[Pkg-voip-commits] [janus] annotated tag v0.2.1 created (now aa82600)
Jonas Smedegaard
dr at jones.dk
Tue Mar 14 10:41:57 UTC 2017
This is an automated email from the git hooks/post-receive script.
js pushed a change to annotated tag v0.2.1
in repository janus.
at aa82600 (tag)
tagging e2a8e496ffc8ceb3057b49c93e037e85bcff8d54 (commit)
replaces v0.2.0
tagged by Lorenzo Miniero
on Tue Dec 13 11:32:19 2016 +0100
- Log -----------------------------------------------------------------
v0.2.1
Akagi201 (6):
fix compile failed on Mac
add compile on macOS to README
update README
update README
update README
support for old macOS
Alexander Clark (1):
support for setting an iceTransportPolicy
Eduardo Barbosa (1):
Added supervisor sample to the documentation
Emmanuel Riou (2):
fix MACOS endianness issue (due to lack of standart environment variables) + make janus-pp-rec compile on MACOS
Merge remote-tracking branch 'upstream/master' into pullreq
Fabrizio Bertone (1):
Update README.md
Lorenzo Miniero (87):
New SDP utilities to replace Sofia SIP SDP stack
Made Sofia SIP a dependency for only the SIP plugin, cleaned up configure.ac and Makefile.am, added enumeration for media direction, and used new SDP utils in VideoRoom plugin too
Return reason for SDP parsing errors
Merge branch 'master' into sdp-home
Added helper method to remove payload types from SDP
Helper method to free an SDP attribute
Support session level connection data
Converted SIP plugin to use the new SDP utils
First take at supporting re-invites/updates in SIP plugin with new SDP utils
Aligned with master (fixed conflicts)
Removed unneeded sdp_parser property
Revert "First take at supporting re-invites/updates in SIP plugin with new SDP utils"
First take at supporting re-invites/updates in SIP plugin (uses #578)
Set pointers to NULL after a g_list_free
Merge branch 'sdp-home' into sip-updates
Increased size of pollfd array to account for pipe file descriptor
Aligned with new v0.2.0
Merge branch 'sdp-home' into sip-updates
Fixed merge introduced error
Fixed memory leak
Merge branch 'sdp-home' into sip-updates
Merge branch 'master' into sdp-home
Merge branch 'sdp-home' into sip-updates
Merge pull request #618 from saghul/sdp_fixes
Merge branch 'sdp-home' into sip-updates
Removed unneeded pragma
Removed unneeded checks before g_free
Larger buffer when parsing crypto
Added JANUS_SDP_DEFAULT (=JANUS_SDP_SENDRECV)
Don't write direction attribute if it's JANUS_SDP_DEFAULT
Added fmts list, and fixed datachannels negotiation
Merge branch 'sdp-home' into sip-updates
Merge branch 'master' into sdp-home
Merge branch 'sdp-home' into sip-updates
Bump version number
Merge branch 'master' into sdp-home
Merge branch 'sdp-home' into sip-updates
Merge pull request #589 from meetecho/sip-updates
Merge pull request #578 from meetecho/sdp-home
Fixed small typos in documentation
Fixed small typos in documentation
Fixes to get Janus working with Edge again (see #651)
Don't drop video support on Edge, leave it to the application
Fixed a few leaks
Merge pull request #662 from uxmaster/master
Changed the verbosity of some log messages in WebSockets plugin
Merge pull request #670 from seb3s/seb3s-patch-1
Updated instructions for building libsrtp (1.5.4)
Make sure there's always an event to return in HTTP long poll
Fix crash in SIP plugin when no remote IP is found for RTP in the SDP
Fixed checks for adapter.js (were broken in Chrome 56)
Fix check for auth token support in admin.html/js
Added setting to modify own volume (percent) in audiobridge (see #668)
Fix SSRCs in RTCP before encrypting and not after, in SIP plugin
Updated resources list
New debugging level in janus.js (vdebug), a few changes in JS logging, and new slowLink event handler (example in echotest.js), plus updates to documentation
Use free instead of g_free for strings allocated by json_dumps (fixes #679)
Merge pull request #690 from fbertone/patch-1
Merge pull request #678 from eduardomb/master
Removed extra unlock (see #694)
Mention libcurl as optional dependency in the documentation (see #691)
Add optional authentication support to RTSP streaming (see issue #692)
Merge pull request #666 from akfork/master
Edited README guidelines for MacOS (see #666)
Fixed error when compiling Streaming plugin without libcurl
Made configure smarter (see issue #689)
Merge pull request #695 from agclark81/master
Lock forwarder mutex before using forwarder hash table (pull #686)
Added support for (some) RTP extensions
Added playout-delay to the RTP extensions
Automatically try using SIP INFO for DTMF in SIP demo when not on Chrome
Fixed typo
Implemented timeout/GET_PARAMETER support for RTSP in Streaming plugin
Make negotiation of audio-level RTP ext in AudioBridge configurable
Make negotiation of new RTP extensions in VideoRoom configurable
Merge pull request #697 from meetecho/extmap
Fixed --disable-unix-sockets check in configure.ac (fixes #701)
Autodetect libsrtp version (1.5.x vs 2.0.x)
Updated code to reflect API changes in case libsrtp2 is detected
Shimmed libsrtp2 API
Merge pull request #706 from Sean-Der/videoroom-listparticipants-ssrc
Merge pull request #707 from dazzl-tv/pullreq
Fixed async/sync AJAX request for detach/destroy (fixes #704)
Reduced verbosity of a couple of transport related messages
Modified RTCP code to recognize XR packets
Merge pull request #702 from meetecho/libsrtp2
Added failIfNoAudio/failIfNoVideo capture-related flags to janus.js, both default to false (fixes #705)
Saúl Ibarra Corretgé (4):
Style
Fix compilation
Fix crash if attribute value is empty
sip: reply with 488 if offer doesn't contain audio or video
Sean DuBois (1):
Include publisher's internal_audio_ssrc and internal_video_ssrc in plugin_videoroom listparticipants
Sébastien Saint-Sevin (1):
fix documentation
uxmaster (2):
typo fix
typo fix
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