75 lines
4.9 KiB
XML
75 lines
4.9 KiB
XML
<?xml version="1.0" encoding="utf-8"?>
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<resources>
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<string name="app_name">baresip</string>
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<string name="aboutText">
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<b>Baresip based Android application</b>\n\n
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Juha Heinanen jh@tutpro.com\n\n
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Version %1$s\n\n
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<b>Usage Hints</b>\n\n
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• Check that config default values meet your needs (click item titles for help).\n
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• Then create one or more accounts (again click item titles for help).\n
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• Peers of calls and messages can be added to contacts by long clicks.\n
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• Long clicks can also be used to remove calls, chats, messages, and contacts.\n
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• Call click when callee has not been given can be used for re-dial.\n
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• If you cannot hear the other party of a call, try to increase Media volume.\n
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</string>
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<string name="noAccounts">No accounts</string>
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<string name="dispName">Name (if any) used in From URI of outbound requests.</string>
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<string name="authUser">Authentication username if required by outbound proxy.</string>
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<string name="authPass">Authentication password if required by outbound proxy.</string>
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<string name="obProxies">SIP URI of one or two proxies that must be used when sending requests.
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If two is given, REGISTER requests are sent to both and other requests are sent to
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one that responds. If no outbound proxy is given, requests are sent based on
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DNS NAPTR/SRV/A record lookup of callee URI hostpart. If hostpart of SIP URI is an IPv6
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address, the address must be written inside brackets [].\nExamples:\n
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• sip:foo.com:5060;transport=tls\n
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• sip:[2001:67c:223:777::10]:5060;transport=tcp
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</string>
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<string name="register">If checked, registration is enabled and REGISTER requests are sent at
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12 minute intervals.
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</string>
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<string name="ice">If checked, ICE (Interactive Connectivity Establishment) is enabled for NAT
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traversal.
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</string>
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<string name="stunServer">A specific Stun Server used by ICE of form host[:port].
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Factory default value is \'stun.l.google.com:19302\', pointing to public Google STUN server.
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Username and password are currently not supported.
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</string>
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<string name="auCodecs">List of supported audio codecs in priority order.</string>
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<string name="mediaEnc">Selects media transport encryption protocol.\n
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• ZRTP (recommended) means that ZRTP end-to-end media encryption negotiation is tried after
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the call has been established.\n
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• DTLS-SRTPF means that UDP/TLS/RTP/SAVPF is offered in outgoing call and that RTP/SAVP,
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RTP/SAVPF, UDP/TLS/RTP/SAVP, or UDP/TLS/RTP/SAVPF is used if offered in incoming call.\n
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• SRTP-MANDF means that RTP/SAVPF is offered in outgoing call and required in incoming call.\n
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• SRTP-MAND means that RTP/SAVP is offered in outgoing call and required in incoming call.\n
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• SRTP means that RTP/AVP is offered in outgoing call and that RTP/SAVP or RTP/SAVPF is used
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if offered in incoming call.\n
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• None means that media encryption is not supported and that media transport protocol
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thus is RTP/AVP.
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</string>
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<string name="vmUri">SIP URI for checking of voicemail messages. If left empty, voicemail
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messages (Message Waiting Indications) are not subscribed to.
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</string>>
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<string name="defaultAccount">If checked, this account is selected when baresip is started.
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</string>>
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<string name="autoStart">If checked, baresip starts automatically after device (re)start.</string>
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<string name="listenAddress">IP address and port of form \'address:port\' at which baresip listens
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for incoming SIP requests. If IP address is an IPv6 address, it must be written inside
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brackets []. IPv4 address 0.0.0.0 or IPv6 address [::] makes baresip listen at all
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available addresses. If left empty (factory default), baresip listens at port 5060 of
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all available addresses.
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</string>
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<string name="preferIPv6">Prefer IPv6 if both IPv4 and IPv6 are available.</string>
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<string name="dnsServers">Comma separated list of DNS servers. Each DNS server is of form
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\'server:port\'. If server is an IPv6 address, the address must be written inside brackets [].
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Factory default value is \'8.8.8.8:53,[2001:4860:4860::8888]:53\' pointing to public Google
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DNS servers.</string>
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<string name="opusBitRate">Average maximum bit rate used by Opus audio stream.
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Valid value are 6000-510000. Factory default is 28000.</string>
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<string name="iceLite">If checked, ICE Lite Mode is used.</string>
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<string name="callVolume">If set, default call audio volume at scale 1-10.</string>
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<string name="debug">If checked, debug and info level log messages are available in logcat.</string>
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<string name="reset">If checked, configuration is reset to factory default.</string>
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</resources>
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