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* srtp: split srtp-module into 'srtp' and 'libsrtp'Alfred E. Heggestad2014-06-18
| | | | | | | | | | | libre v0.4.9 has a very fast and robust SRTP-stack and we want to use it in baresip. however libsrtp is also working so we want to keep the module using libsrtp around for a while. two modules providing SDES-SRTP: - libsrtp.so -- SRTP using libsrtp - srtp.so -- Native SRTP-stack in libre
* dtls_srtp: use DTLS-api from libre v0.4.9Alfred E. Heggestad2014-06-18
| | | | | | | | for the dtls_srtp module to compile, you now need libre v0.4.9 or later. also added note about dependency to libre v0.4.9 in README and Debian file
* remove support for Symbian OSAlfred E. Heggestad2014-06-15
| | | | | R.I.P. Symbian it was nice to know you, too bad the people at Nokia did not take better care of you ..
* fix warningAlfred E. Heggestad2014-05-20
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* Merge branch 'lmangani-master'Alfred E. Heggestad2014-05-20
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| * rtpstat: fixup some formattingAlfred E. Heggestad2014-05-20
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| * Cleanup of XRTP stats print functionLorenzo Mangani2014-05-20
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| * Fixed JI value to timestamp-units based on codec audiorateLorenzo Mangani2014-05-19
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| * Minor cleanupLorenzo Mangani2014-05-18
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| * Move XRTP stats to audio and implement EN/DE parametersLorenzo Mangani2014-05-18
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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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* | 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!
* | fix call mute-state for multiple callsAlfred E. Heggestad2014-05-18
| | | | | | | | | | | | | | added audio_ismuted() and remove static state from menu.c thanks to Remik who reported the issue and suggested a nice solution :)
| * Fix return condition to avoid injecting stats when there are noneLorenzo Mangani2014-05-18
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| * Basic support for X-RTP-Stat reports in BYE/200 OKLorenzo Mangani2014-05-18
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* audio: fix timestamp calculation for sendingAlfred E. Heggestad2014-05-17
| | | | | | | | | | according to RFC 3551 the RTP timestamp is independent of number of channels and encoding. fix the timestamp calculation for outgoing RTP packets, by dividing the total number of samples on number of channels. the bug was not affecting calls with mono audio, but was very likely to affect calls with stereo audio (e.g. OPUS)
* rtp_stats: skip if no packets were sent/recv'dAlfred E. Heggestad2014-05-17
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* account: add sample with outbound transportAlfred E. Heggestad2014-05-17
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* dtls_srtp: clear openssl error queue on errorsAlfred E. Heggestad2014-05-17
| | | | | | | openssl has a nice global error queue, if an error occurs we must read out the error and then call ERR_clear_error(). otherwise other users of openssl in the same process will get our errors, and things will stop working.
* ice: set default candidates for 'lite' modeAlfred E. Heggestad2014-05-01
| | | | | | | | | | | when using ice-lite mode, the baresip ICE module is not setting RTP/RTCP connection line in SDP properly. this should be fixed by making a call to this function after the local candidates have been gathered: icem_lite_set_default_candidates(m->icem); the issue was reported by Juha Heinanen, thanks
* 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!
* modules: add USE_AVCAPTURE and try to detect itAlfred E. Heggestad2014-04-21
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* rst: use logging functionsAlfred E. Heggestad2014-04-20
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* dshow: use info() instead of re_printf()Alfred E. Heggestad2014-04-20
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* avcapture: use warning() instead of re_printf()Alfred E. Heggestad2014-04-20
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* rst: remove psize, use sampc insteadAlfred E. Heggestad2014-04-20
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* v4l2: remove trailing whitespaceAlfred E. Heggestad2014-04-14
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* whitespaceDmitrij D. Czarkoff2014-04-14
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* v4l2: add format negotiation and OpenBSD supportDmitrij D. Czarkoff2014-04-14
| | | | | | | | | | | Changes are: * libv4l2 is not required for proper operation any more (it is still desirable, as it guarantees that librem-supported format will be available); * v4l2 module now iterates through the list of available formats and settles on first one supported by librem; * if libv4l2 is present, baresip will try to use native format and fall back to emulated one only if no usable combination found.
* metric: fix average bitrate calculationAlfred E. Heggestad2014-04-12
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* modules: use new logging functionsAlfred E. Heggestad2014-04-12
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* added some missing commentsAlfred E. Heggestad2014-04-12
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* use logging functionsAlfred E. Heggestad2014-04-12
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* play: fix warnings when compiling with OPT_SIZE=1Alfred E. Heggestad2014-04-12
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* selftest: new module for testing baresip coreAlfred E. Heggestad2014-04-12
| | | | | | | | | | | | | | | the selftest module is using the public API of the Baresip core (in baresip.h) to verify the expected behaviour and to test various protocol scenarios. the module can be loaded with the 'module' directive or 'module_tmp' directive: module_tmp selftest.so OR: module selftest.so
* Merge pull request #6 from alfredh/ausrc_apiAlfred E. Heggestad2014-04-11
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| * Merge branch 'master' into ausrc_apiAlfred E. Heggestad2014-04-11
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* | dtmfio: include unistd for unlink()Alfred E. Heggestad2014-04-11
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* | debian: baresip now needs libre v0.4.8 or laterAlfred E. Heggestad2014-04-11
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| * sndio: update to new ausrc/auplay APIAlfred E. Heggestad2014-04-11
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| * Merge branch 'master' into ausrc_apiAlfred E. Heggestad2014-04-11
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* | Merge pull request #5 from alfredh/nextAlfred E. Heggestad2014-04-11
|\ \ | | | | | | Next
| * \ Merge branch 'master' into nextAlfred E. Heggestad2014-04-11
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* | | sndio: audio driver for OpenBSDAlfred E. Heggestad2014-04-09
| | | | | | | | | | | | It was contributed by Dmitrij D. Czarkoff, thank you!
| * | Merge branch 'master' into nextAlfred E. Heggestad2014-04-05
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* | | config: add opensles to templateAlfred E. Heggestad2014-03-30
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| * | stream_debug: also print local SDP addressAlfred E. Heggestad2014-02-22
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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
| | * Merge branch 'master' into ausrc_apiAlfred E. Heggestad2014-03-29
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