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* make struct vidsrc and vidisp publicAlfred E. Heggestad2018-04-02
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* timer: make tmr_jiffies_usec() publicAlfred E. Heggestad2018-03-21
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* video: change video_fps from int to double floatAlfred E. Heggestad2018-02-25
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* timestamp: new file for timestamp helpersAlfred E. Heggestad2018-02-14
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* stream: print a message when incoming RTP stream is establishedAlfred E. Heggestad2018-02-11
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* metric: fix calculation of average bitrateAlfred E. Heggestad2018-01-31
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* timer: add tmr_jiffies_usecAlfred E. Heggestad2018-01-05
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* vidutil: new file for video utility functionsAlfred E. Heggestad2017-12-30
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* ua: add UA_EVENT_CALL_RTCP eventAlfred E. Heggestad2017-12-09
| | | | | | this event is called for every received RTCP report. mqtt: add JSON encoding of RTCP stats
* audio: track timestamps for incoming RTP packetsAlfred E. Heggestad2017-11-09
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* ua: remove accessor to SIP file descriptor (fd)Alfred E. Heggestad2017-09-11
| | | | | | | - The fd is platform specific, and should not be exposed in the public API. (this function was used a long time ago for iOS applications)
* core: dont open RTP ports when receiving OPTIONSAlfred E. Heggestad2017-06-11
| | | | | | | - also added struct stream_param which contains common parameters for the media stream, passed down from call.c to stream.c ref #265
* add RTP Header extension for Client-to-Mixer Audio Level Indication (#264)Alfred E. Heggestad2017-06-02
| | | | | | | | | | | | | | | | * add RTP Header extension for Client-to-Mixer Audio Level Indication https://tools.ietf.org/html/rfc6464 requires libre from this commit or later: https://github.com/creytiv/re/commit/1544a1e375c76a80084b411d21b0431f95e9cdfb * fix warnings * fix warnings * minor cleanup
* rtpext: add RTP Header Extensions (RFC 5285) (#257)Alfred E. Heggestad2017-05-23
| | | | | | * rtpext: add RTP Header Extensions (RFC 5285) * whitespace fix
* audio: add offerer flag to audio_allocAlfred E. Heggestad2017-04-29
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* account: add tests and accessorsAlfred E. Heggestad2017-01-21
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* mnat: make it re-entrantAlfred E. Heggestad2016-12-26
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* video: start video on updateAlfred E. Heggestad2016-12-07
| | | | | | | - if we receive an updated SDP from re-INVITE which has the video line enabled, start the video stream Thanks to Gary Metalle for the original patch
* use PATH_MAX if definedAlfred E. Heggestad2016-11-02
| | | | | | ref: https://github.com/alfredh/baresip/pull/165
* core: add FS_PATH_MAXAlfred E. Heggestad2016-10-30
| | | | ref: https://github.com/alfredh/baresip/pull/143
* remove ua_prm(), use ua_account() insteadAlfred E. Heggestad2016-09-04
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* call: move local-address up to ua.cAlfred E. Heggestad2016-08-14
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* add '@' command to switch calls using line numbersAlfred E. Heggestad2016-07-26
| | | | | | | | | | | 1. for a given UA, each call has a unique line number starting from 1 2. the list of calls is sorted in an arbitrary order, but the last list element indicates "current call" 3. the '@' command takes a numeric argument which is the line-number of the wanted call
* update doxygen commentsAlfred E. Heggestad2016-07-19
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* add support for rtp_timeout and redialAlfred E. Heggestad2016-07-19
| | | | | | | | | | | | | | | | 1. Added support for RTP timeout. The feature is disabled by default and can be enabled with config "rtp_timeout N" where N is the number of seconds of RTP inactivity. If this is detected, the call is closed with a "special" SIP reason code of 701. 2. Added support for automatic re-connect in the menu module. This can be enabled by setting the 2 config items: redial_attempts 3 redial_delay 5 This work was contributed by Sveriges Radio. Thanks goes to Ola Palm and Jim Eld.
* net: make networking code re-entrantAlfred E. Heggestad2016-06-06
| | | | | | | | | | | - The network instance is now in struct network and does not use any local/static data - A new top-level struct in baresip.c owns the single instance of struct network it is a long-term goal to remove all local/static data from libbaresip and make it fully re-entrant.
* make call_is_onhold() publicAlfred E. Heggestad2016-02-27
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* move playing of ringtones to the menu-moduleAlfred E. Heggestad2016-01-09
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* ua: add support for hold+answerAlfred E. Heggestad2015-10-11
| | | | | | | - added ua_hold_answer() which will first put on-hold the active call (if exist) and then answer the new incoming call fixes #50
* make sip_req_send() publicAlfred E. Heggestad2015-09-26
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* change ausrc/auplay base-class to pointerAlfred E. Heggestad2015-06-16
| | | | | | | - change from a memory-reference to a const pointer - this saves one mem_deref() in the destructor and also decreases the chance of introducing memory leaks - test on debian
* add video error handlerAlfred E. Heggestad2015-03-21
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* presence/options patch from JuhaAlfred E. Heggestad2014-11-01
| | | | | | | | | | | | | most of this work was done by Juha Heinanen, and reviewed by me. thanks! - menu: add support for sending SIP OPTIONS requests to any peer, using the 'o' command - presence: add support for PUBLISH, set ;pubint=N to activate - ua: add Allow/Supported header to OPTIONS reply add MESSAGE to allowed methods
* metric: add metric_init()Alfred E. Heggestad2014-09-21
| | | | | - metric_init calls tmr_start(). tmr_start must be called from the re_main() thread.
* rtpstat: fixup some formattingAlfred E. Heggestad2014-05-20
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* Merge remote-tracking branch 'upstream/master'Lorenzo Mangani2014-05-18
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| * move struct stream to core.hAlfred E. Heggestad2014-05-18
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* | Basic support for X-RTP-Stat reports in BYE/200 OKLorenzo Mangani2014-05-18
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* re api changes:Alfred E. Heggestad2014-02-22
| | | | | - sip_param_decode() renamed to msg_param_decode() - use msg_ctype_cmp() to check for Content-Type
* baresip 0.4.10Alfred E. Heggestad2014-02-09