baresip baresip+ About baresip About baresip+ Baresip library based SIP User Agent

Juha Heinanen <jh@tutpro.com>

Version %1$s

Usage Hints

Privacy Policy

Privacy policy is available here.

Source code

Source code is available at GitHub, where also issues can be reported.

Licenses

]]> Baresip library based SIP User Agent with video calls

Juha Heinanen <jh@tutpro.com>

Version %1$s

Usage Hints

Known Issues

Privacy Policy

Privacy policy is available here.

Source code

Source code is available at GitHub, where also issues can be reported.

Licenses

]]> Account Account Nickname Nickname (if any) used to identify this account within baresip app. Nickname Invalid Account Nickname \'%1$s\' Nickname \'%1$s\' already exists Display Name Your Name Name (if any) used in From URI of outbound requests. Invalid Display Name \'%1$s\' Authentication Username Authentication username if authentication of SIP requests is required. Default value is account\'s username. Invalid Authentication Username \'%1$s\' Authentication Password Authentication Password up to 64 characters. If Authentication Username is given, but Password is not given, it will be asked when baresip is started. Invalid Authentication Password \'%1$s\' Outbound Proxies SIP URI of one or two proxies that must be used when sending requests. If two is given, REGISTER requests are sent to both and other requests are sent to one that responds. If no outbound proxy is given, requests are sent based on DNS NAPTR/SRV/A record lookup of callee URI hostpart. If hostpart of SIP URI is an IPv6 address, the address must be written inside brackets []. \nExamples: \n • sip:example.com:5061;transport=tls \n • sip:[2001:67c:223:777::10];transport=tcp \n • sip:192.168.43.50:443;transport=wss SIP URI of Proxy Server SIP URI of another Proxy Server Invalid Proxy Server URI \'%1$s\' Register If checked, registration is enabled and REGISTER requests are sent at the interval specified by Registration Interval. Registration Interval Tells how often (in seconds) baresip sends REGISTER requests. Valid values are from 60 to 3600. Invalid Registration Interval\'%1$s\' Media NAT Traversal Selects media NAT traversal protocol (if any). Possible choices are STUN (Session Traversal Utilities for NAT, RFC 5389) and ICE (Interactive Connectivity Establishment, RFC 5245). STUN/TURN Server A STUN/TURN Server URI of form scheme:host[:port][\?transport=udp|tcp], where scheme is \'stun\', \'stuns\', \'turn\', or \'turns\'. Factory default STUN Server for STUN and ICE protocols is \'stun:stun.l.google.com:19302\' pointing to public Google STUN server. There is no factory default TURN server. STUN/TURN Server URI Invalid STUN/TURN Server URI \'%1$s\' stun:stun.l.google.com:19302 STUN/TURN Username Username if required by STUN/TURN server Invalid Username \'%1$s\' STUN/TURN Password Password if required by STUN/TURN server Invalid Password \'%1$s\' Media Encryption Selects media transport encryption protocol (if any). \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. Prefer IPv6 Media If checked, offer to use IPv6 media protocol (if available) when media protocol of peer cannot be automatically determined. RTCP Multiplexing If checked, RTP and RTCP packets are multiplexed on a single port (RFC 5761). Reliable Provisional Responses If checked, indicate support for reliable provisional responses (RFC 3262). DTMF Mode Selects how DTMF tones 0–9, #, *, and A-D are sent. In-band RTP Events SIP INFO Requests In-band RTP or SIP INFO Answer Mode Selects how incoming calls are answered. Redirect Mode Selects if call redirect request is followed automatically or if confirmation is requested. Manual Automatic Voicemail URI SIP URI for checking of voicemail messages. If left empty, voicemail messages (Message Waiting Indications) are not subscribed to. Invalid Voicemail URI \'%1$s\' Country Code E.164 country code of this account. If From URI userpart of incoming call or message contains a telephone number that does not start with \'+\' sign and if contact lookup fails, the number is prefixed with this country code and contact lookup is tried again. If the telephone number starts with a single digit \'0\', digit \'0\' is removed before the number is prefixed. +code Invalid Country Code \'%1$s\' Telephony Provider SIP URI host part used in calls to telephone numbers. Factory default is account\'s domain. If not given, this account cannot be used to call telephone numbers. SIP URI host part Invalid SIP URI host part \'%1$s\' Default Account If checked, this account is selected when baresip is started. Accounts New Account When a new account is created, account\'s port number and transport protocol may be optionally given: <user>@<domain>[:<port>][;transport=udp|tcp|tls]. If <port> is given and transport protocol is not given, transport protocol defaults to udp. If <port> is not given and transport protocol is given, <port> defaults to 5060 or 5061 (TLS). If neither is given and no outbound proxy is specified, account\'s registrar (if any) is determined solely based on domain\'s DNS information. user@domain Invalid user@domain[:port][;transport=udp|tcp|tls] \'%1$s\' Account \'%1$s\' already exists. "Failed to allocate new account. Encrypt Password Decrypt Password Do you want to delete account \'%1$s\'\? Answer Reject Incoming call from Missed call from Missed calls %1$d missed calls Call transfer request to Message from Auto-rejected call from \`%1$s\` Call History Call Details Call calls call Peer Direction Time Duration Do you want to call or send message to \'%1$s\'\? Do you want to add \'%1$s\' to contacts or delete %2$s from call history\? Do you want to delete \'%1$s\' %2$s from call history\? Delete Disable Enable Do you want to delete call history of account \'%1$s\'\? Chat Messages Chat with %1$s New message Do you want to delete message or add peer \'%1$s\' to contacts\? Do you want to delete message\? Add Contact Sending of message failed Failed Chat History Today You New Chat Peer Invalid chat peer URI Do you want to delete chat with peer \'%1$s\' or add peer to contacts\? Do you want to delete chat with \'%1$s\'\? Delete Do you want to delete chat history of account \'%1$s\'\? Audio Codecs List of audio codecs in priority order. Drag to reorder, swipe right to enable or disable. Video Codecs List of video codecs in priority order. Drag to reorder, swipe right to enable or disable. Reorder Settings Start Automatically If checked, baresip starts automatically after device (re)start. Automatic start needs Appear on Top permission. Battery Optimizations Disable battery optimizations (recommended) if you want to reduce likelihood that Android restricts baresip\'s access to network or enters baresip to standby state. Default Phone App Dialer role is not available If checked, baresip is the default phone app. Do not check if your device may need to handle also other than SIP calls or messages. Listen Address 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. Invalid Listen Address 0.0.0.0:5060 Address Family Chooses which IP addresses baresip is using. If IPv4 or IPv6 is chosen, baresip uses only IPv4 or IPv6 addresses. If neither is chosen, baresip uses both IPv4 and IPv6 addresses. DNS Servers Comma separated list of addresses of DNS servers. If not given, DNS server addresses are obtained dynamically from the system. Each DNS address is of form \'ip:port\' or \'ip\'. If port is omitted, it defaults to 53. If ip is an IPv6 address and also port is given, ip must be written inside brackets []. As an example, list \'8.8.8.8:53,[2001:4860:4860::8888]:53\' points to IPv4 and IPv6 addresses of public Google DNS servers. Invalid DNS Servers Failed to set DNS servers TLS Certificate File If checked, a file containing TLS certificate and private key of this baresip instance has been or will be loaded. In Android versions 9 and below, a file called \'cert.pem\' is loaded from Download folder. For security reasons, delete the file after loading. Verify Server Certificates If checked, baresip verifies TLS certificates of SIP User Agent and SIP Proxy Servers when TLS transport is used. TLS CA File If checked, a file has been or will be loaded that contains TLS certificates of such Certificate Authorities that are not included in Android OS. In Android versions 9 and below, a file called \'ca_certs.crt\' is loaded from Download folder. Audio Settings Speaker Phone If checked, speaker phone is turned automatically on when call starts. Audio Modules Audio codecs provided by the checked modules are available for use by the accounts. Failed to load module. Acoustic Echo Cancellation If checked, echo cancellation is attempted on call audio. AEC Extended Filter If checked, echo cancellation is using extended filter. Opus Bit Rate Average maximum bit rate used by Opus audio stream. Valid values are 6000-510000. Factory default is 28000. 28000 Expected Opus packet-loss Expected Opus audio stream packet loss percentage, from 0–100. Factory default value is 1. Value 0 also turns off Opus Forward Error Correction (FEC). 0 Invalid Opus bitrate Invalid Opus Packet Loss Percentage Audio Delay Time (in milliseconds) to wait audio from callee when call is established. Set to a higher value if you miss audio from callee at the beginning of the call. Invalid Audio Delay \'%1$s\'. Valid values are from 100 to 3000. Default Call Volume If set, default call audio volume at scale 1–10. Tone Country Country of call ringing, waiting, and callee busy tones Dark Theme Force dark display theme Video Frame Size Size of transmitted video frames (width x height). Factory default is 800x600. Chooses if baresip contacts, Android contacts, or both are used. If both and contact with the same name exists in both, baresip contact is chosen. Both Debug If checked, provides debug and info level log messages to Logcat. SIP Trace If checked and if Debug is checked, Logcat messages include also SIP request and response trace. Unchecked automatically at baresip start. Reset to Factory Defaults If checked, settings are reset to factory default values. Are you sure you want to reset settings to factory default values\? Reset Failed to read file \'cert.pem\'. Failed to read file \'ca_certs.crt\'. You need to restart baresip in order to activate the new settings. Restart now\? Consent Request If Android contacts is chosen, they can be used in calling and messaging as references to SIP and tel URIs. baresip app does not store Android contacts nor share them with anyone. In order to make Android contacts available in baresip, Google requires that you accept their use as described here and in app\'s Privacy Policy. Contact New Contact Name SIP or tel URI user@domain or telephone number Favorite Invalid contact name \'%1$s\' Contact \'%1$s\' already exists. Invalid SIP URI If checked, this contact is added to Android contacts. Profile image Contacts Do you want to call or send message to \'%1$s\'\? Send Message Do you want to delete contact \'%1$s\'\? Your maximum number of contacts %1$d has been exceeded. Alert Info Notice Cancel OK Yes No Accept Deny User ID Password SIP URI Add Delete Edit Send Status Error Help Confirmation Anonymous Unknown … \u2022 %1$s Invalid SIP or tel URI \'%1$s\' baresip Android Backup Restore About Restart Quit Call to … Call from … Diverted by … Transferring call to … Invalid SIP URI \'%1$s\' Account \'%1$s\' has no Telephony Provider Callee Hangup Video call Video Request Accept sending and receiving video with \'%1$s\'\? Accept sending of video to \'%1$s\'\? Accept receiving video from \'%1$s\'\? Call Hold/Unhold Call is on hold Microphone On/Off Recording can be turned on or off only when call is not connected Call Transfer Blind Attended Transfer destination Choose destination URI Transfer Transfer failed DTMF Call Info No info available Duration: %1$s (secs) Codecs Current Rate: %1$s (Kbits/s) Average Rate: %1$s (Kbits/s) Packets Lost Jitter: %1$s (ms) Voicemail Voicemail Messages You have one new message new messages one old message old messages and You have no messages Listen Messages Dialpad You already have an active call. Baresip failed to start. This may be due to an invalid Settings value. Check Listen Address, TLS Certificate File, and TLS CA File. Then restart baresip. Registering of \`%1$s\` failed. Verify Request Do you want to verify SAS <%1$s>\? Transfer Request Do you accept to transfer this call to \'%1$s\'\? Automatic redirection to \'%1$s\'\ Redirect Request Do you accept call redirection to \'%1$s\'\? Call failed Call is closed This call is NOT secure! This call is SECURE, but peer is NOT verified! This call is SECURE and peer is VERIFIED! Do you want to unverify the peer\? Unverify Application data has been backed up to file \'%1$s\'. In Android versions 9 and below, the file is in Download folder. Failed to back up application data to file \'%1$s\'. Check Apps → baresip → Permissions → Storage. Restart Request Application data restored. baresip needs to be restarted. Restart now\? Failed to restore application data. Check that you gave correct password and that the backup file is from this application. In Android versions 9 and below, also check Apps → baresip → Permissions → Storage and that file \'%1$s\' exists in Download folder. Failed to restore application data. Android version 14 and above does not allow restoring data that was backed up before %1$s version %2$s. You are not able to use this application without \"Notifications\" permission. baresip needs \"Microphone\" permission for voice calls. baresip is not able to detect Bluetooth connectivity without \"Nearby devices\" permission. Grant \"Camera\" permission to make or answer video calls. You are not able create backup without \"Storage\" permission. You are not able restore backup without \"Storage\" permission. You are not able to access Android contacts without \"Contacts\" permission. You don\'t have any supported video cameras. Show Password No network connection! Audio focus denied! Permissions rationale baresip needs \"Microphone\" permission for voice calls, \"Nearby devices\" permission for Bluetooth microphone/speaker detection, and \"Notifications\" permission for posting notifications. baresip+ needs \"Microphone\" permission for voice calls, \"Camera\" permission for video calls, \"Nearby devices\" permission for Bluetooth microphone/speaker detection, and \"Notifications\" permission for posting notifications.