baresip
baresip+
About baresip
About baresip+
Baresip library based SIP User Agent
Juha Heinanen <jh@tutpro.com>
Version %1$s
Usage Hints
- In Android\'s \`Apps & notifications\` settings, allow baresip app\'s notifications and
disable battery optimization (if you want to reduce likelihood that Android
restricts baresip\'s access to network or enters baresip to standby state).
- Check that default values in baresip\'s Settings meet your needs
(touch item titles for help).
- Then in Accounts, create one or more accounts (again touch item titles for help).
- A new account can be partially configured automatically. See
Wiki
for more information.
- Registration status of an account is shown with a colored dot: green (registration
succeeded), yellow (registration is in progress), red (registration failed), white (registration
has not been activated).
- Touch on the dot leads directly to account configuration.
- Swipe down gesture causes re-registration of the currently shown account.
- Swipe left/right gesture toggles between the accounts.
- Icon at the top of the main screen toggles speakerphone.
- Icons at the bottom of the main screen lead to voicemail (if Voicemail URI has been
configured for the account), contacts, messages, and call history, and allow toggling
between numeric and alphanumeric keyboard.
- Previous call party can be reselect by touching the call icon when Callee is empty.
- Peers of calls and messages can be added to contacts by long touches.
- Long touches can also be used to remove calls, chats, messages, and contacts.
- Touch/long touch of contact icon can be used to install/remove image avatar.
Known Issues
- Due to limitations in underlying libraries, baresip does not currently support multiple,
concurrently active network interfaces. Active network interface preference order is VPN,
Internet, other.
Source code
Source code is available at GitHub,
where also issues can be reported.
Licenses
- BSD-3-Clause except the following:
- Apache 2.0 AMR codec and TLS security
- LGPL 2.1 G.722 and G.726 codecs
- AGPLv4 ZRTP media encryption
- GNU GPLv3 G.729 codec
]]>
Baresip library based SIP User Agent with video calls
Juha Heinanen <jh@tutpro.com>
Version %1$s
Usage Hints
- In Android\'s \`Apps & notifications\' settings, allow baresip+ app\'s notifications and
turn off battery optimization (if you want to reduce likelihood that Android kills
baresip+).
- Check that default values in baresip+\'s Settings meet your needs
(touch item titles for help).
- Then in Accounts, create one or more accounts (again touch item titles for help).
- A new account can be partially configured automatically. See
Wiki
for more information.
- Registration status of an account is shown with a colored dot: green (registration
succeeded), yellow (registration is in progress), red (registration failed), white (registration
has not been activated).
- Touch on the dot leads directly to account configuration.
- Swipe down gesture causes re-registration of the currently shown account.
- Swipe left/right gesture toggles between the accounts.
- Icon at the top of the main screen toggles speakerphone.
- Icons at the bottom of the main screen lead to voicemail (if Voicemail URI has been
configured for the account), contacts, messages, and call history, and allow toggling
between numeric and alphanumeric keyboard.
- Previous call party can be reselect by touching the call icon when Callee is empty.
- Peers of calls and messages can be added to contacts by long touches.
- Long touches can also be used to remove calls, chats, messages, and contacts.
- Touch/long touch of contact icon can be used to install/remove image avatar.
Known Issues
- Due to limitations in underlying libraries, multiple concurrently active network
interfaces are not supported. Active network interface preference order is VPN,
Internet, other.
- In video calls, the device needs to be held in landscape
mode rotated 90 degrees left from portrait orientation.
- Selfview is not properly shown when video stream is sendonly.
Source code
Source code is available at GitHub,
where also issues can be reported.
Licenses
- BSD-3-Clause except the following:
- Apache 2.0 AMR codec and TLS security
- LGPL 2.1 G.722 and G.726 codecs
- AGPLv4 ZRTP media encryption
- GNU GPLv2 H.264 and H.265 codecs
- GNU GPLv3 G.729 codec
- AOMedia AV1 codec
]]>
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][?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$s\'
Media Encryption
Selects media transport encryption protocol (if any).
\n • ZRTP (recommended) means that ZRTP end-to-end media encryption negotiation is tried after
the call has been established.
\n • DTLS-SRTPF means that UDP/TLS/RTP/SAVPF is offered in outgoing call and that RTP/SAVP,
RTP/SAVPF, UDP/TLS/RTP/SAVP, or UDP/TLS/RTP/SAVPF is used if offered in incoming call.
\n • SRTP-MANDF means that RTP/SAVPF is offered in outgoing call and required in incoming call.
\n • SRTP-MAND means that RTP/SAVP is offered in outgoing call and required in incoming call.
\n • SRTP means that RTP/AVP is offered in outgoing call and that RTP/SAVP or RTP/SAVPF is used
if offered in incoming call.
Prefer IPv6 Media
If checked, offer to use IPv6 media protocol (if available) when media protocol of peer cannot be automatically determined.
DTMF Mode
Selects how DTMF tones 0–9, #, *, and A-D are sent.
In-band RTP Events
SIP INFO Requests
Answer Mode
Selects how incoming calls are answered.
Manual
Automatic
Voicemail URI
SIP URI for checking of voicemail messages. If left empty, voicemail
messages (Message Waiting Indications) are not subscribed to.
Invalid Voicemail URI \'%1$s\'
Default Account
If checked, this account is selected when baresip is started.
Accounts
New Account
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$s\'?
Answer
Reject
Incoming call from
Missed call from
Call transfer request to
Message from
Tap to start application
Call History
Call
calls
call
Do you want to call or send message to \'%1$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, 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
Server Certificates cannot be verified without
TLS CA file.
If checked, baresip verifies TLS certificates of SIP User
Agent and SIP Proxy Servers when TLS transport is used. If checked, also TLS CA File has
to be loaded.
TLS CA File
If checked, a file containing TLS certificates
of needed Certificate Authorities has been or will be loaded. In Android versions 9 and
below, a file called \'ca_certs.crt\' is loaded from Download folder.
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 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. 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.
Dark Theme
Force dark display theme
Video Frame Size
Size of transmitted video frames (width x height).
Factory default is 800x600.
Debug
If checked, provides debug and info level log messages to Logcat.
SIP Trace
If checked and if Debug is checked, provides also SIP
request and response trace to Logcat. 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?
Contact
New Contact
Name
Invalid contact name \'%1$s\'
Contact \'%1$s\' already exists.
Invalid SIP URI
Android
If checked, this contact is available also in Android contacts.
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$d has been exceeded.
Alert
Info
Notice
Cancel
OK
Yes
No
Accept
Deny
User ID
Password
SIP URI
Add
Delete
Edit
Send
Status
Error
Help
Confirmation
…
\u2022 %1$s
Backup
Restore
About
Restart
Quit
Outgoing call to …
Incoming call from …
Transferring call to …
Invalid SIP URI \'%1$s\'
Callee
Hangup
Video call
Video Request
Accept sending and receiving video with \'%1$s\'?
Accept sending of video to \'%1$s\'?
Accept receiving video from \'%1$s\'?
Call Hold/Unhold
Microphone On/Off
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 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. In Android versions 9 and below, also check Apps → baresip → Permissions → Storage
and that file \'%1$s\' exists in Download folder.
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