finish hooking up livekit

This commit is contained in:
Matthew Hodgson
2024-11-05 14:03:17 +00:00
parent 01b3f9f435
commit defa69734a
8 changed files with 426 additions and 11 deletions

View File

@@ -7,6 +7,10 @@ secrets:
file: secrets/postgres/postgres_password file: secrets/postgres/postgres_password
synapse_signing_key: synapse_signing_key:
file: secrets/synapse/${DOMAIN}.signing.key file: secrets/synapse/${DOMAIN}.signing.key
livekit_api_key:
file: secrets/livekit/livekit_api_key
livekit_secret_key:
file: secrets/livekit/livekit_secret_key
services: services:
# dependencies for optionally generating default configs + secrets # dependencies for optionally generating default configs + secrets
@@ -145,7 +149,7 @@ services:
redis: redis:
condition: service_started condition: service_started
postgres: postgres:
condition: service_healthy condition: service_started
init: init:
condition: service_completed_successfully condition: service_completed_successfully
@@ -208,7 +212,7 @@ services:
command: "server --config=/data/config.yaml" command: "server --config=/data/config.yaml"
depends_on: depends_on:
postgres: postgres:
condition: service_healthy condition: service_started
init: init:
condition: service_completed_successfully condition: service_completed_successfully
@@ -261,7 +265,14 @@ services:
command: --config /etc/livekit.yaml --node-ip ${LIVEKIT_NODE_IP} command: --config /etc/livekit.yaml --node-ip ${LIVEKIT_NODE_IP}
ports: ports:
# - 7880:7880 # HTTP listener # - 7880:7880 # HTTP listener
- 7881:7881 # WS signalling - 7881:7881 # TCP WebRTC transport, advertised via SDP
# TODO: expose livekit-turn on TCP & UDP 443 via nginx
# At least this would allow UDP turn on port 443 for better perf.
# You can't expose a massive range here as it literally sets up 10,000 userland listeners, which takes forever
# and will clash with any existing high-numbered ports.
# So for now, tunnel everything via TCP 7881. FIXME!
#- 50000-60000:50000-60000/tcp # TCP media #- 50000-60000:50000-60000/tcp # TCP media
#- 50000-60000:50000-60000/udp # UDP media #- 50000-60000:50000-60000/udp # UDP media
networks: networks:
@@ -271,3 +282,32 @@ services:
condition: service_completed_successfully condition: service_completed_successfully
redis: redis:
condition: service_started condition: service_started
livekit-jwt:
build:
# evil hack to pull in bash so we can run an entrypoint.sh
# FIXME: it's a bit wasteful; the alternative would be to modify lk-jwt-service to pick up secrets from disk
dockerfile_inline: |
FROM ghcr.io/element-hq/lk-jwt-service:latest-ci AS builder
FROM alpine:latest
RUN apk update && apk add bash
COPY --from=builder /lk-jwt-service /
restart: unless-stopped
volumes:
- ${VOLUME_PATH}/data/nginx/ssl/rootCA.pem:/etc/ssl/certs/ca-certificates.crt
- ${VOLUME_PATH}/init/livekit-jwt-entrypoint.sh:/entrypoint.sh
entrypoint: /entrypoint.sh
env_file: .env
deploy:
restart_policy:
condition: on-failure
networks:
- backend
secrets:
- livekit_api_key
- livekit_secret_key
depends_on:
init:
condition: service_completed_successfully
livekit:
condition: service_started

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@@ -0,0 +1,14 @@
{
"default_server_config": {
"m.homeserver": {
"base_url": "https://${ELEMENT_CALL_FQDN}",
"server_name": "${DOMAIN}"
}
},
"livekit": {
"livekit_service_url": "https://${LIVEKIT_JWT_FQDN}"
},
"features": {
"feature_use_device_session_member_events": true
}
}

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@@ -0,0 +1,321 @@
${CONFIG_HEADER}
# From https://github.com/livekit/livekit/blob/master/config-sample.yaml
# 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: false
# # 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:
${SECRETS_LIVEKIT_API_KEY}: '${SECRETS_LIVEKIT_SECRET_KEY}'
# 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

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@@ -3,6 +3,11 @@ ${CONFIG_HEADER}
# taken from https://element-hq.github.io/synapse/latest/reverse_proxy.html # taken from https://element-hq.github.io/synapse/latest/reverse_proxy.html
# mixed with https://github.com/wmnnd/nginx-certbot/tree/master/etc/nginx/conf.d/nginx # mixed with https://github.com/wmnnd/nginx-certbot/tree/master/etc/nginx/conf.d/nginx
# log_format vhosts '$host $remote_addr - $remote_user [$time_local] '
# '"$request" $status $body_bytes_sent '
# '"$http_referer" "$http_user_agent"';
# access_log /dev/stdout vhosts;
server { server {
server_name ${DOMAIN}; server_name ${DOMAIN};
server_tokens off; server_tokens off;
@@ -92,6 +97,22 @@ server {
location / { location / {
proxy_pass http://livekit:7880; proxy_pass http://livekit:7880;
proxy_http_version 1.1;
proxy_set_header Upgrade ${DOLLAR}http_upgrade;
proxy_set_header Connection "upgrade";
proxy_set_header Host ${DOLLAR}host;
proxy_set_header X-Forwarded-For ${DOLLAR}remote_addr;
}
}
server {
server_name ${LIVEKIT_JWT_FQDN};
server_tokens off;
include /etc/nginx/conf.d/include/ssl.conf;
location / {
proxy_pass http://livekit-jwt:8080;
proxy_set_header X-Forwarded-For ${DOLLAR}remote_addr; proxy_set_header X-Forwarded-For ${DOLLAR}remote_addr;
} }
} }

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@@ -12,7 +12,7 @@
"org.matrix.msc4143.rtc_foci": [ "org.matrix.msc4143.rtc_foci": [
{ {
"type": "livekit", "type": "livekit",
"livekit_service_url": "https://${LIVEKIT_FQDN}" "livekit_service_url": "https://${LIVEKIT_JWT_FQDN}"
} }
] ]
} }

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@@ -102,4 +102,13 @@ experimental_features:
admin_token: '${SECRETS_MAS_MATRIX_SECRET}' admin_token: '${SECRETS_MAS_MATRIX_SECRET}'
account_management_url: "https://${MAS_FQDN}/account" account_management_url: "https://${MAS_FQDN}/account"
# MSC3266: Room summary API. Used for knocking over federation
msc3266_enabled: true
# The maximum allowed duration by which sent events can be delayed, as
# per MSC4140. Must be a positive value if set. Defaults to no
# duration (null), which disallows sending delayed events.
# Needed for MatrixRTC
max_event_delay_duration: 24h
# vim:ft=yaml # vim:ft=yaml

10
init/livekit-jwt-entrypoint.sh Executable file
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@@ -0,0 +1,10 @@
#!/bin/bash
# launch lk-jwt-service with secrets from disk
export LK_JWT_PORT=8080
export LIVEKIT_URL=wss://${LIVEKIT_FQDN}
export LIVEKIT_KEY=$(</run/secrets/livekit_api_key)
export LIVEKIT_SECRET=$(</run/secrets/livekit_secret_key)
exec /lk-jwt-service