########################################################
# reConServer configuration file
########################################################

########################################################
# Logging settings
########################################################

# Logging Type: syslog|cerr|cout|file
LoggingType = file

# Logging level: NONE|CRIT|ERR|WARNING|INFO|DEBUG|STACK
LoggingLevel = WARNING

# Log Filename
LogFilename = /var/log/reConServer/reConServer.log

# Log file Max Size
LogFileMaxLines = 0


########################################################
# Transport settings
########################################################


# Local IP Address to bind SIP transports to.
# In general the IP address to bind to is queried from the host OS.This switch allows specification
# of the IP address for OS's that cannot be queried, or for machines that have mulitple NICs.
#IPAddress = 192.168.1.106
#IPAddress = 2001:5c0:1000:a::6d
IPAddress =

# Local port number to use for SIP messaging over TCP - 0 to disable
TCPPort = 5062

# Local port number to use for SIP messaging over UDP - 0 to disable
UDPPort = 5062

# Local port number to use for TLS SIP messaging - 0 to disable
# By default, reConServer will listen for TLS connections on port 5063, use this switch to specify
# something different.  Note SIP certificiates must be present in executable directory for windows
# hosts and ~/.sipCerts directory on linux hosts.
TLSPort = 0

# Local port number to start allocating from for RTP media
MediaPortStart = 17384


# URI of a proxy server to use a SIP outbound proxy.
# By default reConServer does not use an outbound proxy.  Use this switch to route all
# outbound, out-of-dialog requests through a fixed proxy despite the destination URI.
OutboundProxyUri =

# By default, no secure media is offered in outbound SIP requests.  Use this option to
# change that behaviour.  Note:  Inbound secure media is always accepted.
# Srtp     - use SRTP with keying outside of media stream (SDES key negotiation)
#            via SDP.  RTP/AVP profile is used, and transport capability of RTP/SAVP is
#            listed, in order to implement best-effort SRTP.  Note:  The crypo attribute
#            is provided outside of the SDP capability, as this is required by SNOM for
#            optional SRTP offers.
# SrtpReq  - use SRTP with keying outside of media stream (SDES key negotiation)
#            via SDP.  RTP/SAVP profile is used to indicate that SRTP is mandatory.
# SrtpDtls - use SRTP with DTLS key negotiation.  RTP/AVP is use as a default, and a
#            transport capability of UDP/TLS/RTP/SAVP is listed, in order to impelement
#            best-effort DTLS-SRTP.
# SrtpDtlsReq - use SRTP with DTLS key negotiation.  UDP/TLS/RTP/SAVP profile is used to
#            indicate that Dtls-Srtp use is mandatory.
#SecureMediaMode =

# By default, no NAT traversal strategies are used.  Use this switch to specify one:
# None - do not use any NAT traversal strategy (this is the default)
# Bind - use Binding discovery on a STUN server, to discover and use "public" address
#        and port in SDP negotiations
# UdpAlloc - Use a TURN server as a media relay.  Communicate to the TURN
#            server over UDP and Allocate a UDP relay address and port to
#            use in SDP negotiations
# TcpAlloc - Use a TURN server as a media relay.  Communicate to the TURN
#            server over TCP and Allocate a UDP relay address and port to
#            use in SDP negotiations
# TlsAlloc - Use a TURN server as a media relay.  Communicate to the TURN
#            server over TLS and Allocate a UDP relay address and port to
#NatTraversalMode =

# Hostname of the NAT STUN/TURN server. If NatTraversalMode is used then you MUST specify the
# STUN/TURN server name/address.
NatTraversalServerHostname =

# Port of the NAT STUN/TURN server. If NatTraversalMode is used then you MUST specify the STUN/TURN
# server port.
# Default for UDP and TCP is 3478 and for TURN over TLS it is 5349
NatTraversalServerPort = 3478

########################################################
# Authentication settings
########################################################

# STUN/TURN username to use for NAT server. Use this option if the STUN/TURN server requires
# authentication.
StunUsername =

# STUN/TURN password to use for NAT server. Use this option if the STUN/TURN server requires
# authentication.
StunPassword =

########################################################
# UNIX related settings
########################################################

# Must be true or false, default = false, not supported on Windows
Daemonize = false

# On UNIX it is normal to create a PID file
# if unspecified, no attempt will be made to create a PID file
#PidFile = /var/run/reConServer/reConServer.pid

# UNIX account information to run process as
#RunAsUser = recon
#RunAsGroup = recon

########################################################
# Misc settings
########################################################

# This option is used to specify the SIP URI for this instance of reConServer.  reConServer uses
# this setting (-ip is not specified) in order to find the regisration server.  If
# nothing is specified, then the default of sip:noreg@<ipaddress> will be used.
SIPUri =

# SIP password of this SIP user
# Use this switch in cases where the proxy digest challenges sip messaging.
Password =

# Comma separated list of DNS servers, overrides default OS detected list (leave blank
# for default)
DNSServers =

# Domain name to use for TLS server connections.
# By default, reConServer will query the OS for a local hostname for TLS, use this switch
# to override the OS queried result.
TLSDomain =

# Set this to disable registration with SIP Proxy.
# By default, if a SIP uri is specified, reConServer will attempt to register with it
DisableRegistration = false

# Enabling autoanswer will cause reConServer to automatically answer any inbound SIP calls
# and place them in the lowest numbered conversation currently created.
EnableAutoAnswer = true

# Set this to disable sending of keepalives.
# By default, reConServer will enable UDP CRLF keepalives every 30 seconds and TCP keepalives
# every 180 seconds.  Use this switch to disable CRLF keepalives.
DisableKeepAlives = false

# Local audio support - enables requirement for local audio hardware.
# Note:  if local audio support is disabled, then local participants cannot be created.
EnableLocalAudio = false

# Keyboard control from stdin
# Only permitted when run in the foreground (not as a daemon process)
KeyboardInput = true

# Whether to use a global media interface (bridge) for all conferences/conversations or
# a distinct media interface instance for each conference/conversation
# The global approach allows participants to be engaged in more than one concurrent
# conversation but it also limits the total number of participants in the system
# at one time.
# For servers hosting multiple conferences, the global media interface is usually
# not desirable so it is disabled by default
GlobalMediaInterface = false

# The default and maximum sample rate to use when creating flow graphs
# in sipXtapi.
# For cases where narrowband audio is used (G.711a or G.711u),
# the default value, 8000hz, is appropriate.
# For users wanting HD audio (for example, with the G.722 or Opus codecs),
# it is highly desirable to set these to the value associated with
# the codec, for example:
#
#   G.722:    16000
#   Opus:     48000
#
# If the internal sample rate (specified here) is not consistent with
# the sample rate used by the participants the sipXtapi media stack
# will convert the sample rate of the signals on the fly but this has an
# impact on CPU load.
# For sipXtapi resampling to work, sipXtapi must be linked against libspeexdsp
# (even if not using speex).
DefaultSampleRate = 8000
MaximumSampleRate = 8000

# Enable the G.722 codec
# G.722 is a high-definition (HD) voice codec supported by many
# desk phones and softphones.  It uses the same amount of bandwidth
# as G.711 while increasing the audio spectrum from 3khz to about 7khz.
# To use it, it is usually desirable to set the sample rate to 16000
# instead of the default 8000
# It is disabled by default.
#EnableG722 = true

# Enable the Opus codec
# Opus is a variable bit-rate voice codec supported by many
# modern softphones and WebRTC browsers.
# To use it, it is highly desirable to set the sample rate to 48000
# instead of the default 8000
# It is disabled by default.
#EnableOpus = true

# Specify the application to run
# The default value is "None" - in this mode, the creation and manipulation
# of conversations/conferences is done manually using the console
# or if auto answer is enabled, all callers simply end up in a single bridge.
# One additional application is provided:
# B2BUA:   with this application, every incoming SIP call automatically
#          creates a new conversation and a new outgoing call (B-leg).
#          The incoming call will receive answer and alerting signals
#          as they are received from the B-leg.  DTMF tones
#          are relayed from either leg to the other.
#Application = B2BUA

# Specify where the B2BUA should route calls
# If B2BUA mode is enabled, the B2BUA will take the user-part of the
# request URI and append the hostname specified here to construct
# a the destination URI for the B-leg of the call.
#B2BUANextHop = sip-host.example.org?transport=tcp

# Specify headers that will be copied to the outgoing INVITE
# A comma-separated list of extension headers that will be copied
# from the incoming INVITE to the outgoing INVITE
#B2BUAReplicateHeaders = X-Customer-ID, X-Ticket-Number