Component Translated Unfinished Unfinished words Unfinished characters Untranslated Checks Suggestions Comments
Glossary Baresip BSD-3-Clause 0% 1 4 23 1 0 0 0
baresip app BSD-3-Clause 3% 333 2,847 19,805 333 0 0 0

Overview

Project website github.com/juha-h/baresip-studio
Project maintainers User avatar juha-h
Language Korean
Language code ko
Text direction Left to right
Number of speakers 78,357,047
2 days ago

String statistics

Strings percent Hosted strings Words percent Hosted words Characters percent Hosted characters
Total 347 2,866 19,926
Translated 3% 13 1% 15 1% 98
Needs editing 0% 0 0% 0 0% 0
Read-only 3% 13 1% 15 1% 98
Failing checks 0% 0 0% 0 0% 0
Strings with suggestions 0% 0 0% 0 0% 0
Untranslated strings 96% 334 99% 2,851 99% 19,828
Last 12 months Previous

Activity in last 12 months

0 changes 0 changes 0 changes 0 changes 0 changes 0 changes 0 changes 0 changes 0 changes 336 changes 0 changes 7 changes 0 changes 9 changes 0 changes 6 changes 0 changes 0 changes 0 changes 0 changes 0 changes 4 changes 0 changes 12 changes
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Quick numbers

2,866
Hosted words
347
Hosted strings
3%
Translated
0
Contributors
and previous 30 days

Trends of last 30 days

+1%
Hosted words
+100%
+2%
Hosted strings
+100%
+1%
Translated
+3%
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Contributors
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Resource updated
The “app/src/main/res/values/strings.xml” file was changed. 2 days ago
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String added in the repository
Logcat
 
2 days ago
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Resource updated
The “app/src/main/res/values/strings.xml” file was changed. 9 days ago
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String updated in the repository
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 <a href="https://raw.githubusercontent.com/juha-h/baresip-studio/master/PrivacyPolicy.txt">Privacy Policy</a>.
 
9 days ago
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String added in the repository
1.0
 
9 days ago
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String added in the repository
Invalid Microphone Gain value
 
9 days ago
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String added in the repository
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.
 
9 days ago
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String added in the repository
Microphone Gain
 
9 days ago
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String updated in the repository
Selects media transport encryption protocol (if any).
• ZRTP (recommended) means that ZRTP end-to-end media encryption negotiation is tried after the call has been established.
• 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.
• SRTP-MANDF means that RTP/SAVPF is offered in outgoing call and required in incoming call.
• SRTP-MAND means that RTP/SAVP is offered in outgoing call and required in incoming call.
• SRTP means that RTP/AVP is offered in outgoing call and that RTP/SAVP or RTP/SAVPF is used if offered in incoming call.
 
9 days ago
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String updated in the repository
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 [].
Examples:
• sip:example.com:5061;transport=tls
• sip:[2001:67c:223:777::10];transport=tcp
• sip:192.168.43.50:443;transport=wss
 
9 days ago
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