forked from mirrors/linphone-iphone
625 lines
20 KiB
Text
625 lines
20 KiB
Text
##
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## This is the default Flexisip configuration file
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##
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##
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## Some global settings of the flexisip proxy.
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##
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[global]
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# Outputs very detailed logs
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# Default value: false
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debug=1
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# Automatically respawn flexisip in case of abnormal termination
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# (crashes)
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# Default value: true
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auto-respawn=true
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# List of white space separated host names pointing to this machine.
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# This is to prevent loops while routing SIP messages.
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# Default value: localhost
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aliases=localhost sip2.linphone.org sipopen.example.org sip.example.org auth.example.org auth1.example.org auth2.example.org client.example.org sipv4.example.org sipv4-nat64.example.org
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# List of white space separated SIP uris where the proxy must listen.Wildcard
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# (*) can be used to mean 'all local ip addresses'. If 'transport'
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# prameter is unspecified, it will listen to both udp and tcp. An
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# local address to bind can be indicated in the 'maddr' parameter,
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# while the domain part of the uris are used as public domain or
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# ip address. Here some examples to understand:
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# * listen on all local interfaces for udp and tcp, on standart
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# port:
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# transports=sip:*
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# * listen on all local interfaces for udp,tcp and tls, on standart
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# ports:
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# transports=sip:* sips:*
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# * listen on 192.168.0.29:6060 with tls, but public hostname is
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# 'sip.linphone.org' used in SIP messages. Bind address won't appear:
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# transports=sips:sip.linphone.org:6060;maddr=192.168.0.29
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# Default value: sip:*
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#transports=sip:192.168.56.101:5060 sips:192.168.56.101:5061
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#note: the ip addresses are explicitely specified here because the machine has several interfaces. In a simple case, using '*' instead of the explicit ip address is sufficient,
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#and there is no need to specify the ipv6 transport addresses.
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transports=sip:94.23.19.176:5060 sips:94.23.19.176:5061;tls-certificates-dir=/etc/flexisip/tls/certificates/cn sips:94.23.19.176:5062;tls-certificates-dir=/etc/flexisip/tls/certificates/altname sips:94.23.19.176:5063;tls-verify-incoming=1 sip:94.23.19.176:5064 sip:[2001:41d0:2:14b0::1]:5060 sips:[2001:41d0:2:14b0::1]:5061;tls-certificates-dir=/etc/flexisip/tls/certificates/cn sips:[2001:41d0:2:14b0::1]:5062;tls-certificates-dir=/etc/flexisip/tls/certificates/altname sips:[2001:41d0:2:14b0::1]:5063;tls-verify-incoming=1 sip:[2001:41d0:2:14b0::1]:5064
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# An absolute path of a directory where TLS server certificate and
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# private key can be found, concatenated inside an 'agent.pem' file.
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# Default value: /etc/flexisip/tls
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tls-certificates-dir=/etc/flexisip/tls/certificates/cn
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#tls-certificates-dir=/media/sf_workspaces/workspace-macosx/flexisip
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##
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## STUN server parameters.
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##
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[stun-server]
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# Enable or disable stun server.
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# Default value: true
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enabled=true
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# Local ip address where to bind the socket.
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# Default value: 0.0.0.0
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bind-address=0.0.0.0
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# STUN server port number.
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# Default value: 3478
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port=3478
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##
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## The NatHelper module executes small tasks to make SIP work smoothly
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## despite firewalls.It corrects the Contact headers that contain
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## obviously inconsistent addresses, and adds a Record-Route to ensure
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## subsequent requests are routed also by the proxy, through the
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## UDP or TCP channel each client opened to the proxy.
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##
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[module::NatHelper]
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# Indicate whether the module is activated.
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# Default value: true
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enabled=true
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# A request/response enters module if the boolean filter evaluates
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# to true. Ex: from.uri.domain contains 'sip.linphone.org', from.uri.domain
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# in 'a.org b.org c.org', (to.uri.domain in 'a.org b.org c.org')
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# && (user-agent == 'Linphone v2')
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# Default value:
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filter=
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# Internal URI parameter added to response contact by first proxy
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# and cleaned by last one.
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# Default value: verified
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contact-verified-param=verified
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##
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## The authentication module challenges SIP requests according to
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## a user/password database.
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##
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[module::Authentication]
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# Indicate whether the module is activated.
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# Default value: false
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enabled=true
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no-403=user-agent contains 'tester-no-403'
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# A request/response enters module if the boolean filter evaluates
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# to true. Ex: from.uri.domain contains 'sip.linphone.org', from.uri.domain
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# in 'a.org b.org c.org', (to.uri.domain in 'a.org b.org c.org')
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# && (user-agent == 'Linphone v2')
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# Default value:
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filter= from.uri.domain contains 'sip.example.org' || from.uri.domain contains 'auth.example.org' || from.uri.domain contains 'auth1.example.org' || from.uri.domain contains 'auth2.example.org' || from.uri.domain contains 'anonymous.invalid'
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# List of whitespace separated domain names to challenge. Others
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# are denied.
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# Default value:
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auth-domains= sip.example.org auth.example.org auth1.example.org auth2.example.org
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# List of whitespace separated IP which will not be challenged.
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# Default value:
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trusted-hosts=127.0.0.1 94.23.19.176
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# Database backend implementation [odbc, file].
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# Default value: odbc
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db-implementation=file
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# Odbc connection string to use for connecting to database. ex1:
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# DSN=myodbc3; where 'myodbc3' is the datasource name. ex2: DRIVER={MySQL};SERVER=host;DATABASE=db;USER=user;PASSWORD=pass;OPTION=3;
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# for a DSN-less connection. ex3: /etc/flexisip/passwd; for a file
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# containing one 'user@domain password' by line.
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# Default value:
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datasource=/etc/flexisip/userdb.conf
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# Odbc SQL request to execute to obtain the password
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# . Named parameters are :id (the user found in the from header),
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# :domain (the authorization realm) and :authid (the authorization
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# username). The use of the :id parameter is mandatory.
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# Default value: select password from accounts where id = :id and domain = :domain and authid=:authid
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request=select password from accounts where id = :id and domain = :domain and authid=:authid
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# Use pooling in odbc
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# Default value: true
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odbc-pooling=true
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# Duration of the validity of the credentials added to the cache
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# in seconds.
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# Default value: 1800
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cache-expire=1800
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# True if retrieved passwords from the database are hashed. HA1=MD5(A1)
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# = MD5(username:realm:pass).
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# Default value: false
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hashed-passwords=false
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# When receiving a proxy authenticate challenge, generate a new
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# challenge for this proxy.
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# Default value: false
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new-auth-on-407=false
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enable-test-accounts-creation=true
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##
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## ...
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##
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[module::GatewayAdapter]
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# Indicate whether the module is activated.
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# Default value: false
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enabled=false
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# A request/response enters module if the boolean filter evaluates
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# to true. Ex: from.uri.domain contains 'sip.linphone.org', from.uri.domain
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# in 'a.org b.org c.org', (to.uri.domain in 'a.org b.org c.org')
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# && (user-agent == 'Linphone v2')
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# Default value:
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filter=
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# A gateway uri where to send all requests, as a SIP url (eg 'sip:gateway.example.net')
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# Default value:
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gateway=
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# Modify the from and to domains of incoming register
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# Default value:
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gateway-domain=
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# The gateway will be added to the incoming register contacts.
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# Default value: true
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fork-to-gateway=true
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# Send a REGISTER to the gateway using this server as a contact
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# in order to be notified on incoming calls by the gateway.
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# Default value: true
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register-on-gateway=true
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# Parameter name hosting the incoming domain that will be sent in
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# the register to the gateway.
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# Default value: routing-domain
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routing-param=routing-domain
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[module::Router]
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# Store and retrieve contacts without using the domain.
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# Default value: false
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use-global-domain=false
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# Fork messages to all registered devices
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# Default value: true
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fork=true
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# Force forking and thus the creation of an outgoing transaction
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# even when only one contact found
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# Default value: true
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stateful=true
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# Fork invites to late registers
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# Default value: false
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fork-late=true
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call-fork-timeout=20
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# All the forked have to decline in order to decline the caller
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# invite
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# Default value: false
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fork-no-global-decline=false
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# Maximum duration for delivering a message (text)
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# Default value: 3600
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message-delivery-timeout=60
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##
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## The Registrar module accepts REGISTERs for domains it manages,
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## and store the address of record in order to route other requests
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## destinated to the client who registered.
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##
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[module::Registrar]
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# Indicate whether the module is activated.
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# Default value: true
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enabled=true
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# A request/response enters module if the boolean filter evaluates
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# to true. Ex: from.uri.domain contains 'sip.linphone.org', from.uri.domain
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# in 'a.org b.org c.org', (to.uri.domain in 'a.org b.org c.org')
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# && (user-agent == 'Linphone v2')
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# Default value:
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filter=
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# List of whitelist separated domain names to be managed by the
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# registrar.
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# Default value: localhost
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reg-domains=localhost sip.example.org sipopen.example.org auth1.example.org sip2.linphone.org client.example.org
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# Maximum number of registered contacts of an address of record.
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# Default value: 15
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max-contacts-by-aor=15
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# List of contact uri parameters that can be used to identify a
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# user's device.
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# Default value: +sip.instance
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#unique-id-parameters=
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# Maximum expire time for a REGISTER, in seconds.
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# Default value: 86400
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max-expires=60
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# Minimum expire time for a REGISTER, in seconds.
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# Default value: 60
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min-expires=1
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# File containing the static records to add to database at startup.
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# Format: one 'sip_uri contact_header' by line. Example:
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# <sip:contact@domain> <sip:127.0.0.1:5460>,<sip:192.168.0.1:5160>
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# Default value:
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static-records-file=
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# Timeout in seconds after which the static records file is re-read
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# and the contacts updated.
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# Default value: 600
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static-records-timeout=600
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# Implementation used for storing address of records contact uris.
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# [redis-async, redis-sync, internal]
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# Default value: internal
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db-implementation=internal
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# Generate a contact from the TO header and route it to the above
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# destination. [sip:host:port]
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# Default value:
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generated-contact-route=
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# Require presence of authorization header for specified realm.
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# [Realm]
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# Default value:
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generated-contact-expected-realm=
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[module::ContactRouteInserter]
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# Indicate whether the module is activated.
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# Default value: true
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enabled=false
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# A request/response enters module if the boolean filter evaluates
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# to true. Ex: from.uri.domain contains 'sip.linphone.org', from.uri.domain
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# in 'a.org b.org c.org', (to.uri.domain in 'a.org b.org c.org')
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# && (user-agent == 'Linphone v2')
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# Default value:
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filter=
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# Hack for workarounding Nortel CS2k gateways bug.
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# Default value: false
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masquerade-contacts-for-invites=false
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##
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## This module performs load balancing between a set of configured
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## destination proxies.
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##
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[module::LoadBalancer]
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# Indicate whether the module is activated.
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# Default value: false
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enabled=false
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# A request/response enters module if the boolean filter evaluates
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# to true. Ex: from.uri.domain contains 'sip.linphone.org', from.uri.domain
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# in 'a.org b.org c.org', (to.uri.domain in 'a.org b.org c.org')
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# && (user-agent == 'Linphone v2')
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# Default value:
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filter=
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# Whitespace separated list of sip routes to balance the requests.
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# Example: <sip:192.168.0.22> <sip:192.168.0.23>
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# Default value:
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routes=
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##
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## The MediaRelay module masquerades SDP message so that all RTP
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## and RTCP streams go through the proxy. The RTP and RTCP streams
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## are then routed so that each client receives the stream of the
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## other. MediaRelay makes sure that RTP is ALWAYS established, even
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## with uncooperative firewalls.
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##
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[module::MediaRelay]
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# Indicate whether the module is activated.
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# Default value: true
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enabled=true
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# A request/response enters module if the boolean filter evaluates
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# to true. Ex: from.uri.domain contains 'sip.linphone.org', from.uri.domain
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# in 'a.org b.org c.org', (to.uri.domain in 'a.org b.org c.org')
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# && (:q
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# Default value:
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filter= (user-agent contains 'Natted Linphone')
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# SDP attribute set by the first proxy to forbid subsequent proxies
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# to provide relay.
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# Default value: nortpproxy
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nortpproxy=nortpproxy
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# Set the RTP direction during early media state (duplex, forward)
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# Default value: duplex
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#early-media-rtp-dir=duplex
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# The minimal value of SDP port range
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# Default value: 1024
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sdp-port-range-min=1024
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# The maximal value of SDP port range
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# Default value: 65535
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sdp-port-range-max=65535
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# Enable I-frame only filtering for video H264 for clients annoucing
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# a total bandwith below this value expressed in kbit/s. Use 0 to
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# disable the feature
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# Default value: 0
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#h264-filtering-bandwidth=0
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# When above option is activated, keep one I frame over this number.
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# Default value: 1
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#h264-iframe-decim=1
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# Sends a ACK and BYE to 200 Ok for INVITEs not belonging to any established call.
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bye-orphan-dialogs=true
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##
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## The purpose of the Transcoder module is to transparently transcode
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## from one audio codec to another to make the communication possible
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## between clients that do not share the same set of supported codecs.
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## Concretely it adds all missing codecs into the INVITEs it receives,
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## and adds codecs matching the original INVITE into the 200Ok. Rtp
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## ports and addresses are masqueraded so that the streams can be
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## processed by the proxy. The transcoding job is done in the background
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## by the mediastreamer2 library, as consequence the set of supported
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## codecs is exactly the the same as the codec set supported by mediastreamer2,
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## including the possible plugins you may installed to extend mediastreamer2.
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## WARNING: this module can conflict with the MediaRelay module as
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## both are changin the SDP. Make sure to configure them with different
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## to-domains or from-domains filter if you want to enable both of
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## them.
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##
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[module::Transcoder]
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# Indicate whether the module is activated.
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# Default value: false
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enabled=false
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# A request/response enters module if the boolean filter evaluates
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# to true. Ex: from.uri.domain contains 'sip.linphone.org', from.uri.domain
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# in 'a.org b.org c.org', (to.uri.domain in 'a.org b.org c.org')
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# && (user-agent == 'Linphone v2')
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# Default value:
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filter=
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# Nominal size of RTP jitter buffer, in milliseconds. A value of
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# 0 means no jitter buffer (packet processing).
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# Default value: 0
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jb-nom-size=0
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# Whitespace separated list of user-agent strings for which audio
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# rate control is performed.
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# Default value:
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rc-user-agents=
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# Whitespace seprated list of audio codecs, in order of preference.
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# Default value: speex/8000 amr/8000 iLBC/8000 gsm/8000 pcmu/8000 pcma/8000
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audio-codecs=speex/8000 amr/8000 iLBC/8000 gsm/8000 pcmu/8000 pcma/8000
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# If true, retransmissions of INVITEs will be blocked. The purpose
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# of this option is to limit bandwidth usage and server load on
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# reliable networks.
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# Default value: false
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block-retransmissions=false
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##
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## This module executes the basic routing task of SIP requests and
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## pass them to the transport layer. It must always be enabled.
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##
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[module::Forward]
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# Indicate whether the module is activated.
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# Default value: true
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enabled=true
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# A request/response enters module if the boolean filter evaluates
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# to true. Ex: from.uri.domain contains 'sip.linphone.org', from.uri.domain
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# in 'a.org b.org c.org', (to.uri.domain in 'a.org b.org c.org')
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# && (user-agent == 'Linphone v2')
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# Default value:
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filter=
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# A sip uri where to send all requests
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# Default value:
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route=
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# Rewrite request-uri's host and port according to above route
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# Default value: false
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rewrite-req-uri=false
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[module::Redirect]
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enabled=true
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filter = (user-agent contains 'redirect') && !(request.uri.params contains 'redirected')
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contact= <sip:sipopen.example.org;redirected>
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##
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## The purpose of the StatisticsCollector module is to collect call
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## statistics (RFC 6035) and store them on the server.
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##
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[module::StatisticsCollector]
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# Indicate whether the module is activated.
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# Default value: false
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enabled=true
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# A request/response enters module if the boolean filter evaluates
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# to true. Ex: from.uri.domain contains 'sip.linphone.org', from.uri.domain
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# in 'a.org b.org c.org', (to.uri.domain in 'a.org b.org c.org')
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# && (user-agent == 'Linphone v2')
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# Default value:
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filter=
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# SIP URI of the statistics collector. Note that the messages destinated
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# to this address will be deleted by this module and thus not be
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# delivered.
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# Default value:
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collector-address=sip:sip.example.org
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##
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## This module performs push notifications to mobile phone notification
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## systems: apple, android, windows, as well as a generic http get/post
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## to a custom server to which actual sending of the notification
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## is delegated. The push notification is sent when an INVITE or
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## MESSAGE request is not answered by the destination of the request
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## within a certain period of time, configurable hereunder as 'timeout'
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## parameter.
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##
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[module::PushNotification]
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# Indicate whether the module is activated.
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# Default value: false
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enabled=true
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# A request/response enters module if the boolean filter evaluates
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# to true. Ex: from.uri.domain contains 'sip.linphone.org', from.uri.domain
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# in 'a.org b.org c.org', (to.uri.domain in 'a.org b.org c.org')
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# && (user-agent == 'Linphone v2')
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# Default value:
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filter=
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# Number of second to wait before sending a push notification to
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# device(if <=0 then disabled)
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# Default value: 5
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timeout=5
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# Maximum number of notifications queued for each client
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# Default value: 10
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max-queue-size=10
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# Enable push notification for apple devices
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# Default value: true
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apple=false
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# Path to directory where to find Apple Push Notification service
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# certificates. They should bear the appid of the application, suffixed
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# by the release mode and .pem extension. For example: org.linphone.dev.pem
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# org.linphone.prod.pem com.somephone.dev.pem etc... The files should
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# be .pem format, and made of certificate followed by private key.
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# Default value: /etc/flexisip/apn
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apple-certificate-dir=/etc/flexisip/apn
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# Enable push notification for android devices
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# Default value: true
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google=false
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# List of couples projectId:ApiKey for each android project that
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# supports push notifications
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# Default value:
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google-projects-api-keys=
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# Enable push notification for windows phone 8 devices
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# Default value: true
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windowsphone=false
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# Set the badge value to 0 for apple push
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# Default value: false
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no-badge=false
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# Instead of having Flexisip sending the push notification directly
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# to the Google/Apple/Microsoft push servers, send an http request
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# to an http server with all required information encoded in URL,
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# to which the actual sending of the push notification is delegated.
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# The following arguments can be substitued in the http request
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# uri, with the following values:
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# - $type : apple, google, wp
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# - $event : call, message
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# - $from-name : the display name in the from header
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# - $from-uri : the sip uri of the from header
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# - $from-tag : the tag of the from header
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# - $call-id : the call-id of the INVITE or MESSAGE request
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# - $to-uri : the sip uri of the to header
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# - $api-key : the api key to use (google only)
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# - $msgid : the message id to put in the notification
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# - $sound : the sound file to play with the notification
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#
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The content of the text message is put in the body of the http
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# request as text/plain, if any.
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# Example: http://192.168.0.2/$type/$event?from-uri=$from-uri&tag=$from-tag&callid=$callid&to=$to-uri
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# Default value:
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external-push-uri=http://127.0.0.1:80/$type/$event?from-uri=$from-uri&tag=$from-tag&callid=$callid&to=$to-uri
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# Method for reaching external-push-uri, typically GET or POST
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# Default value: GET
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external-push-method=GET
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##
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## This module bans user when they are sending too much packets on
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## a given timelapseTo see the list of currently banned ips/ports,
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## use iptables -LYou can also check the queue of unban commands
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## using atq
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##
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[module::DoSProtection]
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# Indicate whether the module is activated.
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# Default value: true
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enabled=true
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# A request/response enters module if the boolean filter evaluates
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# to true. Ex: from.uri.domain contains 'sip.linphone.org', from.uri.domain
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# in 'a.org b.org c.org', (to.uri.domain in 'a.org b.org c.org')
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# && (user-agent == 'Linphone v2')
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# Default value:
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filter=
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# Number of milliseconds to consider to compute the packet rate
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# Default value: 3000
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time-period=15000
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# Maximum packet rate received in [time-period] millisecond(s) to
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# consider it as a DoS attack.
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# Default value: 20
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packet-rate-limit=10
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# Number of minutes to ban the ip/port using iptables (might be
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# less because it justs uses the minutes of the clock, not the seconds.
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# So if the unban command is queued at 13:11:56 and scheduled and
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# the ban time is 1 minute, it will be executed at 13:12:00)
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# Default value: 2
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ban-time=1
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[module::Presence]
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enabled=true
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presence-server = <sip:127.0.0.1:5065;transport=tcp>
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only-list-subscription = !(user-agent contains 'full-presence-support')
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[presence-server]
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expires = 600
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transports = sip:127.0.0.1:5065;transport=tcp
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