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Section [SIP Phone]

Thomas edited this page Feb 6, 2015 · 1 revision
Key Description mandantory default typ
server SIP-Server to connect yes string
username username for SIP-Server yes string
password password for SIP-Server yes string
realm realm for SIP-Server (mostly the domain of SIP-Server) yes string
sipphonetyp current only pjsua no autodetect string
call_timeout TODO max. Zeit in Sekunden ab Beginn des Anrufs und max. bis jemand das Gespräch entgegennimmt no 30 integer
max_call_time TODOmax. Gesprächszeit in Sekunden ab dem Punkt "Gespräch wurde entgegengenommen" Soll verhindern, dass Gespräche ewig laufen weil vergessen wurde aufzulegen no 120 integer
ua.max_calls maximum number of calls to be supported. no 1 integer
ua.nameserver list of nameserver hostnames or IP addresses. Nameserver must be configured if DNS SRV resolution is desired. no [] string getrennt mit ,
ua.stun_domain if nameserver is configured, this can be used to query the STUN server with DNS SRV. no string
ua.stun_host the hostname or IP address of the STUN server. This will also be used if DNS SRV resolution for stun_domain fails. no string
media.audio_frame_ptime specify the length of audio frames in millisecond no 20 integer
media.channel_count specify the number of channels to open the sound device and the conference bridge no 1 integer
media.clock_rate specify the core clock rate of the audio, most notably the conference bridge. no 8000 integer
media.ec_options Echo Canceller option (specify zero) no 1 integer
media.ec_tail_len specify Echo Canceller tail length in milliseconds. Value zero will disable the echo canceller. no 512 integer
media.enable_ice enable Interactive Connectivity Establishment (ICE) no True bool
media.enable_turn enable TURN relay. TURN server settings must also be configured. no False bool
media.ilbc_mode specify iLBC codec mode (must be 30 for now) no 30 typ
media.jb_max specify the maximum jitter buffer size in milliseconds. Put -1 for default. no -1 integer
media.jb_min specify the minimum jitter buffer size in milliseconds. Put -1 for default. no -1 integer
media.max_media_ports specify maximum number of audio ports to be supported by the conference bridge. no 32 integer
media.no_vad disable Voice Activity Detector (VAD) or Silence Detector (SD) no False bool
media.ptime specify the audio packet length of transmitted RTP packet. no 0 integer
media.quality specify the audio quality setting (1-10) no 10 integer
media.tx_drop_pct randomly drop transmitted RTP packets (for simulation). Number is in percent. no 0 integer
media.rx_drop_pct randomly drop received RTP packets (for simulation). Number is in percent. no 0 integer
media.snd_auto_close_time specify the duration in seconds when the sound device should be closed after inactivity period. no 5 integer
media.snd_clock_rate optionally specify different clock rate for the sound device. no 0 integer
media.turn_conn_type specify connection type to the TURN server, from the TURNConnType constant. no 17 integer
media.turn_server specify the domain or hostname or IP address of the TURN server, in "host[:port]" format. no string
log.level specify the input verbosity level no 1 integer
log.console_level specify the output verbosity level. no 1 integer
transport.port port number. no 0 integer
transport.bound_addr optionally specify the address to bind the socket to. Default is empty to bind to INADDR_ANY. no string
transport.public_addr optionally override the published address for this transport. If empty, the default behavior is to get the public address from STUN or from the selected local interface. Format is "host:port". no string