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

Juha Heinanen <jh@tutpro.com>

Version %1$s

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Known Issues

Source code

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

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

Juha Heinanen <jh@tutpro.com>

Version %1$s

Usage Hints

Known Issues

Source code

Source code is available at GitHub, where also issues can be reported. ]]> GitHub Account 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 12 minute intervals. Audio Codecs List of supported audio codecs in priority order Video Codecs List of supported video codecs in priority order 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], where scheme is \'stun\' or \'turn\'. 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$s\' Media Encryption Selects media transport encryption protocol (if any). \n • ZRTP (recommended) means that ZRTP end-to-end media encryption negotiation is tried after the call has been established. \n • DTLS-SRTPF means that UDP/TLS/RTP/SAVPF is offered in outgoing call and that RTP/SAVP, RTP/SAVPF, UDP/TLS/RTP/SAVP, or UDP/TLS/RTP/SAVPF is used if offered in incoming call. \n • SRTP-MANDF means that RTP/SAVPF is offered in outgoing call and required in incoming call. \n • SRTP-MAND means that RTP/SAVP is offered in outgoing call and required in incoming call. \n • SRTP means that RTP/AVP is offered in outgoing call and that RTP/SAVP or RTP/SAVPF is used if offered in incoming call. Prefer IPv6 Media If checked, offer to use IPv6 media protocol (if available) when media protocol of peer cannot be automatically determined. Answer Mode Selects how incoming calls are answered. Manual Automatic Voicemail URI SIP URI for checking of voicemail messages. If left empty, voicemail messages (Message Waiting Indications) are not subscribed to. Invalid Voicemail URI \'%1$s\' Default Account If checked, this account is selected when baresip is started. Accounts New Account Account\'s port number and transport protocol may be optionally given when a new account is created: username@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$s\'? Answer Reject Call from Call transfer request to Message from Tap to start application Call History Call calls call Do you want to call or send message to \'%1$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$s\' to contacts? Do you want to delete message? Add Contact Sending of message failed Failed Chat History Today You New Chat Peer Invalid SIP URI Do you want to delete chat with peer \'%1$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\'? Settings Start Automatically If checked, baresip starts automatically after device (re)start. 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 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, file \'cert.pem\' containing TLS certificate and private key of this baresip instance has been or will be loaded from Download directory. For security reasons, delete the file after loading. TLS CA File If checked, file \'ca_certs.crt\' containing TLS certificates of Certificate Authorities has been or will be loaded from Download directory. Audio 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 Extented 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. By default 0, turning off Opus Forward Error Correction (FEC). 0 Invalid Opus bitrate Invalid Opus Packet Loss Percentage Default Call Volume If set, default call audio volume at scale 1–10. Video Frame Size Size of transmitted video frames (width x height). Factory default is 800x600. Debug Provides debug and info level log message availability in Logcat. Reset to Factory Defaults If checked, settings are reset to factory default values. Failed to read file \'cert.pem\' from Download directory. Failed to read file \'ca_certs.crt\' from Download directory. You need to restart baresip in order to activate the new settings. Restart now? Contact New Contact Name Invalid contact name \'%1$s\' Contact \'%1$s\' already exists. Invalid SIP URI 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 … Backup Restore About Restart Quit Outgoing call to … Incoming call from … Transferring call to … Invalid SIP URI \'%1$s\' Callee Hangup Video call Video Request Accept video call with \'%1$s\'? Hold Call Transfer Transfer destination Transfer Transfer Failed DTMF Call Info No info available Duration Codecs Rate 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> <%2$s>? Transfer Request Do you accept to transfer this call to \'%1$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 backed up to Download folder file \'%1$s\'. Failed to back up application data to Download folder file \'%1$s\'. Check Apps → baresip → Permissions → Storage. Restart Request Application data restored. baresip needs to be restarted. Restart now? Failed to restore application data from Download folder. Check Apps → baresip → Permissions → Storage and that backup file \'%1$s\' exists in the folder and, if so, you gave correct Decrypt Password. You are not able to place or answer calls without Microphone permission. Grant \"Camera\" permission to place or answer video calls. You don\'t have any supported video cameras. Show Password