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dev.ini
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dev.ini
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##section relative to sound settings
[sound]
#sound device name used for playback, as listed in linphonec by "soundcard list"
playback_dev_id=ALSA: default device
#sound device name used for ringing, as listed in linphonec by "soundcard list"
ringer_dev_id=ALSA: default device
#sound device name used for capture, as listed in linphonec by "soundcard list"
capture_dev_id=ALSA: default device
#Alsa special device name
# This option allows to specify a special ALSA card (as defined in ALSA asoundrc config files)
# to be used by linphone. This card can then be referred by playback_dev_id, ringer_dev_id, capture_dev_id options.
# Use this if you are able to understand asoundrc syntax and you know what you are doing.
#alsadev=
#wav file to play to advertise remote ringing
#remote_ring=/usr/local/share/sounds/linphone/ringback.wav
#wav file to play to advertise incoming calls
#local_ring=/usr/loca/share/sounds/linphone/rings/bigben.wav
#turn on/off echo cancellation
echocancellation=1
#Expected delay of echo in milliseconds
#Use this when you have a fixed latency in the sound hardware.
#This allows to reduce the tail length of the echo canceller, which speeds up convergence
#and reduces complexity of computations.
ec_delay=0
#Tail length of echo canceller in milliseconds.
#Ideally it should be no more than the expected duration of the echo.
ec_tail_len=60
#Frame size for AU-MDF echo canceller algorithm
#This is a parameter internal to the echo canceller, recommended is too keep to its default value.
ec_frame_size=128
#static software gain (linear scale) to be applied to microphone signal
mic_gain=1.0
#static software gain (log scale) to be applied to signal sent to speaker
playback_gain_db=0.0
##Video settings
[video]
#Size of sent video among these names: QCIF, QVGA, CIF, VGA, SVGA
size=cif
#Whether video is enabled:
enabled=1
#You can refine whether it is enabled for display or capture or both
display=1
capture=1
#Show local preview between calls.
show_local=1
#Show local view during calls, in a corner of the video window
self_view=1
#Webcam name for capture
device=V4L2: /dev/video0
##Network settings
[net]
#Estimated download bandwidth in kbit/s
download_bw=1024
#The bandwidth settings are used to control the bitrate of video (and sometimes audio) encoder, as well
#as limiting the size or fps of the sent video.
#Estimated upload bandwidth in kbit/s
upload_bw=1024
#Firewall policy:
# 0: assume there is no nat
# 1: use firewall address supplied in "nat_address" item (discouraged)
# 2: use STUN to discover its own public IP address and ports
# 3: use ICE.
firewall_policy=0
#Network's Maximum Transmission Unit
# Use 0 to allow automatic discovery, otherwise set to a number of bytes.
# This parameter is only meaningful with video streams for which RTP packets are big.
mtu=0
#STUN server address to use when in firewall_policy=2
stun_server=stun.ekiga.net
#Firewall address to use when in firewall_policy=1
nat_address=80.112.33.11
##SIP settings
[sip]
#SIP port used
sip_port=5060
#Discover automatically local IP address
guess_hostname=1
#Contact address when no proxy is used
# The host port is always overriden at runtime if guess_hostname
# is set to 1.
contact="aeris2"
#Incoming call answering timeout
inc_timeout=15
#Use SIP INFO to send DTMFs (digits)
use_info=0
#Use RFC2833 (out of band DTMFs) to send digits
use_rfc2833=0
#Use IPv6. caution: it is exclusive with IPv4.
use_ipv6=0
#Send registers only when network is up
register_only_when_network_is_up=1
#Default proxy to use (the number is the index of the proxy configuration in this config file)
# Use -1 for no proxy.
default_proxy=-1
#Keepalive period in milliseconds for sending out SIP UDP keepalive to the proxies.
keepalive_period=10000
#When answering to SDP offers, select only one codec,
#instead of replying with all matching codecs.
only_one_codec=0
#Send an OPTIONS message before doing outgoing calls
#This is used by Linphone to workaround some NAT problems inherent to SIP.
#This is highly recommended.
ping_with_options=1
#Network state automatic monitoring
# When set to 1, linphone will periodically monitor the network state (by checking whether it is possible
# to reach the internet).
# When the operating system has callbacks to notify such information, you can use
# linphone_core_set_network_reachable() to notify the core, in which case no network monitoring will be done internally.
auto_net_state_mon=1
## RTP settings
[rtp]
#Audio RTP (UDP) port
audio_rtp_port=7078
#Video RTP (UDP) port
video_rtp_port=9078
#Nominal audio jitter buffer size in milliseconds
audio_jitt_comp=60
#Nominal video jitter buffer size in milliseconds
video_jitt_comp=60
#RTP timeout in seconds: when no RTP or RTCP
# packets are received for this period, the running call is
# automatically closed.
nortp_timeout=30
## Audio codec descriptions
# These sections are named audio_codec_X, where X is a number.
# This number identifies the position of the described codec
# in the core's audio codec list.
[audio_codec_0]
# sub-mime type as defined in RFC3551 or codec's specific RFC:
mime=speex
# RTP clock-rate as defined in RFC3551 or codec's specific RFC:
rate=8000
# Tells whether is codec is enabled
enabled=1
# Fmtp (format parameters) string to be sent in SDP for this codec, which
# corresponds usually to what we are prefering to receive.
# RFC3551 or codec's specific RFC describes the allowed parameters.
recv_fmtp=vad=on
## Video codec descriptions
# These sections are named video_codec_X, where X is a number.
# This number identifies the position of the described codec
# in the core's video codec list.
[video_codec_0]
# sub-mime type as defined in RFC3551 or codec's specific RFC:
mime=H264
# RTP clockrate as defined in RFC3551 or codec's specific RFC, usually 90000 for video payloads.
rate=90000
# Tells whether the codec is enabled
enabled=1
# Fmtp (format paramters) string to be sent in SDP for this codec, which
# corresponds usually to what we are prefering to receive.
recv_fmtp=packetization-mode=1
## SIP Proxy configuration
# Like with audio_codec, it is possible to define several proxy configuration in the
# form of [proxy_X] section, where X is a number.
[proxy_0]
#SIP address of the proxy
reg_proxy=sip:example.net
#SIP identity for which you are known on this proxy:
reg_identity=
#Expiration period of the registration in seconds
reg_expires=3600
#Whether to send a register or not
reg_sendregister=0
#Route: SIP server address to send all outgoing SIP requests
#It is usually left blank, otherwise it is commonly used to specify this proxy
#must be used as an outbound proxy, for example:
# reg_route=sip:example.net
reg_route=
#Send a PUBLISH request to the proxy to notify about presence information (online, busy, out to lunch)
publish=0
#whether "+" in phone numbers should be replaced by 00
dial_escape_plus=0
#Phone number prefix to be applied to entered destinations.
#Example: prefix=+33
prefix=
## Authentication information
# Similarly, several auth_info_X can be defined
# Authentication information is kept distinct from proxy information
# because there can be authentication challenges from proxies or user
# agents even if we are not registered to any proxy.
#[auth_info_0]
#SIP username
#username=bob
#sip userid (usually the same as username, don't specify unless you know what you are doing)
#userid=bob
#password associated with above username, userid and realm
#passwd=mysecret
#SIP authentication realm (= authentication domain), can be left empty if realm is not known.
#realm="example.net"
##
#SIP favourite contacts (friends, buddies...)
# Again it is represented as a family of [friend_X] sections.
#[friend_0]
#SIP address of buddy
#url=Alice
#Policy for incoming SUBSCRIBEs for presence. Can be:
# accept : we accept to share our presence status with this person
# deny : we don't want to share our presence status with this person
#pol=accept
# Send SUBSCRIBEs for presence to this person, so that
# we are notified about her presence information.
#subscribe=1
##
#Other stuff stored in config files that are not configuration items but rather
#persistent information stored in the same place. They are not described here just but
#mentionned for information:
# call_logs_X : call history items
#