,GSM"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=OPUS,G722,PCMU,PCMA,VP8"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=OPUS,G722,PCMU,PCMA,VP8"/>
<!--
xmpp_client_profile and xmpp_server_profile
@ -424,11 +424,23 @@
<X-PRE-PROCESS cmd="set" data="internal_tls_port=5061"/>
<X-PRE-PROCESS cmd="set" data="internal_ssl_enable=false"/>
<!-- External SIP Profile -->
<X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/>
<X-PRE-PROCESS cmd="set" data="external_sip_port=5080"/>
<X-PRE-PROCESS cmd="set" data="external_tls_port=5081"/>
<X-PRE-PROCESS cmd="set" data="external_ssl_enable=false"/>
<!-- Video Settings -->
<!-- Setting the max bandwdith -->
<X-PRE-PROCESS cmd="set" data="rtp_video_max_bandwidth_in=1mb"/>
<X-PRE-PROCESS cmd="set" data="rtp_video_max_bandwidth_out=1mb"/>
<!-- WebRTC Video -->
<!-- Suppress CNG for WebRTC Audio -->
<X-PRE-PROCESS cmd="set" data="suppress_cng=true"/>
<!-- Enable liberal DTMF for those that can't get it right -->
<X-PRE-PROCESS cmd="set" data="rtp_liberal_dtmf=true"/>
<!-- Helps with WebRTC Audio -->
<X-PRE-PROCESS cmd="set" data="answer_delay=4000"/>
</include>