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Added support for INFO and MESSAGE to the SIP plugin
Merge branch 'master' into offerless-invites
Fixed broken authentication when sending INVITE in SIP plugin
Make sure an 'accepted' event has the transaction ID of the 'accept' that originated it, when doing offerless INVITEs
Added support for offerless INVITEs to the SIP plugin
Fixed typos in SIP plugin docs
Better verbosity when trying to do a REGISTER in the SIP plugin
Use the authuser part when registering in the SIP plugin, if provided
Some REGISTER-related fixes in the SIP plugin, and improved SIP demo UI/options
Added the possibility to specify an outbound proxy in the SIP plugin
Send an event back when a DTMF has been sent via SIP INFO as requested
Allow custom headers in REGISTER too, in SIP plugin
Made username mandatory when registering, guest or not (fixes #885)
Support for on-hold in SIP plugin
Make sure RTCP buffers are reset before they're written to (fixes #833)
Moved IP self-detect of SIP plugin outside of the config parse code
Fixed broken re-INVITE management in SIP plugin
Integrated SDP utils in Record&Play plugin too
Further cleanup of ip-utils related code
Changed API of janus_network_detect_local_ip to better fit ip-utils, and added wrapper (integrated in janus.c and janus_sip.c) that returns an allocated string
Merge branch 'master' into iputils-usage
Make sure media is only updated after a re-INVITE
Moved most of SRTP-related stuff to rtp.h/.c (cleans dtls and janus_sip)
Better integration of new IP tools in Janus core and plugins
Made RTP context and rewriting part of the core, rather than plugins
Fixed #754, and added error message in case of missing/invalid IP
Reconnect sockets to new IP as well
If remote media info changes janus sip plugin will update only remote ports.h
Added support for libsrtp2 to SIP plugin too (fixes #709)
Merge branch 'master' into event-handlers
Fix SSRCs in RTCP before encrypting and not after, in SIP plugin
Fix crash in SIP plugin when no remote IP is found for RTP in the SDP
Aligned with new v0.2.1
Allow plugins to send out-of-context events (no associated session/handle) to event handlers
Merge branch 'sdp-home' into sip-updates
Larger buffer when parsing crypto
sip: reply with 488 if offer doesn't contain audio or video
Merge branch 'master' into sdp-home
Changing log setting on invite without SIP to LOG_WARN
Found a bug in janus_sip.c when the sip stack receives an INVITE without SDP after the inital invite. In this call flow, Janus was assuming the invite would always have an SDP and would segfault when receving an invite without one.
Added a quick patch to check if the received invite actually has an SDP before attepting to parse it, and if it doesn't, just respond 200 OK and let the call continue.
Fixed merge introduced error
Aligned with new v0.2.0
Fixed indentation
Increased size of pollfd array to account for pipe file descriptor
Made plugin response more concise (code suggested by @andreasg123)
Merge branch 'master' into plugins-json
Changed naming of threads, fixed wav header in audiobridge recording, anticipated sessions stuff in Janus startup (to avoid issues when some of the transport plugins drag and requests start arriving)
First take at supporting re-invites/updates in SIP plugin (uses #578)
Revert "First take at supporting re-invites/updates in SIP plugin with new SDP utils"
This reverts commit 87399acb42062e03d7f11ed4db096f8cd7c416d1.
Removed unneeded sdp_parser property
Aligned with master (fixed conflicts)
First take at supporting re-invites/updates in SIP plugin with new SDP utils
Converted SIP plugin to use the new SDP utils
Reject attempts to start SIP calls with datachannels (fixes #581)
Added plugin configuration for whether or not to shoot plugin-specific events (even when global configuration is yes)
Fixed typo introduced in #577
New SDP utilities to replace Sofia SIP SDP stack
Session-refresh handling (2) - free memory
Fixed a couple of potential leaks in SIP plugin
Rudimentary handling of SIP session-refresh (keepalive) added.
Fixes for 64-bit identifiers
Added 'autoack' parameter to 'call' in SIP plugin to drive NUTAG_AUTOACK
Added new approach to new TextRoom plugin and aligned to master
Added events to new TextRoom plugin and aligned to master
sip: style fixes
To match the configuration in editorconfig.
sip: add ability to override User Agent per account
Added outgoing SIP messages to events (to improve/fix)
Added incoming SIP messages to the events (still missing outgoing)
More events, in particular from other plugins than the EchoTest, and added examples to the sample handler plugin
Use json_true() and json_false() where we used 0/1 integers or true/false strings
New mutexes to protect recorders in plugins from race conditions (see #531 and #533)
Merge branch 'master' into recording-codecs
Merge pull request #533 from andreasg123/sip-close-recorder
Avoid race condition when closing recorders in SIP
Optimization of core-to-plugin communication
sip: add ability to customize the display name
In SIP, protect recorders with a mutex to avoid race conditions.
use JANUS_VALIDATE_JSON_OBJECT() and related helpers in all plugins
rolled back changes for early media
Early media for session progress
Early media for 183 session progress scenarios when SDP is availablebefore call accept.
Proceeding call state added
Added call state ‘proceeding’ from sofia nua_callstates. Since there’sno early media, this allows the client interface to play a ringbacktone for the user.
Merge pull request #467 from meetecho/rtcp-rr
First take at RTCP SR/RR in core
Reduced verbosity introduced in latest commit
Pass right codec information to the recorder in the SIP plugin
Support for other codecs and formats in recorder and post-processor