Trying to get droid search to work

.env.example

@@ -94,6 +94,12 @@ COLLABORA_PASSWORD=
 #
 PIHOLE_PASSWORD=
 
+#
+# Livekit settings (comms.scarif.space)
+#
+LIVEKIT_API_KEY=
+LIVEKIT_API_SECRET=
+
 #
 # Jitsi settings (comms.scarif.space)
 # Image: jitsi/web:latest

docker-compose.yml

@@ -6,7 +6,7 @@ x-logging: &logging
       max-file: "2"
 
 x-firecrawl-service: &firecrawl-service
-  image: ghcr.io/mendableai/firecrawl
+  image: ghcr.io/firecrawl/firecrawl
 
 x-firecrawl-env: &firecrawl-env
   REDIS_URL: redis://redis:6379
@@ -166,6 +166,7 @@ services:
       - db
 
   librechat:
+    profiles: ["prod", "dev"]
     image: ghcr.io/danny-avila/librechat-dev-api:latest
     ports:
       - 3080:3080
@@ -222,6 +223,7 @@ services:
       - /mnt/tower/stardust/files/chris/:/files
 
   searxng:
+    profiles: ["prod", "dev"]
     image: docker.io/searxng/searxng:latest
     restart: unless-stopped
     networks:
@@ -664,6 +666,29 @@ services:
         aliases:
           - jvb.meet.jitsi
 
+  livekit:
+    profiles: ["prod", "dev"]
+    <<: *logging
+    image: livekit/livekit-server:latest
+    container_name: livekit-server
+    restart: unless-stopped
+    ports:
+      - "7880:7880"
+      - "7881:7881"
+      - "7882:7882/udp"
+      - "3478:3478/udp"
+      - "5349:5349"
+    environment:
+      - VIRTUAL_HOST=comms-api.${DOMAIN}
+      - VIRTUAL_PORT=7880
+    volumes:
+      - ./livekit/livekit.yaml:/etc/livekit.yaml:ro
+    command: --dev --config /etc/livekit.yaml
+    networks:
+      - nginx
+      - redis
+
+
   db:
     profiles: ["prod", "dev"]
     <<: *logging
@@ -683,7 +708,7 @@ services:
   playwright:
     profiles: ["prod", "dev"]
     <<: *logging
-    image: ghcr.io/mendableai/playwright-service:latest
+    image: ghcr.io/firecrawl/playwright-service:latest
     environment:
       PORT: 3000
       BLOCK_MEDIA: true

livekit/Dockerfile

@@ -0,0 +1,40 @@
+FROM node:20-alpine as builder
+
+RUN npm install -g pnpm
+
+# Install necessary tools
+RUN apk add --no-cache git git-lfs sed
+
+# Fetch the application source code
+WORKDIR /app
+RUN git clone -b main https://github.com/livekit-examples/meet.git .
+
+# Add 'standalone' mode configuration to next.config.js
+RUN if ! grep -q "output: 'standalone'" next.config.js; then \
+  sed -i "/^const nextConfig = {/a \  output: 'standalone'," next.config.js; \
+  fi
+
+# Install dependencies
+RUN pnpm install
+
+# Set environment variables
+ENV NEXT_PUBLIC_LK_TOKEN_ENDPOINT=/api/token
+
+# Build the application
+RUN pnpm run build
+
+# Runtime stage
+FROM node:20-alpine
+WORKDIR /app
+
+# Copy built files and necessary resources
+COPY --from=builder /app/.next/standalone ./
+COPY --from=builder /app/.next/static ./.next/static
+COPY --from=builder /app/public ./public
+
+# Set environment variables
+ENV PORT 3000
+EXPOSE 3000
+
+# Start the application
+CMD ["node", "server.js"]

livekit/livekit.yaml

@@ -0,0 +1,322 @@
+# Copyright 2024 LiveKit, Inc.
+#
+# Licensed under the Apache License, Version 2.0 (the "License");
+# you may not use this file except in compliance with the License.
+# You may obtain a copy of the License at
+#
+#     http://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS,
+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+# See the License for the specific language governing permissions and
+# limitations under the License.
+
+# main TCP port for RoomService and RTC endpoint
+# for production setups, this port should be placed behind a load balancer with TLS
+port: 7880
+
+# when redis is set, LiveKit will automatically operate in a fully distributed fashion
+# clients could connect to any node and be routed to the same room
+redis:
+  address: redis:6379
+  # db: 0
+  # username: myuser
+  # password: mypassword
+  # To use sentinel remove the address key above and add the following
+  # sentinel_master_name: livekit
+  # sentinel_addresses:
+  # - livekit-redis-node-0.livekit-redis-headless:26379
+  # - livekit-redis-node-1.livekit-redis-headless:26379
+  # If you use a different set of credentials for sentinel add
+  # sentinel_username: user
+  # sentinel_password: pass
+  #
+  # To use TLS with redis
+  # tls:
+  #   enabled: true
+  #   # when set to true, LiveKit will not verify the server's certificate, defaults to true
+  #   insecure: false
+  #   server_name: myserver.com
+  #   # file containing trusted root certificates for verification
+  #   ca_cert_file: /path/to/ca.crt
+  #   client_cert_file: /path/to/client.crt
+  #   client_key_file: /path/to/client.key
+  #
+  # To use cluster remove the address key above and add the following
+  # cluster_addresses:
+  # - livekit-redis-node-0.livekit-redis-headless:6379
+  # - livekit-redis-node-1.livekit-redis-headless:6380
+  # And it will use the password key above as cluster password
+  # And the db key will not be used due to cluster mode not support it.
+
+# WebRTC configuration
+rtc:
+  # UDP ports to use for client traffic.
+  # this port range should be open for inbound traffic on the firewall
+  port_range_start: 50000
+  port_range_end: 60000
+  # when set, LiveKit enable WebRTC ICE over TCP when UDP isn't available
+  # this port *cannot* be behind load balancer or TLS, and must be exposed on the node
+  # WebRTC transports are encrypted and do not require additional encryption
+  # only 80/443 on public IP are allowed if less than 1024
+  tcp_port: 7881
+  # when set to true, attempts to discover the host's public IP via STUN
+  # this is useful for cloud environments such as AWS & Google where hosts have an internal IP
+  # that maps to an external one
+  use_external_ip: true
+  # # there are cases where the public IP determined via STUN is not the correct one
+  # # in such cases, use this setting to set the public IP of the node
+  # # use_external_ip takes precedence, for this to take effect, set use_external_ip to false
+  # node_ip: <external-ip-of-node>
+  # # when set, LiveKit will attempt to use a UDP mux so all UDP traffic goes through
+  # # listed port(s). To maximize system performance, we recommend using a range of ports
+  # # greater or equal to the number of vCPUs on the machine.
+  # # port_range_start & end must not be set for this config to take effect
+  # udp_port: 7882-7892
+  # # when set to true, server will use a lite ice agent, that will speed up ice connection, but
+  # # might cause connect issue if server running behind NAT.
+  # use_ice_lite: true
+  # # optional STUN servers for LiveKit clients to use. Clients will be configured to use these STUN servers automatically.
+  # # by default LiveKit clients use Google's public STUN servers
+  # stun_servers:
+  #   - server1
+  # # optional TURN servers for clients. This isn't necessary if using embedded TURN server (see below).
+  # turn_servers:
+  #   - host: myhost.com
+  #     port: 443
+  #     # tls, tcp, or udp
+  #     protocol: tls
+  #     username: ""
+  #     credential: ""
+  # # allows LiveKit to monitor congestion when sending streams and automatically
+  # # manage bandwidth utilization to avoid congestion/loss. Enabled by default
+  # congestion_control:
+  #   enabled: true
+  #   # in the unlikely event of highly congested networks, SFU may choose to pause some tracks
+  #   # in order to allow others to stream smoothly. You can disable this behavior here
+  #   allow_pause: true
+  # # allows automatic connection fallback to TCP and TURN/TLS (if configured) when UDP has been unstable, default true
+  # allow_tcp_fallback: true
+  # # number of packets to buffer in the SFU for video, defaults to 500
+  # packet_buffer_size_video: 500
+  # # number of packets to buffer in the SFU for audio, defaults to 200
+  # packet_buffer_size_audio: 200
+  # # minimum amount of time between pli/fir rtcp packets being sent to an individual
+  # # producer. Increasing these times can lead to longer black screens when new participants join,
+  # # while reducing them can lead to higher stream bitrate.
+  # pli_throttle:
+  #   low_quality: 500ms
+  #   mid_quality: 1s
+  #   high_quality: 1s
+  # # when set, Livekit will collect loopback candidates, it is useful for some VM have public address mapped to its loopback interface.
+  # enable_loopback_candidate: true
+  # # network interface filter. If the machine has more than one network interface and you'd like it to use or skip specific interfaces
+  # # both inclusion and exclusion filters can be used together. If neither is defined (default), all interfaces on the machine will be used.
+  # # If both of them are set, then only include takes effect.
+  # interfaces:
+  #   includes:
+  #     - en0
+  #   excludes:
+  #     - docker0
+  # # ip address filter. If the machine has more than one ip address and you'd like it to use or skip specific ips,
+  # # both inclusion and exclusion CIDR filters can be used together. If neither is defined (default), all ip on the machine will be used.
+  # # If both of them are set, then only include takes effect.
+  # ips:
+  #   includes:
+  #     - 10.0.0.0/16
+  #   excludes:
+  #     - 192.168.1.0/24
+  # # Set to true to enable mDNS name candidate. This should be left disabled for most users.
+  # # when enabled, it will impact performance since each PeerConnection will process the same mDNS message independently
+  # use_mdns: true
+  # # Set to false to disable strict ACKs for peer connections where LiveKit is the dialing side,
+  # # ie. subscriber peer connections. Disabling strict ACKs will prevent clients that do not ACK
+  # # peer connections from getting kicked out of rooms by the monitor. Note that if strict ACKs
+  # # are disabled and clients don't ACK opened peer connections, only reliable, ordered delivery
+  # # will be available.
+  # strict_acks: true
+  # # enable batch write to merge network write system calls to reduce cpu usage. Outgoing packets
+  # # will be queued until length of queue equal to `batch_size` or time elapsed since last write exceeds `max_flush_interval`.
+  # batch_io:
+  #    batch_size: 128
+  #    max_flush_interval: 2ms
+  # # max number of bytes to buffer for data channel. 0 means unlimited.
+  # # when this limit is breached, data messages will be dropped till the buffered amount drops below this limit.
+  # data_channel_max_buffered_amount: 0
+
+# when enabled, LiveKit will expose prometheus metrics on :6789/metrics
+# prometheus_port: 6789
+
+# API key / secret pairs.
+# Keys are used for JWT authentication, server APIs would require a keypair in order to generate access tokens
+# and make calls to the server
+keys:
+  7e510eafd39852bee31c4d5bfa87847e: 083ca120732019b2cf4a350d1c928173 
+# Logging config
+# logging:
+#   # log level, valid values: debug, info, warn, error
+#   level: info
+#   # log level for pion, default error
+#   pion_level: error
+#   # when set to true, emit json fields
+#   json: false
+#   # for production setups, enables sampling algorithm
+#   # https://github.com/uber-go/zap/blob/master/FAQ.md#why-sample-application-logs
+#   sample: false
+
+# Default room config
+# Each room created will inherit these settings. If rooms are created explicitly with CreateRoom, they will take
+# precedence over defaults
+# room:
+#   # allow rooms to be automatically created when participants join, defaults to true
+#   # auto_create: false
+#   # number of seconds to keep the room open if no one joins
+#   empty_timeout: 300
+#   # number of seconds to keep the room open after everyone leaves
+#   departure_timeout: 20
+#   # limit number of participants that can be in a room, 0 for no limit
+#   max_participants: 0
+#   # only accept specific codecs for clients publishing to this room
+#   # this is useful to standardize codecs across clients
+#   # other supported codecs are video/h264, video/vp9, video/av1, audio/red
+#   enabled_codecs:
+#     - mime: audio/opus
+#     - mime: video/vp8
+#   # allow tracks to be unmuted remotely, defaults to false
+#   # tracks can always be muted from the Room Service APIs
+#   enable_remote_unmute: true
+#   # control playout delay in ms of video track (and associated audio track)
+#   playout_delay:
+#     enabled: true
+#     min: 100
+#     max: 2000
+#   # improves A/V sync when playout_delay set to a value larger than 200ms. It will disables transceiver re-use
+#   # so not recommended for rooms with frequent subscription changes
+#   sync_streams: true
+
+# Webhooks
+# when configured, LiveKit notifies your URL handler with room events
+# webhook:
+#   # the API key to use in order to sign the message
+#   # this must match one of the keys LiveKit is configured with
+#   api_key: <api_key>
+#   # list of URLs to be notified of room events
+#   urls:
+#     - https://your-host.com/handler
+
+# Signal Relay
+# since v1.4.0, a more reliable, psrpc based signal relay is available
+# this gives us the ability to reliably proxy messages between a signal server and RTC node
+# signal_relay:
+#   # amount of time a message delivery is tried before giving up
+#   retry_timeout: 30s
+#   # minimum amount of time to wait for RTC node to ack,
+#   # retries use exponentially increasing wait on every subsequent try
+#   # with an upper bound of max_retry_interval
+#   min_retry_interval: 500ms
+#   # maximum amount of time to wait for RTC node to ack
+#   max_retry_interval: 5s
+#   # number of messages to buffer before dropping
+#   stream_buffer_size: 1000
+
+# PSRPC
+# since v1.5.1, a more reliable, psrpc based internal rpc
+# psrpc:
+#   # maximum number of rpc attempts
+#   max_attempts: 3
+#   # initial time to wait for calls to complete
+#   timeout: 500ms
+#   # amount of time added to the timeout after each failure
+#   backoff: 500ms
+#   # number of messages to buffer before dropping
+#   buffer_size: 1000
+
+# customize audio level sensitivity
+# audio:
+#   # minimum level to be considered active, 0-127, where 0 is loudest
+#   # defaults to 30
+#   active_level: 30
+#   # percentile to measure, a participant is considered active if it has exceeded the
+#   # ActiveLevel more than MinPercentile% of the time
+#   # defaults to 40
+#   min_percentile: 40
+#   # frequency in ms to notify changes to clients, defaults to 500
+#   update_interval: 500
+#   # to prevent speaker updates from too jumpy, smooth out values over N samples
+#   smooth_intervals: 4
+#   # enable red encoding downtrack for opus only audio up track
+#   active_red_encoding: true
+
+# turn server
+# turn:
+#   # Uses TLS. Requires cert and key pem files by either:
+#   # - using turn.secretName if deploying with our helm chart, or
+#   # - setting LIVEKIT_TURN_CERT and LIVEKIT_TURN_KEY env vars with file locations, or
+#   # - using cert_file and key_file below
+#   # defaults to false
+#   enabled: false
+#   # defaults to 3478 - recommended to 443 if not running HTTP3/QUIC server
+#   # only 53/80/443 are allowed if less than 1024
+#   udp_port: 3478
+#   # defaults to 5349 - if not using a load balancer, this must be set to 443
+#   tls_port: 5349
+#   # set UDP port range for TURN relay to connect to LiveKit SFU, by default it uses a any available port
+#   relay_range_start: 1024
+#   relay_range_end: 30000
+#   # set external_tls to true if using a L4 load balancer to terminate TLS. when enabled,
+#   # LiveKit expects unencrypted traffic on tls_port, and still advertise tls_port as a TURN/TLS candidate.
+#   external_tls: true
+#   # needs to match tls cert domain
+#   domain: turn.myhost.com
+#   # optional (set only if not using external TLS termination)
+#   # cert_file: /path/to/cert.pem
+#   # key_file: /path/to/key.pem
+
+# ingress server
+# ingress:
+#   # Prefix used to generate RTMP URLs for RTMP ingress.
+#   rtmp_base_url: "rtmp://my.domain.com/live"
+#   # Prefix used to generate WHIP URLs for WHIP ingress.
+#   whip_base_url: "http://my.domain.com/whip"
+
+# Region of the current node. Required if using regionaware node selector
+# region: us-west-2
+
+# # node selector
+# node_selector:
+#   # default: any. valid values: any, sysload, cpuload, regionaware
+#   kind: sysload
+#   # priority used for selection of node when multiple are available
+#   # default: random. valid values: random, sysload, cpuload, rooms, clients, tracks, bytespersec
+#   sort_by: sysload
+#   # used in sysload and regionaware
+#   # do not assign room to node if load per CPU exceeds sysload_limit
+#   sysload_limit: 0.7
+#   # used in regionaware
+#   # list of regions and their lat/lon coordinates
+#   regions:
+#     - name: us-west-2
+#       lat: 44.19434095976287
+#       lon: -123.0674908379146
+
+# # node limits
+# # set to -1 to disable a limit
+# limit:
+#   # defaults to 400 tracks in & out per CPU, up to 8000
+#   num_tracks: -1
+#   # defaults to 1 GB/s, or just under 10 Gbps
+#   bytes_per_sec: 1_000_000_000
+#   # how many tracks (audio / video) that a single participant can subscribe at same time.
+#   # if the limit is exceeded, subscriptions will be pending until any subscribed track has been unsubscribed.
+#   # value less or equal than 0 means no limit.
+#   subscription_limit_video: 0
+#   subscription_limit_audio: 0
+#   # limit size of room and participant's metadata, 0 for no limit
+#   max_metadata_size: 0
+#   # limit size of participant attributes, 0 for no limit
+#   max_attributes_size: 0
+#   # limit length of room names
+#   max_room_name_length: 0
+#   # limit length of participant identity
+#   max_participant_identity_length: 0