Usage Hints

Check that configuration default values meet your needs (click item titles for help).
Then create one or more accounts (again click item titles for help).
Peers of calls and messages can be added to contacts by long clicks.
Long clicks can also be used to remove calls, chats, messages, and contacts.
Call click when callee has not been given can be used for re-dial.
If you cannot hear the other party of a call, try to increase Media volume.
Audio echo cancellation is currently not supported when call uses Opus codec.

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. Authentication Username Authentication username if required by outbound proxy. Authentication Password Authentication password if required by outbound proxy. 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:foo.com:5060;transport=tls\n • sip:[2001:67c:223:777::10]:5060;transport=tcp SIP URI of Proxy Server SIP URI of another Proxy Server 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. 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 Server A STUN Server 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. stun.l.google.com:19302 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. Voicemail URI SIP URI for checking of voicemail messages. If left empty, voicemail messages (Message Waiting Indications) are not subscribed to. Default Account If checked, this account is selected when baresip is started. Accounts user@domain Invalid user@domain \'%1$s\' Account \'%1$s\' already exists. "Failed to allocate new account. Encrypt Password Decrypt Password Export Exported accounts to Download folder file \'accounts.bs\'. "Failed to export accounts to Download folder. Check Apps -> baresip -> Permissions -> Storage." Import Imported accounts from Download folder. You need to restart baresip. Failed to import accounts from Download folder file. Check Apps -> baresip -> Permissions -> Storage and that file \'contacts.bs\' exists in the folder and, if so, you gave correct Decrypt Password." Do you want to delete account \'%1$s\'? Answer Reject Call from Call transfer request to Message from Call History Call calls call Do you want to call or send message to \'%1$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? 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\'? Configuration 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. 0.0.0.0:5060 Prefer IPv6 Prefer IPv6 if both IPv4 and IPv6 are available. 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. Acoustic Echo Cancellation If checked, echo is tried to cancel during calls. Does not currently apply to calls using opus codec. Opus Bit Rate Average maximum bit rate used by Opus audio stream. Valid value are 6000-510000. Factory default is 28000. 28000 ICE Lite Mode If checked, ICE Lite Mode is used. Default Call Volume If set, default call audio volume at scale 1–10. Debug If checked, debug and info level log messages are available in logcat. Reset to Factory Default If checked, configuration is reset to factory default. Contact New Contact Name Export Import 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. Exported contacts to Download folder file \'contacts.bs\'. "Failed to export contacts to Download folder. Check Apps -> baresip -> Permissions -> Storage." Imported contacts from Download folder. Failed to import contacts from Download folder. Check Apps -> baresip -> Permissions -> Storage and that file \'contacts.bs\' exists in the folder. Alert Info Notice Cancel OK Yes No Accept Deny User ID Password SIP URI Add Delete Edit Send Status Error … About Quit Outgoing call to … Incoming call from … Transferring call to … Invalid SIP URI Callee Hangup Hold DTMF Call Info 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! Listen Address was reset. Restart baresip. Registering of \`%1$s\` failed. You have not granted microphone permission. Verify Do you want to verify SAS <%1$s> <%2$s>? Do you accept to transfer call to \'%1$s\'? Call failed Call closed You need to restart baresip in order to activate the new config! 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