Quick VoIP roll-out #8
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| Original file line number | Diff line number | Diff line change |
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| services: | ||
| freeswitch: | ||
| image: docker.io/rocketchat/freeswitch:latest | ||
| # build: . | ||
| restart: always | ||
| environment: | ||
| EXTERNAL_ADDRESS_FQDN: ${VOIP_DOMAIN:-} | ||
| FREESWITCH_PASSWORD: ${FREESWITCH_PASSWORD:-} | ||
| RTP_START_PORT: 20000 | ||
| RTP_END_PORT: 20046 | ||
| DEBUG_MODE: "true" | ||
| ACCEPT_EXTERNAL_CALLS: false | ||
| USE_LEGACY_ARCH: false | ||
| DRACHTIO_DOMAIN: ${VOIP_DRACHTIO_DOMAIN:-drachtio} | ||
| expose: | ||
| - 5060 | ||
| ports: | ||
| - 20000-20046:20000-20046/udp | ||
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| drachtio: | ||
| image: drachtio/drachtio-server:latest | ||
| restart: always | ||
| ports: | ||
| - "9022:9022" | ||
| - "5060:5060/udp" | ||
| volumes: | ||
| # Drachtio params are configured on this file | ||
| # You must modify it to configure a strong and unique secret string | ||
| - ./files/drachtio/drachtio.conf.xml:/etc/drachtio.conf.xml | ||
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| <drachtio> | ||
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| <!-- udp port to listen on for client connections (default 9022), and shared secret used to authenticate clients --> | ||
| <admin port="9022" secret="set a strong secret here">0.0.0.0</admin> | ||
|
There was a problem hiding this comment. It would be best if we could configure drachtio without needing to change this file directly. I'll check what our options are and try to make something to avoid this. |
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| <!-- the server can either accept inbound connections from node apps, or make outbound requests | ||
| to node apps. You must globally choose one or the other approach but not both. | ||
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| Inbound connections requires no configuration other than the <admin/> tag above, and are useful | ||
| when a server is hosting a single application, or has one primary function. | ||
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| If, on the other hand, you want a server to support multiple applications, then we have the issue | ||
| of determining how to route an incoming request (INVITE, REGISTER, etc) to the appropriate application. | ||
| This is what outbound connections are designed for. | ||
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| By uncommenting the <request-handlers/> configuration below, you are placing the server into outbound mode, and | ||
| providing an HTTP URL that the server will hit to determine how to route an incoming request. | ||
| The following parameters will be passed to the http service as query string parameters in an HTTP GET: | ||
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| callingNumber - the user part of the From header | ||
| calledNumber - the user part of the To header | ||
| contentType - the Content-Type header (if any) | ||
| domain - the domain in the request uri | ||
| method - the sip method | ||
| protocol - the transport protocol | ||
| source_address - the sending address | ||
| uriUser - the user part of the sip uri | ||
| uri - the complete sip uri | ||
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| It is the responsibility of the HTTP server that receives the request to return a json response | ||
| with a top-level "action" of either "reject", "redirect", "proxy", or "route". | ||
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| reject - reject the call with a specified status code | ||
| redirect = redirect the call to one or more specified sip uris | ||
| proxy - proxy the call to one or more specified sip uris | ||
| route - connect to a nodejs application that is listening on the specified host:port | ||
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| example responses: | ||
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| { | ||
| "action": "reject": | ||
| "data": { | ||
| "status": 503, | ||
| "reason": "Max Calls Exceeded" | ||
| } | ||
| } | ||
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| { | ||
| "action": "redirect": | ||
| "data": { | ||
| "contacts": [ | ||
| "sip:foo@bar.com" | ||
| ] | ||
| } | ||
| } | ||
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| { | ||
| "action": "proxy": | ||
| "data": { | ||
| "destination": [ | ||
| "sip:foo@bar.com" | ||
| ], | ||
| recordRoute: true | ||
| } | ||
| } | ||
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| { | ||
| "action": "route": | ||
| "data": { | ||
| "url": "call-recording.default.svc.cluster.local:4000" | ||
| } | ||
| } | ||
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| { | ||
| "action": "route": | ||
| "data": { | ||
| "tag": "my-conferencing-app" | ||
| } | ||
| } | ||
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| --> | ||
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| <!-- comment this in and edit http url to use outbound connections | ||
| Note: currently only HTTP GET is supported as an HTTP METHOD | ||
| <request-handlers> | ||
| <request-handler sip-method="INVITE" http-method="GET">http://35.187.89.96:80</request-handler> | ||
| </request-handlers> | ||
| --> | ||
| <!-- sip configuration --> | ||
| <sip> | ||
| <contacts> | ||
| <!-- <contact>sip:*;transport=tcp</contact> --> | ||
| <contact>sip:*;transport=udp</contact> | ||
| <!-- local sip address to bind to. Default: 'sip:*', which means listens on port 5060 on all interfaces and transports --> | ||
| <!-- | ||
| Other examples: | ||
| sip:192.168.1.100 listen on a specific address | ||
| sip:*;transport=tcp tcp connections only | ||
| sip:*:5061 listen on port 5061 | ||
| --> | ||
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| <!-- here's how to listen for every protocol on their default ports --> | ||
| <!-- | ||
| <contact>sip:*:80;transport=ws</contact> | ||
| <contact>sips:*;transport=tls</contact> | ||
| <contact>sips:*:443;transport=wss</contact> | ||
| --> | ||
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| </contacts> | ||
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| <!-- uncomment and modify these if you want to change the sip stack timers (globally) | ||
| <timers> | ||
| <t1>500</t1> | ||
| <t2>4000</t2> | ||
| <t4>5000</t4> | ||
| <t1x64>32000</t1x64> | ||
| </timers> | ||
| --> | ||
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| <!-- uncomment this if you send encapsulated SIP messages to homer | ||
| <capture-server port="9060" hep-version="3" id="101">127.0.0.1</capture-server> | ||
| --> | ||
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| <!-- if you want to terminate SIP over TLS connections (example shows letsencrypt.org typical file locations, but use any CA) | ||
| <tls> | ||
| <key-file>/etc/letsencrypt/live/yourdomain/privkey.pem</key-file> | ||
| <cert-file>/etc/letsencrypt/live/yourdomain/cert.pem</cert-file> | ||
| <chain-file>/etc/letsencrypt/live/yourdomain/chain.pem</chain-file> | ||
| <dh-param>/var/local/dh4096.pem</dh-param> | ||
| </tls> | ||
| --> | ||
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| <!-- if you want to increase the default mtu size for udp packets; stack will force tcp when sending larger packets | ||
| <mtu-size>4096</mtu-size> | ||
| --> | ||
| <udp-mtu>8192</udp-mtu> | ||
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| <!-- uncommenting this will cause all outbound new requests (i.e. requests outside of a dialog) to go through the specified proxy --> | ||
| <!-- | ||
| <outbound-proxy>sip:10.10.10.1</outbound-proxy> | ||
| --> | ||
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| <!-- optional list of header values which, if encountered in a message will trigger an action | ||
| [tcp-]action="reject|discard" | ||
| reject means send a sip non-success response immediately | ||
| discard means sliently discard the request, generating no response | ||
| --> | ||
| <spammers action="reject" tcp-action="discard"> | ||
| <header name="User-Agent"> | ||
| <value>sip-cli</value> | ||
| <value>sipcli</value> | ||
| <value>friendly-scanner</value> | ||
| </header> | ||
| <header name="To"> | ||
| <value>sipvicious</value> | ||
| </header> | ||
| </spammers> | ||
| </sip> | ||
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| <!-- set to true if you want the server to cdr events to a connected client --> | ||
| <cdrs>false</cdrs> | ||
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| <monitoring> | ||
| <prometheus port="8088">127.0.0.1</prometheus> | ||
|
There was a problem hiding this comment. This should also be configured to work with the prometheus configured on the existing |
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| </monitoring> | ||
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| <!-- logging configuration --> | ||
| <logging> | ||
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| <!-- uncomment this if you want logging dumped to console, in addition to any other log destinations | ||
| --> | ||
| <console/> | ||
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| <!-- uncomment if you want logging sent to syslog (over UDP) --> | ||
| <!-- | ||
| <syslog> | ||
| <address>my.syslog.serve</address> | ||
| <port>514</port> | ||
| <facility>local6</facility> | ||
| </syslog> | ||
| --> | ||
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| <!-- file-based logging --> | ||
| <file> | ||
| <name>/var/log/drachtio/drachtio.log</name> | ||
| <archive>/var/log/drachtio/archive</archive> | ||
| <size>100</size> | ||
| <maxFiles>20</maxFiles> | ||
| <auto-flush>true</auto-flush> | ||
| </file> | ||
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| <!-- sofia (internal sip library) log level, from 0 (minimal) to 9 (verbose) --> | ||
| <sofia-loglevel>3</sofia-loglevel> | ||
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| <!-- general process log level: notice, warning, error, info, debug. Default: info --> | ||
| <loglevel>info</loglevel> | ||
| </logging> | ||
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| </drachtio> | ||
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Maybe we should allow customizing these ports via envvars on the .env file as well? Though we would need to set the same values on the port list as well.
I'm also not 100% sure if those should be the default values for those ports. This default list limits FS to 23 concurrent calls, which might be too small a number for production envs. We were using those values up to now because EKS has a limit of 50 open ports.
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Sure. Could we do this in the next iteration when we have a basic stack building repo ready? Ideally I dont want to compose this file here but instead, link it to the one from here i.e the VoiP server image repo. Please let me know once you push your PR #52. Then I can work on adding a reference instead of copying the file.