b9ce5ddb4d
Changelog: * 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) |
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