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authorAlfred E. Heggestad <alfred.heggestad@gmail.com>2017-06-23 21:05:02 +0200
committerAlfred E. Heggestad <alfred.heggestad@gmail.com>2017-06-23 21:05:02 +0200
commit5fc916c6648a5895e3631d4f4b4f3cb31b50394b (patch)
tree42e2a386d046eff850a26093fd1ee0f5f554dd1d /docs/ChangeLog
parentac510eac6d39c7d3244d6c8b08bc5fb22a147b6b (diff)
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+2017-06-24 Alfred E. Heggestad <alfred.heggestad@gmail.com>
+
+ * Version 0.5.4
+
+ * GIT URL: https://github.com/alfredh/baresip.git
+ * GIT tag: v0.5.4
+ * NOTE: Requires libre v0.5.4 or later
+ Requires librem v0.5.0 or later
+
+ * config:
+ - audio_level yes|no Enable audio level RTP extension
+
+ * baresip-core:
+ - add support for Client-to-Mixer Audio Level Indication (RFC 6464)
+ - add support for RTP Header Extensions (RFC 5285)
+ - module: dont load same static module twice
+ - ua: add ua_progress()
+ - ua: check for Accept header in incoming OPTIONS request
+ - use a dummy RTP port for incoming OPTIONS (ref #265)
+ - vidcodec: make the API re-entrant
+ - vidfilt: make the API re-entrant
+ - vidisp: make the API re-entrant
+ - vidsrc: make the API re-entrant
+
+ * selftest:
+ - add test for audio level indication in call
+ - add test for call progress
+
+ * Modules:
+
+ * (all video modules updated with API-changes)
+
+ * zrtp: check for RTP packet in send handler (ref #262)
+ (thanks to MobiSciLab for reporting the bug)
+
+ - registered zrtp_log function with zrtp engine
+ - improved info message on how to verify remote peer
+ - improved setting and printing of zrtp cache file
+ (thanks Juha Heinanen)
+
+
2017-05-14 Alfred E. Heggestad <alfred.heggestad@gmail.com>
* Version 0.5.3