puppet/modules/rhizo_base/templates/sems.conf.erb

# $Id$
#
# sems.conf.sample
#
# Sip Express Media Server (sems)
# 
# sample configuration file
# 
#
# whitespaces (spaces and tabs) are ignored
# comments start with a "#" and may be used inline
# 
# example: option=value # i like this option
#
# @filename includes mod_config_path/filename
# @/absolute/path/to/file includes file

############################################################
# Network configuration

# If only one signaling (SIP) and media (RTP) interface is
# to be used, configure the options
#    sip_ip / sip_port / media_ip/ rtp_low_port /
#    rtp_high_port / public_ip / sig_sock_opts
# If more than one interface is to be used, configure the option
# 'interfaces' and those options for each interface.

############# configuration for single interface ############
# optional parameter: sip_ip=<ip_address>|<device>
#
# - SIP IP where the SIP stack is bound to. This also sets 
#   the value used for contact header in outgoing calls and 
#   registrations.
# - If neither 'media_ip' nor 'sip_ip' are set, defaults 
#   to first non-loopback interface, and the port configured below
#   is ignored.
#
# Example:
#  sip_ip=10.0.0.34
#  sip_ip=en0

# optional parameter: sip_port=<port_number>
#
# - port where its SIP stack should be bound to, ignored if sip_ip not set.
#   default: 5060
#
#sip_port=5080

# optional parameter: media_ip=<ip_address>|<device>
#
# - IP address or interface that SEMS uses to send and receive media.
# - If neither 'media_ip' nor 'sip_ip' are set, defaults
#   to first non-loopback interface. If 'sip_ip' is set,
#   'media_ip' defaults to 'sip_ip.
#
# Examples:
#  media_ip=10.0.0.34
#  media_ip=eth0

# optional parameter: rtp_low_port=<port>
#
# - sets lowest for RTP used port (Default: 1024)
rtp_low_port=10000

# optional parameter: rtp_high_port=<port>
#
# - sets highest for RTP used port (Default: 0xffff)
rtp_high_port=60000

# optional parameter: public_ip=<ip_address>
# 
# - near end NAT traversal. when running SEMS behind certain static
#   NATs, use this parameter to inform SEMS of its public IP
#   address. If this parameter is set, SEMS will write this value
#   into SDP bodies and Contact.
#   If this parameter is not set, the local IP address is used.
#   N.B., there is no support for port translation; the local 
#   RTP port is advertised in SDP in either case.
#   
# Example:
#  public_ip=75.101.219.48
#  

# optional parameter: sig_sock_opts=option,option,option,...
#
# - signaling socket options
#      o force_via_address         - force sending replies to 1st Via
#      o no_transport_in_contact   - do not add transport to contact in replies
#
# Example:
#  sig_sock_opts=force_via_address,no_transport_in_contact

# optional parameter: tcp_connect_timeout=<timeout in millisec>
# Default: 2000 (2 sec)

# optional parameter: tcp_idle_timeout=<timeout in millisec>
# Default: 3600000 (1 hour)

############# configuration for multiple interfaces ############
#   interfaces = <list of interface names>
#
# 'interfaces' must be set if more than one interface is to be used
# for the same purpose (e.g. more than one interface for SIP).
# Configure additional interfaces if networks should be bridged or
# separate networks should be served.
#
# For each interface, a set of parameters suffixed with the
# interface name should be configured.
#
# Please note that for each additional interface,
# 'sip_ip_[if_name]' is mandatory (but can be the interface
# name, then the first assigned IP is used). The other
# parameters are optional. 'media_ip_[if_name]'
# is derived from 'sip_ip_[if_name]' if not set.
# 'public_ip_[ip_name]' is also based on 'sip_ip_[if_name]'
# if not set explicitly.
# 
# Example:
interfaces=rhizo,vsat
#
sip_ip_rhizo=<%= @vpn_ip_address %>
sip_port_rhizo=5095
media_ip_rhizo=<%= @vpn_ip_address %>
rtp_low_port_rhizo=12000
rtp_high_port_rhizo=13000
#
sip_ip_vsat=<%= @vsat_ip_address %>
sip_port_vsat=5090
media_ip_vsat=<%= @vsat_ip_address %>
rtp_low_port_vsat=14000
rtp_high_port_vsat=15000
#sig_sock_opts_vsat=force_via_address
tcp_connect_timeout_vsat=1000
tcp_idle_timeout_vsat=900000

############# other network configuration ############################## 

# NAT handling for SIP:sip_nat_handling={yes|no}
#
# Learn remote next hop address from the source of the address where
# requests are received from. This option does not apply to the sbc module.
#
# default: no
#
#sip_nat_handling=yes

# NAT handling for RTP: force_symmetric_rtp={yes|no}
#
# Force comedia style "symmetric RTP" NAT handling, i.e.
# learn remote RTP address from where RTP packets come from
# (This option does not apply to the sbc module's RTP relay)
#
# default: no
#
force_symmetric_rtp=no

# optional parameter: outbound_proxy=uri
#
# - this sets an outbound proxy for dialogs and registrations initiated 
#   by SEMS.  A preloaded Route header containing the uri is added to
#   each initial request.  The request is then sent to destination obtained
#   by resolving the uri. If outbound_proxy is not set (default setting),
#   no preloaded Route header is added and request is sent to destination
#   obtained by resolving r-uri.  Resolving is done by SIP stack with DNS
#   if uri contains domain name. Warning: If uri can not be resolved, no  
#   requests will be sent out at all!
#  
#   default: empty
#
# Example:
#   outbound_proxy=sip:proxy.mydomain.net

# optional parameter: force_outbound_proxy={yes|no}
#
# - forces SEMS to set outbound_proxy for any requests (not just for
#   registrations and dialog initiating requests). See above what setting
#   of outbound_proxy means.  This option will only have an effect if the
#   outbound_proxy option has been set, and it will break 3261 compatibility
#   in some cases; better use next_hop.
#
#   default: no
#
# Example:
#   force_outbound_proxy=yes

# optional parameter: next_hop=address[:port][/transport]
# - if this is set, all outgoing requests will be sent to
#   this address (IP address or domain name), regardless of R-URI etc.
#
# Examples:
#   next_hop=192.168.5.1
#   next_hop=foo.example.com:5080/tcp

# optional parameter: next_hop_1st_req={yes|no}
# - if set to yes, next_hop behavior (routing without pre-loaded route set)
#   applies only to initial request.  Subsequent requests are routed
#   normally based on route set learned from reply to initial request.
#
#   default: no
#
# Example:
#   next_hop_1st_req=yes

# optional parameter:next_hop_for_replies
# - use next_hop for replies, too?
#
#next_hop_for_replies=yes

# RTP multiplexing configuration - see doc/rtp_mux/
#
# IP:port to open an incoming (listening) RTP MUX port

rtp_mux_ip=<%= @vsat_ip_address %>
rtp_mux_port=<%= @rtp_mux_port %>

#
# Interface to send outgoing RTP MUX packets from (port is auto-assigned)

rtp_mux_out_interface=vsat

#
# Parameters for outgoing mux:
#   MTU threshold: frame is sent if over this threshold; leave 1 RTP payload size to actual MTU
rtp_mux_mtu_threshold=135
#   max frame age: .. or this old (millisec), i.e. max delay introduced
rtp_mux_max_frame_age_ms=30
#
# For gatewaying, additionally a_rtp_mux_ip/a_rtp_mux_port or b_rtp_mux_ip/
# b_rtp_mux_port must be configured in the SBC profile.
#

############################################################
# modules and application configuration
#
# Configuration of plugin (module) loading:
# - if load_plugins is set, only those are loaded.
# - if load_plugins is not set, all modules in the plugin_path
#   directory are loaded, except those which are listed 
#   in exclude_plugins.
#

# optional parameter: plugin_path=<path>
# 
# - sets the path to the plug-ins' binaries
# - may be absolute or relative to CWD
plugin_path=/usr/lib/sems/plug-in/

# optional parameter: load_plugins=<modules list>
# 
# semicolon-separated list of modules to load.
# If empty, all modules in plugin_path are loaded.
#
# example for announcement with only g711 and ilbc codecs  
# load_plugins=wav;ilbc;announcement
load_plugins=wav;isac;l16;speex;g722;gsm;ilbc;stats;sbc;xmlrpc2di

# optional parameter: exclude_plugins=<modules list>
#
# semicolon-separated list of modules to exclude from loading
# ('blacklist'). If empty, all modules in plugin_path are loaded.
# This has only effect it load_plugins is not set.
#
# o precoded_announce: no precoded sample files present
# o py_sems: conflicts with ivr (in some cases)
# o db_reg_agent: needs DB tables
# o cc_* : sbc call control modules (loaded from sbc.conf)
exclude_plugins=precoded_announce;py_sems;db_reg_agent;cc_call_timer;cc_ctl;cc_pcalls;cc_prepaid;cc_prepaid_xmlrpc;cc_rest;cc_syslog_cdr

# optional: load_plugins_rtld_global=<modules list>
#
# load these plugins with RTLD_GLOBAL (by default py_sems,
# dsm, ivr, sbc, diameter_client, registrar_client, uac_auth)
#
#load_plugins_rtld_global=myapp

# optional parameter: application
# 
# This controls which application is to be executed for incoming calls
# is no explicit application requested (outgoing calls)
#
# This can be one of 
#    $(ruri.user)       - user part of ruri is taken as application,
#                         e.g. sip:announcement@host
#    $(ruri.param)      - uri parameter "app", e.g. 
#                         sip:joe@host.net;app=announcement
#    $(apphdr)          - the value of the P-App-Name header is used
#
#    $(mapping)         - regex=>application mapping is read from
#                         app_mapping.conf (see app_mapping.conf)
#    <application name> - application name configured here, e.g.
#                         application=announcement
#   
# examples:
# application = conference
# application = $(mapping)
# application = $(ruri.user)
# application = $(ruri.param)
# application = $(apphdr)
application=sbc

# parameter: plugin_config_path=<path>
#
# - in this path configuration files of the applications 
#   (e.g. announcement.conf) are searched
plugin_config_path=/etc/sems/etc/

# optional parameter: exclude_payloads=<payload list>
#
# semicolon-separated list of payloads to exclude from loading
# ('blacklist'). 
# 
# For example, to only use low bandwidth codecs:
#  exclude_payloads=PCMU;PCMA;G726-40;G726-32;G721;L16
# or, to use only codecs which are not CPU-intensive:
#  exclude_payloads=iLBC;speex;
# only use G711 (exclude everything else):
#  exclude_payloads=iLBC;speex;G726-40;G726-32;G721;G726-24;G726-16;GSM;L16

############################################################
# logging and running

# optional parameter: fork={yes|no}
# 
# - specifies if sems should run in daemon mode (background)
#   (fork=no is the same as -E)
fork=yes

# optional parameter: stderr={yes|no}
#
# - debug mode: do not fork and log to stderr
#   (stderr=yes is the same as -E)
stderr=no

# optional parameter: loglevel={0|1|2|3}
#
# - sets log level (error=0, warning=1, info=2, debug=3)
#   (same as -D)
loglevel=2

# optional parameter: syslog_facility={DAEMON|USER|LOCAL[0-7]}
#
# - sets the log facility that is used for syslog. Using this,
#   the log can for example be filtered into a special file 
#   by the syslog daemon.
#
# Default: DAEMON
#
# Example:
# syslog_facility=LOCAL0

# optional parameter: log_sessions=[yes|no]
# 
# Default: no
# 
# If log_sessions=yes is set, INFO level log messages are generated
# for each session when it is started and stopped.
#
# log_sessions=yes

# optional parameter: log_events=[yes|no]
# 
# Default: no
# 
# If log_eventy=yes is set, generic DBG level log messages are 
# generated for each event that is posted into an event queue.
#
# log_events=yes

# optional parameter: max_shutdown_time=<time in seconds>
#
# Limit on server shutdown time (time to send/resend BYE
# to active calls). 0 to disable (infinite).
#
# Default: 10
#
#max_shutdown_time = 10

# optional parameter: shutdown_mode_reply="<code> <reason>"
#
# Error reply that is used as reply to INVITE and OPTION
# when SEMS is shutting down.
#
# Default: shutdown_mode_reply="503 Server shutting down"

# optional parameter: cps_limit=<limit>;<err code>;<err reason>
# 
# - this sets a maximum calls per sec limit. If that limit is 
#   reached, no further calls are accepted, but the error reply 
#   with err code/err reason is sent out.
# 
# Default: 0 (None)
#
# Example:
#  cps_limit="100;503;Server overload"

###########################################################
# if build with ZRTP support (see Makefile.defs)
# enable ZRTP support in endpoint calls:
#enable_zrtp=yes (default: yes)
#

# enable ZRTP debug log? (prints lots of info)
#enable_zrtp_debuglog=no (default: yes)

############################################################
# tuning

# optional parameter: session_processor_threads=<num_value>
# 
# - controls how many threads should be created that
#   process the application logic and in-dialog signaling. 
#   This is only available if compiled with threadpool support!
#   (set USE_THREADPOOL in Makefile.defs)
#   Defaults to 10
#
# session_processor_threads=50

# optional parameter: media_processor_threads=<num_value>
# 
# - controls how many threads should be created that
#   process media - on single-processor systems set this 
#   parameter to 1 (default), on MP systems to a higher
#   value
#
# media_processor_threads=1

# optional parameter: rtp_receiver_threads=<num_value>
#
# - controls how many threads should be created that
#   receive and relay RTP media - on single-processor systems set this
#   parameter to 1 (default), on MP systems to a higher value.
#
# rtp_receiver_threads=1

# optional parameter: session_limit=<limit>;<err code>;<err reason>
# 
# - this sets a maximum active session limit. If that limit is 
#   reached, no further calls are accepted, but the error reply 
#   with err code/err reason is sent out.
# 
# Default: 0 (None)
#
# Example:
#  session_limit="1000;503;Server overload"

# optional parameter: options_session_limit=<limit>;<err code>;<err reason>
# 
# - this sets a custom response to OPTIONS, if the session count reaches
#   a certain limit. This way health monitor could raise an alarm to syste
#   administrator. 
# 
# Default: 0 (None)
#
# Example:
#  options_session_limit="900;503;Warning, server soon overloaded"

# optional parameter: dead_rtp_time=<unsigned int>
#
# - if != 0, after this time (in seconds) of no RTP
#   a session is considered dead and stopped. If set
#   to 0 no check is done for rtp timeout.
#
#   default=300 (5 minutes)
#
# Examples:
#    # disable RTP timeout
#    dead_rtp_time=0  
#    # RTP timeout after 10 seconds
#    dead_rtp_time=10  

# optional parameter: use_default_signature={yes|no}
#
# - use a Server/User-Agent header with the SEMS server 
#   signature and version.
#
#   default=no
#
use_default_signature=yes

# optional parameter: signature=<signature string>
#
# - use a Server/User-Agent header with a custom user agent
#   signature. Overridden by default signature if
#   use_default_signature is set.
#
# signature="SEMS media server 1.0"

# optional parameter: single_codec_in_ok={yes|no}
#
# - use single codec in 200 OK response
#
#   default=no
#
# single_codec_in_ok=no

# optional parameter: codec_order=codec_name_1,codec_name2,...
#
# - Codec order used when sending INVITE requests.  Codecs in codec_order
#   will be on the top of the list followed by other supported codecs
#   (if any).
#
#   default=empty
#
# codec_order=iLBC,GSM

# optional parameter: ignore_rtpxheaders={yes|no}
#
# - if this is set to yes, RTP extension headers (e.g. when using ZRTP)
#   are ignored. If set to no, the whole RTP packets with extension 
#   headers will be ignored and a debug message is printed on every 
#   received packet.
#
#   default=no
#
# ignore_rtpxheaders=yes

# optional parameter: dtmf_detector={spandsp|internal}
#
# sets inband DTMF detector to use. spandsp support must be compiled in
# for this to have effect if dtmf_detector=spandsp.
#  
# default: internal
#
# dtmf_detector=spandsp

# optional parameter: unhandled_reply_loglevel={error|warn|info|debug|no}
# 
# the default application logic implemented in the applications is to stop 
# the session right after sending BYE, without waiting for a reply. this 
# leads to many log entries of the form 
# ERROR: [b6fa6bb0] handleSipMsg (AmSipDispatcher.cpp:48): unhandled 
#    reply:  [code:200;phrase:[OK];... ]
#
# This parameter sets the log lovel of unhandled positive (200 class) replies.
#
# default: error
# 
# unhandled_reply_loglevel=info

############################################################
# SIP stack settings
#   default settings (i.e. leave out) for these should be OK
#   for most applications

# Force use of the selected interface? (Default: no)
#
# Use IP_PKTINFO to force use of the selected interface for sending
# packets. Useful in situations with overlapping IP address spaces, when
# the kernel routing table should not be used to determine the sending
# interface.
#
# force_outbound_if=yes

# SIP timers configuration (in milliseconds)
#
# sip_timer_A=<n millisec>
# sip_timer_B=<n millisec>
# ...
# sip_timer_M=<n millisec>
#
# timers A to J as in RFC3261.
# Timer L: handle 200 ACKs automatically in INVITE client trans.
# Timer M: cycle throught multiple addresses in case the R-URI
#          resolves to multiple addresses
#
# Warning: Timer values are not checked whether they are appropriate!
#          Leave to default values if in doubt.
#
# Example:
#  # equivalent to fr_timer=20 in sip-router
#  sip_timer_b=20000
#  sip_timer_f=20000

# sip_timer_t2=<n millisec)   T2 timer configuration
#  (Cap for re-send request/response backoff)
#
# Warning: Timer values are not checked whether they are appropriate!
#          Leave to default values if in doubt.
#
#sip_timer_t2=4000

# skip DNS SRV lookup? [yes, no]
#
# according to RFC, if no port is specified, destination IP address
# should be resolved with a DNS SRV lookup. If SEMS should not do that
# (only an A record lookup), set disable_dns_srv=yes.
#
# Default: no
#
#disable_dns_srv=yes

# support 100rel (PRACK) extension (RFC3262)? [disabled|supported|require]
#
# disabled - disable support for 100rel
# supported - support it if remote end does, default
# require - required
#
# Default: supported
#
#100rel=require

# force the use of outbound interface? [yes, no]
#
# Useful in case of overlapping networks, or if OS routing can/should not be used. 
# Default: no
#
# force_outbound_if=yes

# use raw sockets for sending? [yes, no]
# faster, requires root or CAP_NET_RAW
#
# Default: no
#
# use_raw_sockets=yes

#
# accept forked dialogs on UAS side? [yes|no]
#
#  no - INVITE with existing callid+remote_tag is replied with 482.
# yes - INVITE with existing callid+remote_tag+via_branch is replied with 482.
#       Forked INVITEs (!= via-branch) are accepted. 
#
# Default: yes
#
#accept_forked_dialogs=no

# Make SIP authenticated requests sticky to the proxy? [yes | no]
#
# If enabled, host of request-URI of out-of-dialog requests that are 
# authenticated with SIP auth is changed to the previously resolved
# next-hop IP:port.
#
# default: no
#
# proxy_sticky_auth=yes

# Ignore too low CSeq for NOTIFYs? [yes | no]
#
# May be necessary to interwork with simplistic/old SIP event notification 
# implementations.
#
#ignore_notify_lower_cseq=yes

#
# Accept final replies without To-tag? [yes|no]
#
accept_fr_without_totag=yes

#
# Log raw messages?  [no|debug|info|warn|error]
#
# Default: debug
#
#log_raw_messages=no

#
# Log parsed received messages?  [yes|no]
#
# Default: yes
#
#log_parsed_messages=no

# SIP UDP socket receive buffer size (in bytes)
#
# if not set, system default is used (which usually
# is modest). set sytem wide upper limit with
# e.g. sysctl -w net.core.rmem_max=8388608
#
# udp_rcvbuf = <value>

# Number of SIP UDP receiver threads
#
# Default: 4
#
# sip_server_threads=8

# dump conference streams - experimental
# play with: $play -r <samplerate> -c 1 /tmp/123_1_nnnn.s16 
#  where <samplerate> is in /tmp/123_1_nnnn.s16.samplerate
#  (if it hasn't changed in-between...)
#dump_conference_streams=true
#dump_conference_path=/tmp/