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* video: use double float for estimated framerateAlfred E. Heggestad2018-02-27
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* call: fix memory leak in case sipsess_connect() failsAlfred E. Heggestad2018-02-11
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* call: save session callid on the structAlfred E. Heggestad2018-01-27
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* call: print error-string to reasonAlfred E. Heggestad2018-01-24
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* call: add call_id accessorAlfred E. Heggestad2018-01-17
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* 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
* Vidisp api reentrant (#258)Alfred E. Heggestad2017-05-26
| | | | | | * vidisp: make the API re-entrant * vidisp: update all modules to re-entrant API
* Vidsrc api reentrant (#256)Alfred E. Heggestad2017-05-23
| | | | | | * vidsrc: make the API re-entrant * vidsrc: update all modules to new API
* Revert "modules/zrtp: aligned code with latest libzrtp from Freeswitch:"Juha Heinanen2017-05-08
| | | | This reverts commit d513794c2ea8f746277853638ab2e97be6ce7f95.
* modules/zrtp: aligned code with latest libzrtp from Freeswitch:Juha Heinanen2017-05-08
| | | | | | | | | - zrtp_cache_set_verified is now zrtp_verified_set - local zid is not anymore included in zrtp_config and needs to be given as argument to zrtp_verified_set - zrtp_config.cache_file_cfg is now zrtp_config.def_cache_path call.c: update info message on received request (re-INVITE or INFO)
* attempt to fix SRTP for early-media (ref #229) (#243)Alfred E. Heggestad2017-05-02
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* audio: add offerer flag to audio_allocAlfred E. Heggestad2017-04-29
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* Call event handler for call progress/ringing after media has been set up (#240)Jan Hoffmann2017-04-27
| | | | The handler of the progress event may want to access the audio object of the call (for example to change the audio device). This is only possible if the media stream has already been started.
* additional debug (ref #225)Alfred E. Heggestad2017-03-25
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* call: print warning if sdp_decode failsAlfred E. Heggestad2017-03-25
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* update doxygen commentsAlfred E. Heggestad2016-12-10
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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
* test: added testcase for call with videoAlfred E. Heggestad2016-12-07
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* call: log when medianat was establishedAlfred E. Heggestad2016-11-27
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* remove ua_prm(), use ua_account() insteadAlfred E. Heggestad2016-09-04
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* call: more detailed warning messagesAlfred E. Heggestad2016-09-03
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* call: move local-address up to ua.cAlfred E. Heggestad2016-08-14
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* use KEYCODE_REL instead of 0x00Alfred E. Heggestad2016-07-31
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* define KEYCODE_REL for key releaseAlfred E. Heggestad2016-07-31
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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
* call: add sequential line-numbers for multiple calls (ref #141)Alfred E. Heggestad2016-07-23
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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.
* config: add "call_local_timeout" config optionAlfred E. Heggestad2016-06-05
| | | | | | | the config option is used for incoming calls, if the call is not answered after X seconds. The default value is 120 seconds. If the value is set to 0 the timeout timer is not enabled.
* call: move CALL_EVENT_ESTABLISHED to the endAlfred E. Heggestad2016-02-14
| | | | | | | | | - since the callback handler might destroy the call object, it is safer to call the handler as the last thing in the function. otherwise the call object might be corrupt, i.e. pointing to deallocated memory.
* menu: configurable bell and set incoming uaAlfred E. Heggestad2016-02-06
| | | | | | | | - add config "menu_bool yes|no" to control if the incoming call should print Bell escape characters to the terminal - on incoming call, set the current user-agent to the one with the incoming call
* call: fix for decoding SIP INFO with dtmf-relayAlfred E. Heggestad2016-02-02
| | | | patch from Gary Metalle, thank you :)
* call: add src/call.c path in doxygen headerAlfred E. Heggestad2015-11-28
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* call: check address-family of incoming SDP offerAlfred E. Heggestad2015-10-29
| | | | this fixes #79
* call: add direction flagAlfred E. Heggestad2015-10-25
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* call: fix warning if USE_VIDEO is not setAlfred E. Heggestad2015-10-25
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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
* Add transfer failed call eventCharles Lehner2015-07-05
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* add video error handlerAlfred E. Heggestad2015-03-21
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* 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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| * call: cancel local timer on call_progress()Alfred E. Heggestad2014-05-18
| | | | | | | | | | | | | | | | for incoming calls, we start a local timer of 120 seconds. this timer is stopped when the call is answered with 200 or 183. issue was reported by Victor Sergienko, thanks!
* | Basic support for X-RTP-Stat reports in BYE/200 OKLorenzo Mangani2014-05-18
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* call: check common audio codecs for incoming callAlfred E. Heggestad2014-04-21
| | | | | | | | for incoming calls, check that we have at least 1 common audio codec. if there are no common audio codecs, reject the call with 488 status code. Suggested and Tested by Juha Heinanen -- thanks!
* Merge branch 'master' into nextAlfred E. Heggestad2014-04-05
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| * added call_setup_duration() and some small thingsAlfred E. Heggestad2014-03-27
| | | | | | | | - Thanks to Lorenzo Mangani for this work
| * added dtmfio moduleAaron Herting2014-02-24
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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
* use new logging functionsAlfred E. Heggestad2014-02-09
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* baresip 0.4.10Alfred E. Heggestad2014-02-09
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