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