Support for transport-wide CC on outgoing streams #1889

lminiero · 2019-12-05T17:52:39Z

This is an extension to the work started in #1884, where we did a bit of refactoring to give us more control on RTP extensions. That effort payed off here, as it allowed us to inject our own transport-wide RTP extension, which is the first step to implementing some form of bandwidth estimation in Janus.

More specifically, this patch implements the following changes:

It allows plugins to negotiate transport-wide CC when offering themselves, which was prevented before.
When negotiated, and when video is used, we add a global sequence number to the transport-wide CC extension (http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01). Before we only parsed the extension set by clients, in order to craft our RTCP feedback to send back: this now allows clients (e.g., browsers) to send us feedback via RTCP instead.
There is code in place to parse the transport-cc RTCP feedback sent by clients: we parse the whole message and build a list, which is currently ignored.

This whole effort will be the starting point to implement some actual BWE, which will happen in a separate PR. As it is, even ignoring the feedback we get back as we do now, it still improves some use cases that were partially broken before, e.g., sender-side BWE from both caller and callee in the VideoCall plugin (see #1473 and #1861). In fact, ensuring we negotiate the extension also when we offer ourselves, means browsers will add the extension in those sessions as well, and use the feedback we send back to adapt their bitrate: before, this was only possible when the client was offering (EchoTest, VideoCall caller, VideoRoom publisher, etc.)

Please make sure you test this and provide feedback, as it's an important step forward to make considerable improvements to the bandwitdh management in Janus. Since it's based on #1884, testing this branch you actually test both, and I do need feedback on both. Once I have a clearer idea on whether it works as expected and causes no issues, I'll be able to focus on the more complex stuff, like integrating some algorithm to do basic bandwidth estimation, and provide plugins (and applications) with more precise information, and possibly even automated behaviours.

…d them manually

tmatth · 2019-12-05T19:40:26Z

janus.c

@@ -3407,6 +3409,10 @@ json_t *janus_plugin_handle_sdp(janus_plugin_session *plugin_session, janus_plug
 		}
 		if(ice_handle->stream && ice_handle->stream->mid_ext_id != mid_ext_id)
 			ice_handle->stream->mid_ext_id = mid_ext_id;
+		if(ice_handle->stream && ice_handle->stream->transport_wide_cc_ext_id != transport_wide_cc_ext_id) {
+			ice_handle->stream->do_transport_wide_cc = transport_wide_cc_ext_id > 0 ? TRUE : FALSE;


nit: this could be simplified to transport_wide_cc_ext_id > 0;

oscarvadillog · 2019-12-06T17:03:33Z

Well done @lminiero! The issue with the videocall was solved.
Tested on Chrome 78 🎉 🚀

tmatth · 2019-12-10T16:06:54Z

@lminiero this just got reported and may be related to the various issues ppl have opened in janus around this topic:
https://bugs.chromium.org/p/webrtc/issues/detail?id=11196

lminiero · 2019-12-10T16:12:16Z

@tmatth the issue people reported was caused by the fact VideoCall callees can't use transport-cc in master, because there we only allow it when browsers are the offerers. Since callees are answerers, in master transport-cc is not used for them, and the browser can't do its BWE, probably resulting in those low bitrates. It was confirmed that with this PR the issue is fixed, and that's because this patch enables transport-cc no matter who offers.

Not sure if we're impacted by the issue you mentioned, as we ignore padding packets (but we take them into account for transport-cc), and we currently don't generate our own (we'll probably need to in the future, for probing).

zayim · 2019-12-18T08:27:08Z

Hi, Lorenzo! We tested this branch and it mainly works great, video quality is much better!

We did found 2 issues, however:

Video gets rotated. Happens in video calls involving mobile phones (tested on Android), video stream coming from front camera is rotated 90° CW (it happens right from the beginning, without rotating phone, etc.). Does not happen on master branch.
Sometimes, when making video calls from Chromium on Ubuntu (not Chrome) we get lots of:

[ice.c:janus_ice_outgoing_traffic_handle:4096] [6423777842486859] Error setting transport wide CC sequence number...

error logs. This happens when using SIP plugin (and when terminating to something that is not WebRTC, i.e. FreeSwitch).

lminiero · 2019-12-18T08:35:35Z

Hi, Lorenzo! We tested this branch and it mainly works great, video quality is much better!

Excellent, thanks for testing!

Video gets rotated. Happens in video calls involving mobile phones (tested on Android), video stream coming from front camera is rotated 90° CW (it happens right from the beginning, without rotating phone, etc.). Does not happen on master branch.

That may be because of something we're doing wrong handling the video orientation extension: we parse it, set it in the structure, and are then supposed to re-add it again, but that may not be happening for some reason.

Sometimes, when making video calls from Chromium on Ubuntu (not Chrome) we get lots of:

[ice.c:janus_ice_outgoing_traffic_handle:4096] [6423777842486859] Error setting transport wide CC sequence number...

error logs. This happens when using SIP plugin (and when terminating to something that is not WebRTC, i.e. FreeSwitch).

Not sure what's wrong there, browser shouldn't matter. I'll have a look.

lminiero · 2019-12-19T10:34:30Z

@zayim do both of the issues happen on the SIP plugin, or were you testing on another plugin? Is the video-orientation header negotiated?

lminiero · 2019-12-19T11:04:44Z

@zayim thinking about it, if this is happening with the SIP plugin, then it's expected. As explained in #1884, we strip all extension when we call relay_rtp(), and we only recreate them if they're set in the new structure. So, assuming A is using Janus and B is a regular SIP endpoint, this is what's happening:

A and B negotiate the video-orientation extension in their SIP call;
A adds extension to RTP packet;
Janus core parses extension, sets values in the struct, passes packet+struct to plugin;
SIP plugin relays packet (that still has extension, we don't remove it on the way in) via plain RTP to the peer;
B receives the extension, so everything's fine there;
B adds extension to its own RTP packet;
SIP plugin receives the packet, and relays it to the Janus core (extension still there);
Janus core strips all extensions, and adds those it is aware of (e.g., mid); since the SIP plugin didn't use the new struct to pass info on the video-orientarion extension, it is not re-added to the packet;
A receives RTP packet without extension.

Issue happens to B as well if they're behind Janus too, for the same reason.

Again, this is expected due to the changes in #1884, and the rationale and motivations are explained in detail there. It's up to plugins to fill the struct with info on extensions they want WebRTC users to receive, which means in this case the SIP plugin would need to be aware of extension IDs, parse packets on the way in to set the struct accordingly (a bit like the core does before calling incoming_rtp) and pass that metadata with the packet.

zayim · 2019-12-19T14:33:33Z

Hi, Lorenzo! Thanks for the reply. Yes, both of issues are on SIP plugin. I understand what is issue with this rotated video. Do you guys have any plan to implement parsing packets and adding extensions for SIP plugin some times soon?

lminiero · 2019-12-19T14:36:47Z

Do you guys have any plan to implement parsing packets and adding extensions for SIP plugin some times soon?

Not right away, no. We're going on our holiday break soon, so it's likely this will have to wait until we get back.

lejlasolak · 2019-12-27T15:10:59Z

Hi, we implemented adding video-orientation extension for SIP plugin and it seems to work. There is pull request for this.

jbboisseau · 2020-01-29T09:50:09Z

Hi @lminiero
This branch works great on our side. Do you know when you can merge it with the master?

lminiero · 2020-01-29T09:58:22Z

@jbboisseau thanks for the feedback! We'll merge soon but not sure when: I still need to merge a couple of things first, then we'll create a new tag and merge this and other more "breaking" PRs in a new version.

lminiero · 2020-02-04T11:32:42Z

Merging as, while we currently don't do anything with the feedback, it's still useful as it is anyway.

tmatth · 2020-02-06T23:13:40Z

@lminiero I don't understand why yet, but this change is greatly reducing our outgoing FPS (the previous commit b501e17 is fine though). I will dig further on my end.

lminiero · 2020-02-07T07:57:13Z

@tmatth yeah someone mentioned very low bitrates on the demos too. My guess is that something causes the extension NOT to be negotiated, or it is but there's no feedback, and so Chrome falls back to the minimum bitrate. This is weird because I remember the VideoCall working fine, so it may be a regression in monodirectional streams (unless VideoCall is broken too now). I'll have a look as soon as I am at the office.

lminiero · 2020-02-07T09:38:27Z

@tmatth fixed in 58a8042
It was a stupid typo, where I was checking whether video_send was TRUE, and I should have checked if video_recv was TRUE instead, since we only send the feedback when we receive media from the peer. It was working as expected for bidirectional PeerConnections (e.g., EchoTest and VideoCall) because in that case video_send was indeed TRUE.

tmatth · 2020-02-07T13:08:31Z

@lminiero thanks for the quick fix, confirmed that it's working as expected now.

@mivuDing

* Updated link to project in resources (docs) * Add exception var to catch stmt to fix rollup (meetecho#1848) * Fixed typo * fix nullptr dereference in streaming plugin (meetecho#1855) * VP9 SVC fixes (meetecho#1849) * Fixed SIP hangup not sending CANCEL, when inviting (fixes meetecho#1856) * Use strtol more, and add checks when atoi is used (meetecho#1852) * Fixed broken code in AudioBridge * Fixed regression when setting up DataChannels * Fix RTP fuzzing target according to recent VP9 changes. * Fixed rare race condition in HTTP plugin that could cause leak (fixes meetecho#1665) * add missing closing curly bracket (meetecho#1859) * Don't scan libnice version if it wasn't retrieved (fixes meetecho#1858) * Fixed wrong clock rate being used for RTP header updates when using G.722 * Feature/ignore unreachable ice server (meetecho#1854) * Keep track of clock rates associated to payload types, for RTCP * Don't send RTCP SR if outgoing media has been disabled via SDP update * Bumped version in postprocessing tool as well * Fixes to RTSP latching procedure (fixes meetecho#1536, replaces meetecho#1851) (meetecho#1866) * New functionality to add custom Contact URI params to SIP REGISTER (meetecho#1874) * Reduced verbosity of some lines in the SIP plugin * Reduced default twcc_period value from 1s to 200ms * SIP plugin: custom (non-standard) headers on incoming events (requests) (meetecho#1873) * Bumped to version 0.8.0 * Gzip compression utility in the core (and sample event handler) (meetecho#1846) * New category of plugins for modular logging (meetecho#1814) * Fixed linking error for post-rocessing tools after recent changes * Remove option to enable rtx (now always supported, when negotiated) (meetecho#1877) * Updated documentation to include some info on the new logger modules * Avoid gzip functions when fuzzing in OSS and add zlib dependency when fuzzing locally. * Fixed exception to GPL code (see meetecho#713) * Fixed wrong default folder for loggers * Added link to new video on Simulcast and SVC to docs * Add CHANGELOG.md file into the project (meetecho#1885) * Fix RTSP SETUP when url includes query string parameters (fixes meetecho#1869) (meetecho#1875) * Added changelog (and info on tagged versions) to documentation * [Suggestion] Started the refactoring of the janus.js (meetecho#1830) * Make sure libcurl is available before using CURL_AT_LEAST_VERSION (fixes meetecho#1887) * Fixed small typos in demos * Fixed obsolete value for TWCC period default in docs/hints * Make sure the installed libcurl knows about CURL_AT_LEAST_VERSION * Fixed variable shadowing * Added fwrite checks in record.c (warnings only) * Updated changelog (v0.8.0) * Bumped to version 0.8.1 * Remove SIPre plugin from the repo (meetecho#1894) * Binary data support in data channels (meetecho#1878) * Fixed typo in SIP plugin * Allow RTCP ports to be picked randomly using 0, in Streaming plugin * Check if rtcp port is > 0 before creating a RTCP socket. * Revert "Check if rtcp port is > 0 before creating a RTCP socket." This reverts commit a0b7dbf. * Check if rtcp port is > 0 before creating a RTCP socket, in Videoroom plugin. * Add in mountpoint/forwarder create response the allocated RTCP ports. * he 'referred_by' field currently holds the SIP URI value copied from the (meetecho#1896) * Fixed warnings introduced in meetecho#1896 * Fixed leak in SIP plugin (fixes meetecho#1897) * Fixed occasional memory leak in Streaming plugin (fixes meetecho#1900) * Fix out of bounds array access for last_spatial_layer (meetecho#1906) * startup: only close the logger directory if it was opened (meetecho#1903) * Only close the event handlers directory if it was opened (see meetecho#1903) * fixed typo (meetecho#1916) * Move loggers cleanup to end of logger thread (fixes meetecho#1904) * Fixed late initialization of janus.js constructor callbacks (fixes meetecho#1912) * Added reference to Snap repo in resources (docs) * Fixed warnings when building DTLS bio code * Don't keep TextRoom plugin loaded if data channels were not compiled * Updated year in demos and docs * Use sendBeacon instead of sync XHR in onbeforeunload (fixes meetecho#1902) (meetecho#1918) * Fixed occasional buffer overflow error when post-processing H.264 recordings * Increase buffer when post-processing VP8/VP9 recordings too (see previous commit) * Updated Changelog * Bumped to version 0.8.2 * Fix a possible race condition when joining as a subscriber and destroying the session. (meetecho#1911) * More verbose output on postprocessing output error * Fixed reference to deprecated configuration file * Added check on AudioBridge instance in setup_media (fixes meetecho#1923) * Added missing check on SDP attribute value existence * Add new configuration property to add protected folders not to save to (meetecho#1919) * Fixed undefined reference when building postprocessor utilities * Better parsing of RTSP messages (see meetecho#1922) (meetecho#1925) * Fixed undefined reference when building fuzzers * Add missing mutex unlocks in videoroom message handler. * Add math library when fuzzing locally. * Add audio skew compensation to janus-pp-rec. (meetecho#1870) * Updated man file for janus-pp-rec * Remove odd respond to automatically responded OPTIONS request (meetecho#1930) * Fix g_async_queue usage (meetecho#1929) * typo (meetecho#1934) AudioBridge documentation typo in request mute|unmute * Fixed broken links in docs (plugins list) * Removed deprecated warning in screensharing demo * Removed deprecated text from screensharing demo * Fixed helpers not being able to send SUBSCRIBE requests in SIP plugin * Small tweaks after static analysis * Added Coverity badge * Janus Travis CI integration (meetecho#1932) * Updated Changelog (0.8.2) * Bumped to version 0.9.0 * Refactoring of core-plugin callbacks and RTP extensions termination (meetecho#1884) * Support for transport-wide CC on outgoing streams (meetecho#1889) * Dynamically update NACK queue size depending on RTT (meetecho#1867) * Fixed broken RTP fuzzer * Fixed typo when adding audio attribute to SDP * Fixed RTCP parsing issue found by OSS-fuzz * Fix volume-related functions in janus.js (meetecho#1935) * Fixed leak when parsing broken TWCC RTCP message (Credit to OSS-Fuzz) * Add travis_retry to git clone commands. * Fixed occasional segfault when parsing TWCC RTCP message (Credit to OSS-Fuzz) * Add OSS-Fuzz badge. * Fixed regression on video bitrates when using monodirectional PeerConnections * Update janus_audiobridge.c (meetecho#1938) The target of participant should also acknowledge the latest mute/unmute status which has been made by administrator. * Travis libnice clang flags (meetecho#1941) Do not check cast-alignment errors when compiling libnice with clang. * Fixed occasional error messages on console when trying to add RTP extensions * Update debugging section in Janus documentation. * Optimized parsing of TWCC RTCP message (Credit to OSS-Fuzz) * Renamed corpora file * Avoid RTP header memory misalignment in rtx packets (meetecho#1943) * We should allow to have ICE-TCP enabled without ICE Lite. Recent versions of libnice allow this combination and gather tcp passive candidates etc. in this setup. (meetecho#1946) * conf: transports: document events option (meetecho#1952) * Updated Changelog (0.9.0) * Bumped to version 0.9.1 * Configurable global prefix for log lines (meetecho#1940) * add missing callbacks.error check (meetecho#1959) * janus_sip: add missing check for NULL (meetecho#1963) Fixes meetecho#1962 * Remove Sofia reference from the title of the SIP demo * rtp: drop dead code in rtp_header_update callers (meetecho#1964) * Subtype for some event, and better docs for event handlers (fixes meetecho#1953) (meetecho#1957) * Added link to new event handlers documentation to the doc main page * Removed unused variables * Added license badge to the README * Small tweaks to demo intro text * Detect H264 key frames with smaller SPS units (meetecho#1965) Reduces the H264 keyframe length check from 16 to 6 bytes. 6 bytes seems to be the lower bound of any possibly valid SPS NAL unit, based on Section 7.3 of the H264 specification. For reference, we have been observing Chrome 80 producing SPS units of 12 bytes or less. * Support for strings as unique IDs in AudioBridge, VideoRoom, TextRoom (meetecho#1880) * If glib is too old, generate uuid manually when needed (see meetecho#1880) * Fixed errors creating VideoRoom when strings are used (see meetecho#1880) * Remove duplicated codecs when answering SIP call (meetecho#1966) * Fixed a couple of JSON attributes in VideoRoom when strings are used (see meetecho#1880) * Make sure a publisher exists when asking for a VideoRoom subscriber renegotiation (fixes meetecho#1970) * Added errno info when socket operations fail in Streaming plugin * Fixed typos in TextRoom * Support for strings as unique mountpoint IDs in Streaming plugin (meetecho#1969) * fix meetecho#1967 (meetecho#1968) Fixed error callback not being invoked when an HTTP error happens trying to attach to a plugin * Added checks on nice_address_set_from_string (fixes meetecho#1973) * Fixed broken method signature in Streaming plugin when not using libcurl * Remove /root from the list of protected folders. Make comment text more clear. * Valgrind fixes for sockaddr structs (meetecho#1976) Avoid use of uninitialized members * Hide libcurl from pkg-config when testing travis-ci with LIBCURL = NO. * Fixed leak when creating Streaming mountpoint dynamically * Reduced log level to info when logger and event handlers are not found (meetecho#1980) * Always use base SSRC when recording VideoRoom simulcast participant * Removed wrong comment * Fixed broken DTMF in SIP demo * Add UI to SIP demo to remove helpers, when created * Fixed occasional missing referred-by info in SIP demo * Reply to incoming REFER with 202 right away, not 100, in SIP plugin * Added more checks on nice_address_set_from_string (fixes meetecho#1973) (meetecho#1981) * Several enhancements to SIP demo * Fixed abort at server shutdown after using SIP transfers * Fixed typo in SIP demo code * Updated Changelog (0.9.1) * Bumped to version 0.9.2 * Make prebuffering in AudioBridge configurable (meetecho#1975) * Add G.711 support to the AudioBridge plugin (meetecho#1979) * Added maximum value for AudioBridge prebuffering property * Converted HTTP transport plugin to single thread (meetecho#1173) * Added -f to rm in html Makefile.am (fixes meetecho#1985) * Small fixes for TypeScript declaration file (meetecho#1986) Based on the current RTCConfiguration spec (https://w3c.github.io/webrtc-pc/#dom-rtcconfiguration), iceServers does not expect an array of strings. Updating to type provided by TypeScript's lib.dom.d.ts * ice: ensure that stream is non-NULL (meetecho#1987) This fixes a crash on later stream checks (e.g., transport_wide_cc et al). * Fixed typo in querylogger_parameters (copy/paste error) (meetecho#1989) * Fixed double unlock when listing private rooms in AudioBridge (meetecho#1988) * Make sure the session still has a reference when cleaning up HTTP requests * Fixes to leaks and race conditions in VoiceMail plugin (meetecho#1993) * Several fixes to session management in VideoCall plugin (meetecho#1994) * update dtls ciphers (meetecho#1995) * Implement ECDSA Certificate generation (meetecho#1997) * Small tweaks to meetecho#1997 (renamed, moved and documented RSA property in janus.jcfg) * Fix rare race condition when claiming sessions (meetecho#1990) * Fix occasional deadlock in VideoRoom (2) (credits to @mivuDing, fixes meetecho#1982) (meetecho#1984) * Added option to enforce validation on DTLS certificates (meetecho#1992) Made DTLS ciphers configurable as well * Fixed typo when renegotiating audio in janus.js (fixes meetecho#2002) * Added option to ignore mDNS candidates (meetecho#1998) * Fixed deadlock when using claim on HTTP transport (fixes meetecho#2000) * Support for RTSP 'Content-Base' header in Streaming plugin (meetecho#1999) * Added link to FOSDEM 2020 talk on RTP forwarders to the docs * Fixed small leak in SIP plugin when holding calls * Added called URI to 'incomingcall' and 'missed_call' events in SIP plugin * Add repos for openSUSE and SUSE (meetecho#2009) * Use user_id_str for kicked, leaving, and unpublished events, if enabled. (meetecho#2010) Co-authored-by: Michael Shiel <[email protected]> * http_transport: add NULL checks (meetecho#2012) Refs meetecho#2005 * Update media direction in SIP plugin if remote address is 0.0.0.0 ('hold' fix) (meetecho#2013) * Prepare RTCP Sender Reports by considering the last RTP timestamp sent. (meetecho#2007) * Track pending nack cleanup tasks and cancel them when freeing a stream. (meetecho#2014) * Fixed typo in janus.js error code (fixes meetecho#2018 * Reverted change on janus.js (see meetecho#2018) * Resolve mDNS candidates asynchronously with GResolver (see meetecho#1998) (meetecho#2004) * Reference count janus_request instances (meetecho#2020) Added better management of refcount on HTTP session when using it too, and refcount support to hanus_http_msg as well * Updates to mutex unlocking in textroom and videoroom plugins (meetecho#2026) * Updated Changelog (0.9.2) * Bumped to version 0.9.3 * Add Python aiortc-based functional testing. (meetecho#1971) * test_aiortc: cleanup (meetecho#2027) * Fixed missing refcount init for Admin API (fixes meetecho#2029) * Bumping back to 0.9.2 to re-tag * Updated changelog for 0.9.2 * Bumped to version 0.9.3 (again) * janus_http: return earlier if request is NULL (meetecho#2031) * Fixed janus-pp-rec build warnings when using ffmpeg >= 4.x * Fixed VideoRoom destroy not working when using strings * Fixed av_register_all deprecation check in post-processor * plugins: drop tautology (meetecho#2041) gateway is always set before initialized, so the latter is always true. * Don't set ICE credentials when parsing remote credentials (meetecho#2046) * Detect libsrtp(2) using pkg-config (fixes meetecho#2019) (meetecho#2033) * Added support for static Opus files to Streaming plugin (meetecho#2040) * Added support for generic metadata to Streaming mountpoints * Fixed printout of metadata in Streaming demo * Added notes on building libsrtp (see meetecho#2024) * Add configurable DSCP ToS for PeerConnections (meetecho#2055) * Always add remote candidates from the libnice loop (see meetecho#2045) (meetecho#2048) * Fixed Streaming destroy not working when using strings * Use refcount for Streaming plugin helper threads (meetecho#2039) * Added option to disable building AES-GCM support (see meetecho#2024 and meetecho#2054) * Fixed typo * Fixed outdated info in VideoRoom docs * Fixed syntax error in sample Streaming plugin configuration file * Support for additional constraints on screenshare media (meetecho#2043) * refactoring-clean up (const-var, semicolons, ===, etc.) (meetecho#2044) * Reference subscriber when handling related messages (see meetecho#2045) (meetecho#2061) * Added option to configure time needed to detect a missing simulcast substream (meetecho#2063) * Reverted isTrickleEnabled check in janus.js (fixes meetecho#2064) * Don't show warnings for rtx RTCP packets * Made libnice warning clearer, and upped suggested version (fixes meetecho#2069) * Add missing info to videoroom "list" response (meetecho#2068) * Use custom GSource to handle HTTP request timeouts (see meetecho#2062 and meetecho#2066) (meetecho#2075) * Define the libnice version string as extern in version.h (fixes gcc10 error) * Fixed AudioBridge create API not working properly when using string IDs * Fixed a few typos in AudioBridge errors * Fix copy-paste error in Streaming plugin docs * Fix libasan use after free in janus_videoroom_handler when events are enabled (meetecho#2091) * Added project to resources in the docs * Return mountpoint IP addresses, if a bind interface/IP was provided * Swap RR/SR Report Blocks if the first block contains rtx data. (meetecho#2089) * Add support for playback of audio files in AudioBridge (meetecho#2088) * Updated Changelog (0.9.3) * Bumped to version 0.9.4 * Fixed returned address when adding multicast Streaming mountpoints * More checks when hanging up VideoRoom subscriber (see meetecho#2087) (meetecho#2093) * Added new docker image to the resources in the docs * Updated AudioBridge documentation with new playback feature * Don't wait forever for candidates when half-trickling * Add some missing static declarations to HTTP and WS transports. Co-authored-by: Lorenzo Miniero <[email protected]> Co-authored-by: Agustin Polo <[email protected]> Co-authored-by: Yongje Lee <[email protected]> Co-authored-by: Alessandro Toppi <[email protected]> Co-authored-by: Sebastian Schmid <[email protected]> Co-authored-by: Imer Husejnovic <[email protected]> Co-authored-by: Oscar <[email protected]> Co-authored-by: Irek <[email protected]> Co-authored-by: Tristan Matthews <[email protected]> Co-authored-by: Jon Rafkind <[email protected]> Co-authored-by: kuekerino <[email protected]> Co-authored-by: Yurii Cherniavskyi <[email protected]> Co-authored-by: Meirza Arson <[email protected]> Co-authored-by: Groupboard <[email protected]> Co-authored-by: Cameron Lucas <[email protected]> Co-authored-by: hxl-dy <[email protected]> Co-authored-by: Alessandro Amirante <[email protected]> Co-authored-by: mp16 <[email protected]> Co-authored-by: Paul Zhang <[email protected]> Co-authored-by: Philipp Hancke <[email protected]> Co-authored-by: Sean DuBois <[email protected]> Co-authored-by: Ancor Gonzalez Sosa <[email protected]> Co-authored-by: Michael Shiel <[email protected]> Co-authored-by: Michael Shiel <[email protected]> Co-authored-by: agclark81 <[email protected]> Co-authored-by: Alex Pavlov <[email protected]> Co-authored-by: alexamirante <[email protected]> Co-authored-by: Federico Lorenzi <[email protected]>

lminiero added 4 commits December 2, 2019 17:38

Refactoring of core-plugin callbacks and RTP extensions termination

2f0b8da

Fixed RTP extension insertion in some edge cases

1918f96

Added SDP utils flag to offer extensions, so plugins don't have to ad…

b75bf35

…d them manually

Support for transport-wide CC on outgoing streams

ffec3c6

tmatth reviewed Dec 5, 2019

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lminiero mentioned this pull request Dec 6, 2019

Videocall Plugin low quality in Chrome #1890

Closed

Fixed inverted management of received/not-received vector

c8fc34d

This was referenced Dec 9, 2019

Bitrate dropping on the caller side #1893

Closed

Videocall: Remote video with poor quality/bitrate #1861

Closed

lminiero added 2 commits December 12, 2019 17:39

Aligned to version 0.8.1

7b16fae

Aligned to version 0.8.1

6bacba3

lminiero mentioned this pull request Dec 19, 2019

fix janus crash when pidfile is locked by another process #1901

Closed

lminiero added 3 commits December 19, 2019 12:32

Made video-orientation extension info more developer friendly

4ac2791

Relay RTP extensions info through VideoRoom as well

f646881

Aligned to latest fixes in 'plugins-media' parent branch

a465de1

lminiero and others added 5 commits January 17, 2020 16:22

Aligned to latest changes in master

402514e

Aligned to latest changes in parent branch

d97b9d8

added video orientation and audio level extensions (#1928)

a367ecf

Integrated PR #1928

8d172b5

Aligned to master

247218f

Aligned to changes in parent

92933c8

lminiero changed the base branch from plugins-media to master February 4, 2020 11:24

Fixed conflicts with master

af240a4

lminiero merged commit 298a38b into master Feb 4, 2020

lminiero deleted the transport-cc branch February 4, 2020 11:32

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lminiero commented Feb 7, 2020

tmatth commented Feb 7, 2020

Support for transport-wide CC on outgoing streams #1889

Support for transport-wide CC on outgoing streams #1889

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lminiero commented Feb 7, 2020

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tmatth Dec 5, 2019 •

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lminiero commented Dec 19, 2019 •

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