Jitsi会议脚本和配置

#Sample config with one XMPP and one SIP account configured # Replace {sip-pass-hash} with SIP user password hash # as well as other account properties # Name of default JVB room that will be joined if no special header is included # in SIP invite org.jitsi.jigasi.DEFAULT_JVB_ROOM_NAME=siptest net.java.sip.communicator.impl.protocol.SingleCallInProgressPolicy.enabled=false # Should be enabled when using translator mode #net.java.sip.communicator.impl.neomedia.audioSystem.audiosilence.captureDevice_list=["AudioSilenceCaptureDevice:noTransferData"] # Adjust opus encoder complexity net.java.sip.communicator.impl.neomedia.codec.audio.opus.encoder.COMPLEXITY=10 # Disables packet logging net.java.sip.communicator.packetlogging.PACKET_LOGGING_ENABLED=true net.java.sip.communicator.impl.protocol.sip.acc1403273890647=acc1403273890647 net.java.sip.communicator.impl.protocol.sip.acc1403273890647.ACCOUNT_UID=SIP\:1111@10.10.2.123 net.java.sip.communicator.impl.protocol.sip.acc1403273890647.PASSWORD=MTIzNyMQ== net.java.sip.communicator.impl.protocol.sip.acc1403273890647.PROTOCOL_NAME=SIP net.java.sip.communicator.impl.protocol.sip.acc1403273890647.SERVER_ADDRESS=10.10.2.123 net.java.sip.communicator.impl.protocol.sip.acc1403273890647.USER_ID=1111@10.10.2.123 net.java.sip.communicator.impl.protocol.sip.acc1403273890647.KEEP_ALIVE_INTERVAL=25 net.java.sip.communicator.impl.protocol.sip.acc1403273890647.KEEP_ALIVE_METHOD=OPTIONS net.java.sip.communicator.impl.protocol.sip.acc1403273890647.VOICEMAIL_ENABLED=false net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.AMR-WB/16000=750 net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.G722/8000=700 net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.GSM/8000=0 net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.H263-1998/90000=0 net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.H264/90000=0 net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.PCMA/8000=600 net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.PCMU/8000=650 net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.SILK/12000=0 net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.SILK/16000=0 net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.SILK/24000=0 net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.SILK/8000=0 net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.VP8/90000=0 net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.iLBC/8000=10 net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.opus/48000=1000 net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.red/90000=0 net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.speex/16000=0 net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.speex/32000=0 net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.speex/8000=0 net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.telephone-event/8000=1 net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.ulpfec/90000=0 net.java.sip.communicator.impl.protocol.sip.acc1403273890647.OVERRIDE_ENCODINGS=true net.java.sip.communicator.impl.protocol.sip.acc1403273890647.DEFAULT_ENCRYPTION=false # If an authenticated (hidden) domain is used to connect to a conference, # PREVENT_AUTH_LOGIN will prevent the SIP participant from being seen as a # hidden participant in the conference #net.java.sip.communicator.impl.protocol.sip.acc1403273890647.PREVENT_AUTH_LOGIN=FALSE # Used when incoming calls are used in multidomain environment, used to detect subdomains # used for constructing callResource and eventually contacting jicofo net.java.sip.communicator.impl.protocol.sip.acc1403273890647.DOMAIN_BASE=videochat.test.com.cn # the pattern to be used as bosh url when using bosh in multidomain environment #net.java.sip.communicator.impl.protocol.sip.acc1403273890647.BOSH_URL_PATTERN=https://{host}{subdomain}/http-bind?room={roomName} # can be enabled to disable audio mixing and use translator, jigasi will act as jvb, just forward every ssrc stream it receives. #net.java.sip.communicator.impl.protocol.sip.acc1403273890647.USE_TRANSLATOR_IN_CONFERENCE=true # We can use the prefix org.jitsi.jigasi.xmpp.acc to override any of the # properties that will be used for creating xmpp account for communication. # The following two props assume we are using jigasi on the same machine as # the xmpp server. org.jitsi.jigasi.xmpp.acc.IS_SERVER_OVERRIDDEN=true org.jitsi.jigasi.xmpp.acc.SERVER_ADDRESS=127.0.0.1 org.jitsi.jigasi.xmpp.acc.VIDEO_CALLING_DISABLED=true org.jitsi.jigasi.xmpp.acc.JINGLE_NODES_ENABLED=false org.jitsi.jigasi.xmpp.acc.AUTO_DISCOVER_STUN=false org.jitsi.jigasi.xmpp.acc.IM_DISABLED=true org.jitsi.jigasi.xmpp.acc.SERVER_STORED_INFO_DISABLED=true org.jitsi.jigasi.xmpp.acc.IS_FILE_TRANSFER_DISABLED=true # Or you can use bosh for the connection establishment by specifing the URL to use. # org.jitsi.jigasi.xmpp.acc.BOSH_URL_PATTERN=https://server.com/http-bind?room={roomName} #Used when outgoing calls are used in multidomain environment, used to detect subdomains #org.jitsi.jigasi.xmpp.acc.DOMAIN_BASE=videochat.test.com.cn #org.jitsi.jigasi.xmpp.acc.BOSH_URL_PATTERN=https://{host}{subdomain}/http-bind?room={roomName} # can be enabled to disable audio mixing and use translator, jigasi will act as jvb, just forward every ssrc stream it receives. #org.jitsi.jigasi.xmpp.acc.USE_TRANSLATOR_IN_CONFERENCE=true # If you want jigasi to perform authenticated login instead of anonymous login # to the XMPP server, you can set the following properties. org.jitsi.jigasi.xmpp.acc.USER_ID=jigasi@auth.videochat.test.com.cn org.jitsi.jigasi.xmpp.acc.PASS=jigasi123456 org.jitsi.jigasi.xmpp.acc.ANONYMOUS_AUTH=false org.jitsi.jigasi.MUC_SERVICE_ADDRESS=conference.videochat.test.com.cn org.jitsi.jigasi.xmpp.acc.ALLOW_NON_SECURE=true #org.jitsi.jigasi.xmpp.acc.IS_SERVER_OVERRIDDEN=true #org.jitsi.jigasi.xmpp.acc.SERVER_ADDRESS=xmpp.example.com # If you want to use the SIP user part of the incoming/outgoing call SIP URI # you can set the following property to true. # org.jitsi.jigasi.USE_SIP_USER_AS_XMPP_RESOURCE=true # Activate this property if you are using self-signed certificates or other # type of non-trusted certicates. In this mode your service trust in the # remote certificates always. net.java.sip.communicator.service.gui.ALWAYS_TRUST_MODE_ENABLED=true # Enable this property to be able to shutdown gracefully jigasi using # a rest command # org.jitsi.jigasi.ENABLE_REST_SHUTDOWN=true # Options regarding Transcription. Read the README for a detailed description # about each property #org.jitsi.jigasi.ENABLE_TRANSCRIPTION=false org.jitsi.jigasi.ENABLE_SIP=true # whether to use the more expensive, but better performing # "video" model when doing transcription # org.jitsi.jigasi.transcription.USE_VIDEO_MODEL = false # delivering final transcript # org.jitsi.jigasi.transcription.DIRECTORY=/var/lib/jigasi/transcripts # org.jitsi.jigasi.transcription.BASE_URL=http://localhost/ # org.jitsi.jigasi.transcription.jetty.port=-1 # org.jitsi.jigasi.transcription.ADVERTISE_URL=false # save formats # org.jitsi.jigasi.transcription.SAVE_JSON=false # org.jitsi.jigasi.transcription.SAVE_TXT=true # send formats # org.jitsi.jigasi.transcription.SEND_JSON=true # org.jitsi.jigasi.transcription.SEND_TXT=false # translation # org.jitsi.jigasi.transcription.ENABLE_TRANSLATION=false # record audio. Currently only wav format is supported # org.jitsi.jigasi.transcription.RECORD_AUDIO=false # org.jitsi.jigasi.transcription.RECORD_AUDIO_FORMAT=wav # execute one or more scripts when a transcript or recording is saved # org.jitsi.jigasi.transcription.EXECUTE_SCRIPTS=true # org.jitsi.jigasi.transcription.SCRIPTS_TO_EXECUTE_LIST_SEPARATOR=","