Aquilenet » Services » SIP

[auth_info_0]

#SIP username

username=thomas

#sip userid (usually the same as username, don't specify unless you know what you are doing)

userid=thomas

#password associated with above username, userid and realm

passwd=thomas

#SIP authentication realm (= authentication domain), can be left empty if realm is not known.

realm=""  

  ##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=0

    #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=2

    #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="Bob"

    #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=proxy_0

    #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

[proxy_0]

#SIP address of the proxy

reg_proxy=persephone.aquilenet.fr

#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.

##

#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

#