#F77445 @color/colorPrimary @color/colorBlack #fcfcfc #673AB7 #000000 @color/colorBlack #383838 #9e9e9e #424242 #e0e0e0 #01df01 @color/colorBlack @color/colorWhite @color/colorSecondaryDark #0ca1fd #0073c9 #6ad2ff #ff0000 #00B9A1 #008873 #5cecd2 @color/colorGray @color/colorGrayLight @color/colorDark @color/colorBlack #4CAF50 #FF5722 #FFEB3B #ffffff @color/colorSecondaryDark 16dp 12dp 0 0.0.0.0:5060 1.0 28000 About Ritosip 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

]]> Ritosip 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

]]> About ritosip About ritosip Accept Account "Failed to allocate new account. Account \'%1$s\' already exists. Account Nickname Nickname (if any) used to identify this account within ritosip app. Accounts When a new account is created, account\'s port number and transport protocol may be optionally given: <user>@<domain>[:<port>][;transport=udp|tcp|tls]. If <port> is given and transport protocol is not given, transport protocol defaults to udp. If <port> is not given and transport protocol is given, <port> defaults to 5060 or 5061 (TLS). If neither is given and no outbound proxy is specified, account\'s registrar (if any) is determined solely based on domain\'s DNS information. Add Add Contact Address Family Chooses which IP addresses ritosip is using. If IPv4 or IPv6 is chosen, ritosip uses only IPv4 or IPv6 addresses. If neither is chosen, ritosip uses both IPv4 and IPv6 addresses. Acoustic Echo Cancellation AEC Extended Filter If checked, echo cancellation is using extended filter. If checked, software echo cancellation is attempted on call audio. Alert Accept sending and receiving video with \'%1$s\'\? Accept receiving video from \'%1$s\'\? Accept sending of video to \'%1$s\'\? and Android If checked, this contact is added to Android contacts. Anonymous Answer Answer Mode Selects how incoming calls are answered. ritosip ritosip Automatic start needs Appear on Top permission. Attended ritosip 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. Audio Codecs List of audio codecs in priority order. Drag to reorder, swipe right to enable or disable. 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. Audio focus denied! Audio codecs provided by the checked modules are available for use by the accounts. Audio Modules ritosip needs \"Microphone\" permission for voice calls, \"Nearby devices\" permission for Bluetooth microphone/speaker detection, and \"Notifications\" permission for posting notifications. Audio Settings Authentication Password Authentication Password up to 64 characters. If Authentication Username is given, but Password is not given, it will be asked when ritosip is started. Authentication Username Authentication username if authentication of SIP requests is required. Default value is account\'s username. Automatic Profile image Average Rate: %1$s (Kbits/s) Application data (excluding recordings) backed up to file \'%1$s\'. In Android version 9, the file is in Download folder. Backup Failed to back up application data to file \'%1$s\'. Check Apps → ritosip → Permissions → Storage. baresip Battery Optimizations Disable battery optimizations (recommended) if you want to reduce likelihood that Android restricts baresip\'s access to network or enters baresip to standby state. Blind Both \u2022 %1$s Call You already have an active call. Auto-rejected call from \`%1$s\` Call is closed Call Details Call failed Call History Call Info No info available Call is on hold This call is SECURE and peer is VERIFIED! Do you want to unverify the peer\? This call is NOT secure! Call Request Do you accept request to call \'%1$s\'\? Call Transfer Callee Do you want to add \'%1$s\' to contacts or delete %2$s from call history\? call Do you want to call or send message to \'%1$s\'\? calls Do you want to delete \'%1$s\' %2$s from call history\? Duration Cancel Chat Messages Chat with %1$s Chat History Choose destination URI Reorder Codecs You need to restart ritosip in order to activate the new settings. Restart now\? Settings Confirmation Consent Request Contact Do you want to call or send message to \'%1$s\'\? Contact \'%1$s\' already exists. Do you want to delete contact \'%1$s\'\? Name Contacts If Android contacts is chosen, they can be used in calling and messaging as references to SIP and tel URIs. ritosip app does not store Android contacts nor share them with anyone. In order to make Android contacts available in baresip, Google requires that you accept their use as described here and in app\'s Privacy Policy. Your maximum number of contacts %1$d has been exceeded. Chooses if ritosip contacts, Android contacts, or both are used. If both are used and a contact with the same name exists in both contacts, the ritosip contact will be chosen. 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 Dark Theme Force dark display theme Debug If checked, provides debug and info level log messages to Logcat. Decrypt Password Default Account If checked, this account is selected when ritosip is started. Default Call Volume If set, default call audio volume at scale 1–10. Default Phone App If checked, ritosip is the default phone app. Do not check if your device may need to handle also other than SIP calls or messages. Delete Do you want to delete account \'%1$s\'\? Delete Do you want to delete chat history of account \'%1$s\'\? Delete Do you want to delete call history of account \'%1$s\'\? Deny Dialer role is not available Dialpad Direction Disable Display Name Name (if any) used in From URI of outbound requests. Diverted by … DNS Servers Comma separated list of addresses of DNS servers. If not given, DNS server addresses are obtained dynamically from the system. Each DNS address is of form \'ip:port\' or \'ip\'. If port is omitted, it defaults to 53. If ip is an IPv6 address and also port is given, ip must be written inside brackets []. As an example, list \'8.8.8.8:53,[2001:4860:4860::8888]:53\' points to IPv4 and IPv6 addresses of public Google DNS servers. … DTMF In-band RTP or SIP INFO In-band RTP Events SIP INFO Requests DTMF Mode Selects how DTMF tones 0–9, #, *, and A-D are sent. Duration: %1$d (secs) Edit Enable Encrypt Password Error Failed to load module. Failed to set DNS servers Favorite Hangup Help Call Hold/Unhold Incoming call from Call from … Info Invalid Account Nickname \'%1$s\' Invalid user@domain[:port][;transport=udp|tcp|tls] \'%1$s\' Invalid Audio Delay \'%1$s\'. Valid values are from 100 to 3000. Invalid Authentication Password \'%1$s\' Invalid Authentication Username \'%1$s\' Invalid chat peer URI Invalid contact name \'%1$s\' Invalid SIP URI Invalid Country Code \'%1$s\' Invalid Display Name \'%1$s\' Invalid DNS Servers Invalid Frames Per Second \'%1$d\' Invalid Listen Address Invalid Microphone Gain value Invalid Opus bitrate Invalid Opus Packet Loss Percentage Invalid Proxy Server URI \'%1$s\' Invalid Registration Interval\'%1$s\' Invalid SIP or tel URI \'%1$s\' Invalid SIP URI \'%1$s\' Invalid SIP URI host part \'%1$s\' Invalid Password \'%1$s\' Invalid STUN/TURN Server URI \'%1$s\' Invalid Username \'%1$s\' Invalid User-Agent header field value Invalid Voicemail URI \'%1$s\' Jitter: %1$s (ms) Listen Listen Address IP address and port of form \'address:port\' at which ritosip listens for incoming SIP requests. If IP address is an IPv6 address, it must be written inside brackets []. IPv4 address 0.0.0.0 or IPv6 address [::] makes ritosip listen at all available addresses. If left empty (factory default), ritosip listens at port 5060 of all available addresses. Logcat Do you want to delete chat with peer \'%1$s\' or add peer to contacts\? Do you want to delete message or add peer \'%1$s\' to contacts\? Lost Manual Media Encryption Selects media transport encryption protocol (if any). \n • ZRTP (recommended) means that ZRTP end-to-end media encryption negotiation is tried after the call has been established. \n • DTLS-SRTPF means that UDP/TLS/RTP/SAVPF is offered in outgoing call and that RTP/SAVP, RTP/SAVPF, UDP/TLS/RTP/SAVP, or UDP/TLS/RTP/SAVPF is used if offered in incoming call. \n • SRTP-MANDF means that RTP/SAVPF is offered in outgoing call and required in incoming call. \n • SRTP-MAND means that RTP/SAVP is offered in outgoing call and required in incoming call. \n • SRTP means that RTP/AVP is offered in outgoing call and that RTP/SAVP or RTP/SAVPF is used if offered in incoming call. Media NAT Traversal Selects media NAT traversal protocol (if any). Possible choices are STUN (Session Traversal Utilities for NAT, RFC 5389) and ICE (Interactive Connectivity Establishment, RFC 5245). Failed Message from Messages Microphone On/Off Microphone Gain Multiply microphone volume by this decimal number. Minimum value is 1.0 (factory default) that disables microphone gain. Larger values may negatively affect audio quality. Missed call from Missed calls %1$d missed calls New Account New Chat Peer New Contact New message new messages Nickname No You are not able to access Android contacts without \"Contacts\" permission. You are not able create backup without \"Storage\" permission. ritosip is not able to detect Bluetooth connectivity without \"Nearby devices\" permission. ritosip needs \"Microphone\" permission for voice calls. You don\'t have any supported video cameras. You have no messages No network connection! You are not able to use this application without \"Notifications\" permission. You are not able restore backup without \"Storage\" permission. Account \'%1$s\' has no Telephony Provider Grant \"Camera\" permission to make or answer video calls. Nickname \'%1$s\' already exists Notice OK old messages one new message one old message Opus Bit Rate Average maximum bit rate used by Opus audio stream. Valid values are 6000-510000. Factory default is 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). 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 Call to … Packets Password Peer This call is SECURE, but peer is NOT verified! Permissions rationale Prefer IPv6 Media If checked, offer to use IPv6 media protocol (if available) when media protocol of peer cannot be automatically determined. Quit Current Rate: %1$s (Kbits/s) Failed to read file \'ca_certs.crt\'. Failed to read file \'cert.pem\'. Recording can be turned on or off only when call is not connected Redirect Mode Selects if call redirect request is followed automatically or if confirmation is requested. Automatic redirection to \'%1$s\'\ Redirect Request Do you accept call redirection to \'%1$s\'\? Registration Interval Tells how often (in seconds) ritosip sends REGISTER requests. Valid values are from 60 to 3600. Register If checked, registration is enabled and REGISTER requests are sent at the interval specified by Registration Interval. Registering of \`%1$s\` failed. Reject Reliable Provisional Responses If checked, indicate support for reliable provisional responses (RFC 3262). Reset Reset to Factory Defaults Are you sure you want to reset settings to factory default values\? If checked, settings are reset to factory default values. Restart Restart Request Restore Failed to restore application data. Check that you gave correct password and that the backup file is from this application. In Android versions 9, also check Apps → ritosip → 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$s. Application data restored. ritosip needs to be restarted. Restart now\? RTCP Multiplexing If checked, RTP and RTCP packets are multiplexed on a single port (RFC 5761). Send Send Message Sending of message failed Do you want to delete chat with \'%1$s\'\? Do you want to delete message\? Show Password SIP or tel URI SIP Trace If checked and if Debug is checked, Logcat messages include also SIP request and response trace. Unchecked automatically at ritosip start. SIP URI SIP URI of another Proxy Server SIP URI of Proxy Server Speaker Phone If checked, speaker phone is turned automatically on when call starts. Start Automatically If checked, ritosip starts automatically after device (re)start. Ritosip 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. Status STUN/TURN Password Password if required by STUN/TURN server STUN/TURN Server stun:stun.l.google.com:19302 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 STUN/TURN Username Username if required by STUN/TURN server 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 Time 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 version 9, a file called \'ca_certs.crt\' is loaded from Download folder. TLS Certificate File If checked, a file containing TLS certificate and private key of this ritosip instance has been or will be loaded. In Android version 9, a file called \'cert.pem\' is loaded from Download folder. For security reasons, delete the file after loading. Today Tone Country Country of call ringing, waiting, and callee busy tones Transfer Transfer destination Transfer failed Transfer Request Do you accept to transfer this call to \'%1$s\'\? Call transfer request to Transferring call to … Unknown Unverify User Agent Custom SIP request/response User-Agent header field value user@domain user@domain or telephone number User ID Verify Request Do you want to verify SAS <%1$s>\? Verify Server Certificates If checked, ritosip verifies TLS certificates of SIP User Agent and SIP Proxy Servers when TLS transport is used. Video call Video Codecs List of video codecs in priority order. Drag to reorder, swipe right to enable or disable. Video Frames Per Second Video frame rate that will be offered during the SDP handshake. Valid values are from 10 to 30. Video Request Video Frame Size Size of transmitted video frames (width x height) Voicemail Voicemail Messages Voicemail URI SIP URI for checking of voicemail messages. If left empty, voicemail messages (Message Waiting Indications) are not subscribed to. Yes You You have Your Name