linphone-iphone/tester/flexisip/flexisip.conf

##
## This is the default Flexisip configuration file
##

##
## Some global settings of the flexisip proxy.
##
[global]
# Outputs very detailed logs
#  Default value: false
debug=1

# Automatically respawn flexisip in case of abnormal termination
# (crashes)
#  Default value: true
auto-respawn=true

# List of white space separated host names pointing to this machine.
# This is to prevent loops while routing SIP messages.
#  Default value: localhost
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

# List of white space separated SIP uris where the proxy must listen.Wildcard
# (*) can be used to mean 'all local ip addresses'. If 'transport'
# prameter is unspecified, it will listen to both udp and tcp. An
# local address to bind can be indicated in the 'maddr' parameter,
# while the domain part of the uris are used as public domain or
# ip address. Here some examples to understand:
# * listen on all local interfaces for udp and tcp, on standart
# port:
# 	transports=sip:*
# * listen on all local interfaces for udp,tcp and tls, on standart
# ports:
# 	transports=sip:* sips:*
# * listen on 192.168.0.29:6060 with tls, but public hostname is
# 'sip.linphone.org' used in SIP messages. Bind address won't appear:
# 	transports=sips:sip.linphone.org:6060;maddr=192.168.0.29
#  Default value: sip:*
#transports=sip:192.168.56.101:5060  sips:192.168.56.101:5061

#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, 
#and there is no need to specify the ipv6 transport addresses.
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


# An absolute path of a directory where TLS server certificate and
# private key can be found, concatenated inside an 'agent.pem' file.
#  Default value: /etc/flexisip/tls
tls-certificates-dir=/etc/flexisip/tls/certificates/cn
#tls-certificates-dir=/media/sf_workspaces/workspace-macosx/flexisip

##
## STUN server parameters.
##
[stun-server]
# Enable or disable stun server.
#  Default value: true
enabled=true

# Local ip address where to bind the socket.
#  Default value: 0.0.0.0
bind-address=0.0.0.0

# STUN server port number.
#  Default value: 3478
port=3478



##
## The NatHelper module executes small tasks to make SIP work smoothly
## despite firewalls.It corrects the Contact headers that contain
## obviously inconsistent addresses, and adds a Record-Route to ensure
## subsequent requests are routed also by the proxy, through the
## UDP or TCP channel each client opened to the proxy.
##
[module::NatHelper]
# Indicate whether the module is activated.
#  Default value: true
enabled=true

# A request/response enters module if the boolean filter evaluates
# to true. Ex: from.uri.domain contains 'sip.linphone.org', from.uri.domain
# in 'a.org b.org c.org', (to.uri.domain in 'a.org b.org c.org')
# && (user-agent == 'Linphone v2')
#  Default value:
filter=

# Internal URI parameter added to response contact by first proxy
# and cleaned by last one.
#  Default value: verified
contact-verified-param=verified

##
## The authentication module challenges SIP requests according to
## a user/password database.
##
[module::Authentication]
# Indicate whether the module is activated.
#  Default value: false
enabled=true


no-403=user-agent contains 'tester-no-403'

# A request/response enters module if the boolean filter evaluates
# to true. Ex: from.uri.domain contains 'sip.linphone.org', from.uri.domain
# in 'a.org b.org c.org', (to.uri.domain in 'a.org b.org c.org')
# && (user-agent == 'Linphone v2')
#  Default value: 
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' 

# List of whitespace separated domain names to challenge. Others
# are denied.
#  Default value: 
auth-domains= sip.example.org auth.example.org auth1.example.org auth2.example.org



# List of whitespace separated IP which will not be challenged.
#  Default value: 
trusted-hosts=127.0.0.1 94.23.19.176

# Database backend implementation [odbc, file].
#  Default value: odbc
db-implementation=file

# Odbc connection string to use for connecting to database. ex1:
# DSN=myodbc3; where 'myodbc3' is the datasource name. ex2: DRIVER={MySQL};SERVER=host;DATABASE=db;USER=user;PASSWORD=pass;OPTION=3;
# for a DSN-less connection. ex3: /etc/flexisip/passwd; for a file
# containing one 'user@domain password' by line.
#  Default value:
datasource=/etc/flexisip/userdb.conf

# Odbc SQL request to execute to obtain the password
# . Named parameters are :id (the user found in the from header),
# :domain (the authorization realm) and :authid (the authorization
# username). The use of the :id parameter is mandatory.
#  Default value: select password from accounts where id = :id and domain = :domain and authid=:authid
request=select password from accounts where id = :id and domain = :domain and authid=:authid


# Use pooling in odbc
#  Default value: true
odbc-pooling=true


# Duration of the validity of the credentials added to the cache
# in seconds.
#  Default value: 1800
cache-expire=1800


# True if retrieved passwords from the database are hashed. HA1=MD5(A1)
# = MD5(username:realm:pass).
#  Default value: false
hashed-passwords=false

# When receiving a proxy authenticate challenge, generate a new
# challenge for this proxy.
#  Default value: false
new-auth-on-407=false

enable-test-accounts-creation=true

##
## ...
##
[module::GatewayAdapter]
# Indicate whether the module is activated.
#  Default value: false
enabled=false

# A request/response enters module if the boolean filter evaluates
# to true. Ex: from.uri.domain contains 'sip.linphone.org', from.uri.domain
# in 'a.org b.org c.org', (to.uri.domain in 'a.org b.org c.org')
# && (user-agent == 'Linphone v2')
#  Default value:
filter=

# A gateway uri where to send all requests, as a SIP url (eg 'sip:gateway.example.net')
#  Default value:
gateway=

# Modify the from and to domains of incoming register
#  Default value:
gateway-domain=

# The gateway will be added to the incoming register contacts.
#  Default value: true
fork-to-gateway=true

# Send a REGISTER to the gateway using this server as a contact
# in order to be notified on incoming calls by the gateway.
#  Default value: true
register-on-gateway=true

# Parameter name hosting the incoming domain that will be sent in
# the register to the gateway.
#  Default value: routing-domain
routing-param=routing-domain

[module::Router]

# Store and retrieve contacts without using the domain.
#  Default value: false
use-global-domain=false

# Fork messages to all registered devices
#  Default value: true
fork=true

# Force forking and thus the creation of an outgoing transaction
# even when only one contact found
#  Default value: true
stateful=true

# Fork invites to late registers
#  Default value: false
fork-late=true

call-fork-timeout=20



# All the forked have to decline in order to decline the caller
# invite
#  Default value: false
fork-no-global-decline=false

# Maximum duration for delivering a message (text)
#  Default value: 3600
message-delivery-timeout=60
##
## The Registrar module accepts REGISTERs for domains it manages,
## and store the address of record in order to route other requests
## destinated to the client who registered.
##
[module::Registrar]
# Indicate whether the module is activated.
#  Default value: true
enabled=true

# A request/response enters module if the boolean filter evaluates
# to true. Ex: from.uri.domain contains 'sip.linphone.org', from.uri.domain
# in 'a.org b.org c.org', (to.uri.domain in 'a.org b.org c.org')
# && (user-agent == 'Linphone v2')
#  Default value:
filter=

# List of whitelist separated domain names to be managed by the
# registrar.
#  Default value: localhost
reg-domains=localhost sip.example.org sipopen.example.org auth1.example.org sip2.linphone.org client.example.org

# Maximum number of registered contacts of an address of record.
#  Default value: 15
max-contacts-by-aor=15

# List of contact uri parameters that can be used to identify a
# user's device.
#  Default value: +sip.instance
#unique-id-parameters=

# Maximum expire time for a REGISTER, in seconds.
#  Default value: 86400
max-expires=60

# Minimum expire time for a REGISTER, in seconds.
#  Default value: 60
min-expires=1

# File containing the static records to add to database at startup.
# Format: one 'sip_uri contact_header' by line. Example:
# <sip:contact@domain> <sip:127.0.0.1:5460>,<sip:192.168.0.1:5160>
#  Default value:
static-records-file=

# Timeout in seconds after which the static records file is re-read
# and the contacts updated.
#  Default value: 600
static-records-timeout=600

# Implementation used for storing address of records contact uris.
# [redis-async, redis-sync, internal]
#  Default value: internal
db-implementation=internal








# Generate a contact from the TO header and route it to the above
# destination. [sip:host:port]
#  Default value:
generated-contact-route=

# Require presence of authorization header for specified realm.
# [Realm]
#  Default value:
generated-contact-expected-realm=


[module::ContactRouteInserter]
# Indicate whether the module is activated.
#  Default value: true
enabled=false

# A request/response enters module if the boolean filter evaluates
# to true. Ex: from.uri.domain contains 'sip.linphone.org', from.uri.domain
# in 'a.org b.org c.org', (to.uri.domain in 'a.org b.org c.org')
# && (user-agent == 'Linphone v2')
#  Default value:
filter=

# Hack for workarounding Nortel CS2k gateways bug.
#  Default value: false
masquerade-contacts-for-invites=false

##
## This module performs load balancing between a set of configured
## destination proxies.
##
[module::LoadBalancer]
# Indicate whether the module is activated.
#  Default value: false
enabled=false

# A request/response enters module if the boolean filter evaluates
# to true. Ex: from.uri.domain contains 'sip.linphone.org', from.uri.domain
# in 'a.org b.org c.org', (to.uri.domain in 'a.org b.org c.org')
# && (user-agent == 'Linphone v2')
#  Default value:
filter=

# Whitespace separated list of sip routes to balance the requests.
# Example: <sip:192.168.0.22> <sip:192.168.0.23>
#  Default value:
routes=

##
## The MediaRelay module masquerades SDP message so that all RTP
## and RTCP streams go through the proxy. The RTP and RTCP streams
## are then routed so that each client receives the stream of the
## other. MediaRelay makes sure that RTP is ALWAYS established, even
## with uncooperative firewalls.
##
[module::MediaRelay]
# Indicate whether the module is activated.
#  Default value: true
enabled=true

# A request/response enters module if the boolean filter evaluates
# to true. Ex: from.uri.domain contains 'sip.linphone.org', from.uri.domain
# in 'a.org b.org c.org', (to.uri.domain in 'a.org b.org c.org')
# && (:q
#  Default value:
filter= (user-agent contains 'Natted Linphone')

# SDP attribute set by the first proxy to forbid subsequent proxies
# to provide relay.
#  Default value: nortpproxy
nortpproxy=nortpproxy

# Set the RTP direction during early media state (duplex, forward)
#  Default value: duplex
#early-media-rtp-dir=duplex

# The minimal value of SDP port range
#  Default value: 1024
sdp-port-range-min=1024

# The maximal value of SDP port range
#  Default value: 65535
sdp-port-range-max=65535

# Enable I-frame only filtering for video H264 for clients annoucing
# a total bandwith below this value expressed in kbit/s. Use 0 to
# disable the feature
#  Default value: 0
#h264-filtering-bandwidth=0

# When above option is activated, keep one I frame over this number.
#  Default value: 1
#h264-iframe-decim=1

# Sends a ACK and BYE to 200 Ok for INVITEs not belonging to any established call.
bye-orphan-dialogs=true


##
## The purpose of the Transcoder module is to transparently transcode
## from one audio codec to another to make the communication possible
## between clients that do not share the same set of supported codecs.
## Concretely it adds all missing codecs into the INVITEs it receives,
## and adds codecs matching the original INVITE into the 200Ok. Rtp
## ports and addresses are masqueraded so that the streams can be
## processed by the proxy. The transcoding job is done in the background
## by the mediastreamer2 library, as consequence the set of supported
## codecs is exactly the the same as the codec set supported by mediastreamer2,
## including the possible plugins you may installed to extend mediastreamer2.
## WARNING: this module can conflict with the MediaRelay module as
## both are changin the SDP. Make sure to configure them with different
## to-domains or from-domains filter if you want to enable both of
## them.
##
[module::Transcoder]
# Indicate whether the module is activated.
#  Default value: false
enabled=false

# A request/response enters module if the boolean filter evaluates
# to true. Ex: from.uri.domain contains 'sip.linphone.org', from.uri.domain
# in 'a.org b.org c.org', (to.uri.domain in 'a.org b.org c.org')
# && (user-agent == 'Linphone v2')
#  Default value:
filter=

# Nominal size of RTP jitter buffer, in milliseconds. A value of
# 0 means no jitter buffer (packet processing).
#  Default value: 0
jb-nom-size=0

# Whitespace separated list of user-agent strings for which audio
# rate control is performed.
#  Default value:
rc-user-agents=

# Whitespace seprated list of audio codecs, in order of preference.
#  Default value: speex/8000 amr/8000 iLBC/8000 gsm/8000 pcmu/8000 pcma/8000
audio-codecs=speex/8000 amr/8000 iLBC/8000 gsm/8000 pcmu/8000 pcma/8000

# If true, retransmissions of INVITEs will be blocked. The purpose
# of this option is to limit bandwidth usage and server load on
# reliable networks.
#  Default value: false
block-retransmissions=false

##
## This module executes the basic routing task of SIP requests and
## pass them to the transport layer. It must always be enabled.
##
[module::Forward]
# Indicate whether the module is activated.
#  Default value: true
enabled=true

# A request/response enters module if the boolean filter evaluates
# to true. Ex: from.uri.domain contains 'sip.linphone.org', from.uri.domain
# in 'a.org b.org c.org', (to.uri.domain in 'a.org b.org c.org')
# && (user-agent == 'Linphone v2')
#  Default value:
filter=

# A sip uri where to send all requests
#  Default value:
route=

# Rewrite request-uri's host and port according to above route
#  Default value: false
rewrite-req-uri=false

[module::Redirect]
enabled=true
filter = (user-agent contains 'redirect') && !(request.uri.params contains 'redirected')
contact= <sip:sipopen.example.org;redirected>

##
## The purpose of the StatisticsCollector module is to collect call
## statistics (RFC 6035) and store them on the server.
##
[module::StatisticsCollector]
# Indicate whether the module is activated.
#  Default value: false
enabled=true

# A request/response enters module if the boolean filter evaluates
# to true. Ex: from.uri.domain contains 'sip.linphone.org', from.uri.domain
# in 'a.org b.org c.org', (to.uri.domain in 'a.org b.org c.org')
# && (user-agent == 'Linphone v2')
#  Default value:
filter=

# SIP URI of the statistics collector. Note that the messages destinated
# to this address will be deleted by this module and thus not be
# delivered.
#  Default value:
collector-address=sip:sip.example.org

##
## This module performs push notifications to mobile phone notification
## systems: apple, android, windows, as well as a generic http get/post
## to a custom server to which actual sending of the notification
## is delegated. The push notification is sent when an INVITE or
## MESSAGE request is not answered by the destination of the request
## within a certain period of time, configurable hereunder as 'timeout'
## parameter.
##



[module::PushNotification]
# Indicate whether the module is activated.
#  Default value: false
enabled=true

# A request/response enters module if the boolean filter evaluates
# to true. Ex: from.uri.domain contains 'sip.linphone.org', from.uri.domain
# in 'a.org b.org c.org', (to.uri.domain in 'a.org b.org c.org')
# && (user-agent == 'Linphone v2')
#  Default value:
filter=

# Number of second to wait before sending a push notification to
# device(if <=0 then disabled)
#  Default value: 5
timeout=5

# Maximum number of notifications queued for each client
#  Default value: 10
max-queue-size=10

# Enable push notification for apple devices
#  Default value: true
apple=false

# Path to directory where to find Apple Push Notification service
# certificates. They should bear the appid of the application, suffixed
# by the release mode and .pem extension. For example: org.linphone.dev.pem
# org.linphone.prod.pem com.somephone.dev.pem etc... The files should
# be .pem format, and made of certificate followed by private key.
#  Default value: /etc/flexisip/apn
apple-certificate-dir=/etc/flexisip/apn

# Enable push notification for android devices
#  Default value: true
google=false

# List of couples projectId:ApiKey for each android project that
# supports push notifications
#  Default value:
google-projects-api-keys=

# Enable push notification for windows phone 8 devices
#  Default value: true
windowsphone=false

# Set the badge value to 0 for apple push
#  Default value: false
no-badge=false

# Instead of having Flexisip sending the push notification directly
# to the Google/Apple/Microsoft push servers, send an http request
# to an http server with all required information encoded in URL,
# to which the actual sending of the push notification is delegated.
# The following arguments can be substitued in the http request
# uri, with the following values:
#  - $type : apple, google, wp
#  - $event : call, message
#  - $from-name : the display name in the from header
#  - $from-uri : the sip uri of the from header
#  - $from-tag : the tag of the from header
#  - $call-id : the call-id of the INVITE or MESSAGE request
#  - $to-uri : the sip uri of the to header
#  - $api-key : the api key to use (google only)
#  - $msgid : the message id to put in the notification
#  - $sound : the sound file to play with the notification
#
 The content of the text message is put in the body of the http
# request as text/plain, if any.
#  Example: http://192.168.0.2/$type/$event?from-uri=$from-uri&tag=$from-tag&callid=$callid&to=$to-uri
#  Default value:
external-push-uri=http://127.0.0.1:80/$type/$event?from-uri=$from-uri&tag=$from-tag&callid=$callid&to=$to-uri

# Method for reaching external-push-uri, typically GET or POST
#  Default value: GET
external-push-method=GET

##
## This module bans user when they are sending too much packets on
## a given timelapseTo see the list of currently banned ips/ports,
## use iptables -LYou can also check the queue of unban commands
## using atq
##
[module::DoSProtection]

# Indicate whether the module is activated.
#  Default value: true
enabled=true

# A request/response enters module if the boolean filter evaluates
# to true. Ex: from.uri.domain contains 'sip.linphone.org', from.uri.domain
# in 'a.org b.org c.org', (to.uri.domain in 'a.org b.org c.org')
# && (user-agent == 'Linphone v2')
#  Default value:
filter=

# Number of milliseconds to consider to compute the packet rate
#  Default value: 3000
time-period=15000

# Maximum packet rate received in [time-period] millisecond(s) to
# consider it as a DoS attack.
#  Default value: 20
packet-rate-limit=10

# Number of minutes to ban the ip/port using iptables (might be
# less because it justs uses the minutes of the clock, not the seconds.
# So if the unban command is queued at 13:11:56 and scheduled and
# the ban time is 1 minute, it will be executed at 13:12:00)
#  Default value: 2
ban-time=1

[module::Presence]
enabled=true
presence-server = <sip:127.0.0.1:5065;transport=tcp>
only-list-subscription = !(user-agent contains 'full-presence-support')

[presence-server]
expires = 600
transports = sip:127.0.0.1:5065;transport=tcp