linphone-ios/tester/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=false

# List of white space separated host names pointing to this machine.
# This is to prevent loops while routing SIP messages.
#  Default value: localhost
aliases=auth.example.org auth1.example.org auth2.example.org sip.example.org

# The public ip address of the proxy.
#  Default value: guess
ip-address=guess

# The local interface's ip address where to listen. The wildcard
# (*) means all interfaces.
#  Default value: *
bind-address=sip.example.org

# UDP/TCP port number to listen for sip messages.
#  Default value: 5060
port=5060


##
## TLS specific parameters.
##
[tls]
# Enable SIP/TLS (sips)
#  Default value: true
enabled=true

# The port used for SIP/TLS
#  Default value: 5061
port=5061

# An absolute path of a directory where TLS certificate can be found.
# The private key for TLS server must be in a agent.pem file within
# this directory
#  Default value: /etc/flexisip/tls
certificates-dir=/Users/jehanmonnier/workspaces/workspace-macosx/flexisip


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

# 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=false

# List of domain names in sip from allowed to enter the module.
#  Default value: *
from-domains=*

# List of domain names in sip to allowed to enter the module.
#  Default value: *
to-domains=*


##
## 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

# List of domain names in sip from allowed to enter the module.
#  Default value: *
from-domains=auth.example.org auth1.example.org auth2.example.org

# List of domain names in sip to allowed to enter the module.
#  Default value: *
to-domains=*

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

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

# 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=localhost;DATABASE=dbname;USER=username;PASSWORD=passname;OPTION=3;
# for a DSN-less connection. ex3: /etc/flexisip/passwd; for a file
# containing one 'user@domain password' by line.
#  Default value: 
datasource=./userdb.conf

# Odbc SQL request to execute to obtain the password. Named parameters
# are :id, :domain and :authid.'
#  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

# Maximum length of the login column in database.
#  Default value: 100
max-id-length=100

# Maximum length of the password column in database
#  Default value: 100
max-password-length=100

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

# Display timing statistics after this count of seconds
#  Default value: 0
odbc-display-timings-interval=0

# Display timing statistics once the number of samples reach this
# number.
#  Default value: 0
odbc-display-timings-after-count=0

# Retrieve passwords asynchronously.
#  Default value: false
odbc-asynchronous=false

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

# Retrieve password immediately so that it is cached when an authenticated
# request arrives.
#  Default value: true
immediate-retrieve-password=true

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



##
## 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

# List of domain names in sip from allowed to enter the module.
#  Default value: *
from-domains=auth.example.org auth1.example.org auth2.example.org sip.example.org

# List of domain names in sip to allowed to enter the module.
#  Default value: *
to-domains=*

# List of whitelist separated domain names to be managed by the
# registrar.
#  Default value: localhost
reg-domains=auth.example.org auth1.example.org auth2.example.org sip.example.org


##
## The purpose of the ContactRouteInserter module is to masquerade
## the contact header of incoming registers that are not handled
## locally (think about flexisip used as a SBC gateway) in such a
## way that it is then possible to route back outgoing invites to
## the original address. It is a kind of similar mechanism as Record-Route,
## but for REGISTER.
##
[module::ContactRouteInserter]
# Indicate whether the module is activated.
#  Default value: true
enabled=false

# List of domain names in sip from allowed to enter the module.
#  Default value: *
from-domains=*

# List of domain names in sip to allowed to enter the module.
#  Default value: *
to-domains=*

# 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

# List of domain names in sip from allowed to enter the module.
#  Default value: *
from-domains=*

# List of domain names in sip to allowed to enter the module.
#  Default value: *
to-domains=*

# 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=false

# List of domain names in sip from allowed to enter the module.
#  Default value: *
from-domains=*

# List of domain names in sip to allowed to enter the module.
#  Default value: *
to-domains=*


##
## 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=true

# List of domain names in sip from allowed to enter the module.
#  Default value: *
from-domains=freephonie.net

# List of domain names in sip to allowed to enter the module.
#  Default value: *
to-domains=freephonie.net

# 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 telephone-event/8000
audio-codecs=amr/8000 pcmu/8000 pcma/8000 telephone-event/8000


##
## 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

# List of domain names in sip from allowed to enter the module.
#  Default value: *
from-domains=*

# List of domain names in sip to allowed to enter the module.
#  Default value: *
to-domains=*

# A sip uri where to send all requests
#  Default value: 
#route=<sip:sip.linphone.org>

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