Commit Graph

262 Commits

Author SHA1 Message Date
prlanzarin
93b5f4c93d feat(bbb-html5): add a general forceRelay flag
public.media.forceRelay forces relay usage on all browsers, environments and media modules

If true, overrides public.kurento.[4~forceRelayOnFirefox
2021-12-09 11:35:56 +00:00
prlanzarin
d59f910d84 fix(fullaudio): remove cross wired configs
Offer direction should only be controlled that way in listen only
2021-12-08 23:45:45 -03:00
Anton Georgiev
578332a094
Merge pull request #13731 from schrd/cluster_proxy
Allow BBB to run behind a proxy the avoid gUM permission queries per node
2021-12-03 11:32:07 -05:00
prlanzarin
da6ab02122 chore: add forceRelayOnFirefox option (false by default)
- forceRelayOnFirefox: whether TURN/relay usage should be forced to work
around Firefox's lack of support for regular nomination when dealing with
ICE-litee peers (e.g.: mediasoup).
  * See: https://bugzilla.mozilla.org/show_bug.cgi?id=1034964
- iOS endpoints are ignored from the trigger because _all_ iOS browsers
  are either native WebKit or WKWebView based (so they shouldn't be affected)
2021-11-30 20:31:12 +00:00
Daniel Schreiber
c46556e1f6 Allow BBB to run behind a proxy the avoid gUM permission queries per node
The idea is to run a loadbalancer node which maps each BBB node to a
path. That way each user gets only one gUM permission query for a
cluster. The loadbalancer node only serves the html5 client, each BBB
node will serve its own API and handle the websockets for freeswitch and
bbb-webrtc-sfu.

Configuring a cluster setup
===========================

* let bbb-lb.example.com be the loadbalancer node
* let bbb-01.eaxmple.com be a BBB node

Loadbalancer
------------

On the loadbalancer node add an nginx configuration similar to this one
for each BBB node:

```
location /bbb-01/html5client/ {
  proxy_pass https://bbb-01.example.com/bbb-01/html5client/;
  proxy_http_version 1.1;
  proxy_set_header Upgrade $http_upgrade;
  proxy_set_header Connection "Upgrade";
}

```

BBB Node
--------

On the BBB node add the following options to
`/etc/bigbluebutton/bbb-web.properties`:

```
defaultHTML5ClientUrl=https://bbb-lb.example.com/bbb-01/html5client/join
presentationBaseURL=https://bbb-01.example.com/bigbluebutton/presentation
accessControlAllowOrigin=https://bbb-lb.example.com
```

Add the following options to `/etc/bigbluebutton/bbb-html5.yml`:

```
public:
  app:
    basename: '/bbb-01/html5client'
    bbbWebBase: 'https://bbb-01.eaxmple.com/bigbluebutton'
    learningDashboardBase: 'https://bbb-01.eaxmple.com/learning-dashboard'
  media:
    stunTurnServersFetchAddress: 'https://bbb-01.eaxmple.com/bigbluebutton/api/stuns'
    sip_ws_host: 'bbb-01.eaxmple.com'
  presentation:
    uploadEndpoint: 'https://bbb-01.eaxmple.com/bigbluebutton/presentation/upload'
```

Create the following unit file overrides:

* `/etc/systemd/system/bbb-html5-frontend@.service.d/cluster.conf`
* `/etc/systemd/system/bbb-html5-backend@.service.d/cluster.conf`

with the following content:

```
[Service]
Environment=ROOT_URL=https://127.0.0.1/bbb-01/html5client
```

Change the nginx `$bbb_loadbalancer_node` variable to the name of the
load balancer node in `/etc/bigbluebutton/nginx/loadbalancer.nginx` to
allow CORS requests:

```
set $bbb_loadbalancer_node https://bbb-lb.example.com
```

Prepend the mount point of bbb-html5 in all location sections except
from the `location @html5client` section in
`/etc/bigbluebutton/nginx/bbb-html5.nginx`

```
location @html5client {
    ...
}
location /bbb-01/html5client/locales {
    ...
}
```
2021-11-20 22:13:47 +01:00
Mario Jr
a719f8f5e4 fix(audio): update fullaudio bridge according to sonar's checks 2021-11-09 12:10:44 -03:00
Mario Jr
117bb91a0e fix(audio): rename fullaudio bridge to FullAudioBridge
updated whitelist, also according to sonar checks
2021-11-09 12:10:36 -03:00
Mario Jr
db4e2f9c47 update(audio): correctly retrieves the media-server-fullaudio from meta params
Previously we were using the same for listenonly, which for default
environments points to kurento, and didn't make any difference in media server
selection.
This could be problematic in those environments where meta param
for listeonly media server is set , though.
Fullaudio has now it's own meta param that can be passed through API:
"meta_media-server-fullaudio"
2021-11-09 12:09:28 -03:00
Mario Jr
f9dbefe764 feat(audio): basic funcionality for fullaudio bridge
This commit allows user to join/leave audio using the fullaudio bridge.
This is still under development, but to use this now we must set values of
skipCheck to false, and defaultFullAudioBridge to fullaudio. This
depends on newest version of bbb-webrtc-sfu
2021-11-09 12:09:18 -03:00
Mario Jr
42778adeaf feat(audio): add base code for fullaudio bridge
This bridge will work with bbb-webrtc-sfu to handle microphone audio.
2021-11-09 12:08:45 -03:00
Mario Jr
86ef834476 fix(audio): bridges can be dynamically imported
Added bridge whitelist , to allow bridge modules to be dynamically imported.
2021-11-09 12:08:18 -03:00
Mario Jr
7e218c3eca feat(audio): add bridge configurable scheme
we are now able to switch between audio bridges, by selecting it in
config files.
2021-11-09 12:08:09 -03:00
Anton Georgiev
24d672a832 Merge branch 'v2.4.x-release' of github.com:bigbluebutton/bigbluebutton into merge-24-dev 2021-10-14 15:13:17 +00:00
Mario Jr
7ed1cbfc30 Merge branch 'v2.4.x-release' into fix-13242 2021-09-29 09:20:58 -03:00
Mario Jr
519883e89a update(audio): adjust reference to ignoreCallState flag 2021-09-24 14:48:22 -03:00
prlanzarin
1cb412529d feat(webrtc): add EXPERIMENTAL option to disable ICE candidate signaling
ICE lite servers (eg mediasoup) dont need candidates signaled out-of-band; neither does KMS in certain scenarios

Disable their signaling saves us some ticks in bbb-webrtc-sfu and some bandwidth all around
2021-09-24 17:24:06 +00:00
Anton Georgiev
7c11600518
Merge pull request #13266 from mariogasparoni/dynamic-bridges
feat(audio): add bridge configurable scheme
2021-09-22 16:00:46 -04:00
Mario Jr
1c08fc852a fix(audio): audio controls crash when ending call during brekout audio transfer
Restored the old behavior when ending breakout rooms while user is in the
breakout audio transfer, which is to the trigger the reconnection to the audio
in the main room.
This behavior could be improved by (instead of reconnecting) transfering user
back to the main room, but this requires some changes in akka-apps/fsesl
which can be treated in a different issue.

Closes #13242
2021-09-21 11:22:05 -03:00
Mario Jr
b74e4c6ca5 fix(audio): bridges can be dynamically imported
Added bridge whitelist , to allow bridge modules to be dynamically imported.
2021-09-01 13:22:43 -03:00
Mario Jr
4fcd051223 feat(audio): add bridge configurable scheme
we are now able to switch between audio bridges, by selecting it in
config files.
2021-08-31 15:50:53 -03:00
prlanzarin
58a8e99195 feat: add media server configuration via metadata
Applies to video, listen only and screen sharing

New metadata values: media-server-video, media-server-listenonly, media-server-screenshare; parameter is a String
2021-08-31 00:39:45 +00:00
prlanzarin
c57fb0b388 feat(audio): add media server adapter config for listen only in bbb-html5
Allows configuring, via bbb-html5, which media server adapter will be used by listen only; server wide
2021-08-31 00:24:41 +00:00
Anton Georgiev
4ecb24b4fa Merge branch 'v2.3.x-release' of github.com:bigbluebutton/bigbluebutton into merge-aug30 2021-08-30 18:11:16 +00:00
Anton Georgiev
34453b8528
fix: Fix log message (LGTM)
Updating string for audio connected audio. Thanks @stweil
2021-08-25 13:17:27 -04:00
Mario Jr
cf366e5090 Get user and audio data, shows it and allows it to be copied to clipboard
Using getStats api and peer information to retrieve upload/download rates
and transport information
2021-08-12 16:39:04 -03:00
prlanzarin
54d04fdb77 refactor(listen-only): let the server generate subscriber offers 2021-08-12 13:45:18 +00:00
Mario Jr
c0d1255924 Merge branch 'v2.3.x-release' into fix-1005-logging 2021-07-30 12:28:34 -03:00
prlanzarin
168f66310e fix(listen-only): avoid leaving a dangling HTMLMediaElement in paused state 2021-07-23 02:35:47 +00:00
Mario Jr
2f75f6107e fix(audio): prevent false positive alerts of 1005 error
After reconnecting (with 1007 or 1005), user may gets 1005 when meeting
is ended by moderator.
2021-07-22 09:28:14 -03:00
Mario Jr
6db69c39d8 fix(audio): prevent duplicated error logging
We now let audio-manager log errors, using the specified error code.
These errors are still logged in bridge layer, but as a warning.
This commit doens't change error codes behavior, they are still being
logged as errors and with the same code numbers.
2021-07-20 11:10:04 -03:00
Mario Jr
431a6c7c3b chore(audio): add secondsToGatherIce as extraInfo in server log 2021-07-16 11:16:24 -03:00
Mario Jr
f5713869bf fix(audio): correct log the time needed for ice gathering 2021-07-15 11:15:53 -03:00
Mario Jr
287c4eb682 feat(audio): use kurento's trickle-ice to improve mic negotiation
Here's what we do when user activates mic:
1 - When we do something similar to listenonly's joining process
until we find a valid candidate-pair. The information about this
local candidate is store.
2 - We then start a new userAgent, and as soon as browser finds
a candidate with the same local ip address, we leave only this
candidate in the SDP and send this to FreeSWITCH. SDP should
contain only a single candidate.
3 - The rest of signaling process is basically the same.
2021-07-01 16:26:44 -03:00
Ramon Souza
9d590a74d3 remove unused imports 2021-05-03 09:24:07 -03:00
Mario Jr
13bb0c8493 fix: match exact deviceId when retrieving mic streams
Needed for firefox for consistently changing microphone using device selector
Refs #12054
2021-04-26 22:08:22 -03:00
Mario Jr
2f78fc05d2 fix: breakout audio don't use previously selected output device
When joining breakout audio, the output device selected in the main room is
used in breakout.
When returning from breakout rooms, the output audio device previously set in
the main room is restored.

Some specific info:
SIPSession doesn't handle Storage anymore, we do this in SIPBridge, since
it has more info about the current selected device and it doesn't depend of
a session being oppened.
We also now pass the  output device ID when joining audio sessions. We can
then keep this information in the Storage.
Closes #11663
2021-04-23 11:28:30 -03:00
Mario Jr
e8d59ed14a fix: mic selection (firefox/all browsers) and muted alert when mic is changed
This commit contains three fixes: one already reported and two detected
during the investigation of the solution.
This started as a fix for firefox (#12023), but i also fixed the muted
alert/banner when device changes: the banner wasn't detecting device changes,
unless audio was deactived/actived.

There's another fix for the microphone stream: we now keep sender's track
disabled if it was already disabled for the sender's track of the previous
selected device.

Also did small refactor for eslint checking.

Some technical information: in sip bridge (bridge/sip.js), setInputStream and
liveChangeInputDevice function were both fully turned into promises, which
guarantees we have everything ready when it resolves to the respective values.
This helps AudioManager (audio-manager/index.js) to sequentially sets and
tracks the state of the current microphone stream (inputStream), when calling
liveChangeInputDevice function: we first set the current stream to null,
creats a new one and then set it to the newly created value - this is needed
because MutedAlert (muted-alert/component.jsx) can then gracefully
allocate/deallocate the cloned stream when it is set to a non-null/null value
(the cloned stream is used for speech detection with hark).
In MutedAlert we also make sure to enable the cloned stream's audio
tracks, just in case the user change the device when muted (audio track is
disabled in this case), which also leaves the cloned stream muted (we then
enable the track to allow speech detection).

Closes #12023
2021-04-16 10:45:40 -03:00
znerol
5399473f46 Send browser UA string in SIP UA, also add BBB server and client version 2021-04-04 22:21:33 +02:00
Mario Jr
951fc0ade8 fix: MEDIA_ERROR when joining room using Firefox
Firefox doesn't create a  device called 'default' and we were trying
to set this when user is joining the room. We don't do this anymore, letting
devices to be changed when there's some user request.

Moved outputDeviceId inputDeviceId information to be managed in bridge
(just like we do with inputDeviceId), we don't store this duplicated
information in audio container anymore.

Fixed the eslint warning in "playAlertSound(url) { ..."

We are safe to let users try to change input/output devices because the
device list is retrieved from enumerateDevices.
2021-04-01 15:53:43 -03:00
Anton Georgiev
ff45cccd66 Merge branch 'v2.2.x-release' of github.com:bigbluebutton/bigbluebutton into apr1-merge 2021-04-01 18:11:13 +00:00
Ramon Souza
05a0d5afbf move browser info to helper + remove unused libs 2021-04-01 08:14:24 -03:00
Mario Jr
60378e8c63 Merge branch 'develop' into merge-update-pr-10129 2021-03-29 19:07:27 -03:00
Mario Jr
cddca95ad1 Fixes and improvements on pr #10129 - dynamic change audio device
Allow listenonly users to change output devices
Fixed dynamic audio device change for firefox
Fixed shortcuts for audio join/leave
Show (with a bold font) the current selected device
[performance] Prevent calling mediaDevices.enumerateDevices every time we render
the selector. This adds a delay (~200ms, on my chrome setup) to render this component
[performance] Do not call enumerateDevices to search for new devices, instead we listen on mediaDevices.deviceChange event
Small refactoring and fixed a few errors that were being throw in browser's console
Fixed device selection when this is done in audio-settings modal
Fallback to default device when current device is removed
Truncate device name length
Renamed "Input","Output" labels to "Microphone","Speakers", respectively
Update eslint rule for accessKey
2021-03-29 18:55:17 -03:00
Mario Jr
cec88c996d Merge remote-tracking branch 'tainan/issue-9723' into merge-update-pr-10129 2021-03-29 18:52:12 -03:00
prlanzarin
2e51482c7c audio: make sdpSemantics configurable
Plan B is going to be phased out of Chrome soon and we should be testing it with unified plan
2021-03-29 20:01:33 +00:00
prlanzarin
f20fb3eef2 audio: make sdpSemantics configurable
Plan B is going to be phased out of Chrome soon and we should be testing it with unified plan
2021-03-12 03:04:55 +00:00
Mario Jr
209b06db2e Avoid setting an unknown input deviceId
Without 'exact' match, the browser fallbacks to the default inputDeviceId

This prevents the error (input device error)  when breakout is ended and we try
to skipCheck the microphone when user returns to main room (assuming the
user had the microphone active before joining breakout room).
2021-03-05 00:33:54 -03:00
Mario Jr
de05622d19 Avoid setting an unknown input deviceId
Without 'exact' match, the browser fallbacks to the default inputDeviceId

This prevents the error (input device error)  when breakout is ended and we try
to skipCheck the microphone when user returns to main room (assuming the
user had the microphone active before joining breakout room). Related
to the feature c451666d52
2021-02-28 17:15:08 -03:00
Mario Jr
b582b1ca78 Correctly return MediaStream object when calling audio-manager's inputStream
This was missed in recent audio/sip.js changes.
MediaStream is now returned from SIP.js
Related to #10733
2021-01-29 19:05:51 -03:00
Anton Georgiev
09e6ba8dfd Merge branch 'v2.2.x-release' of github.com:bigbluebutton/bigbluebutton into merge-Jan30 2021-01-29 20:58:05 +00:00
Mario Jr
f2de2806eb Add setting for enabling/disabling microphone audio filters
As explained in #11143, disabling audio filters is desired in some scenarios.
This basically adds an option for user to disable default constraints.
When user doesn't change this value in Settings > Application, the default
value for each audio constraints is retrieved from settings.yml.
When user changes this value in Settings > Application, audio
filters (AGC, Noise Supression and Echo Cancellation) are all set to
true/false, according to the value selected in the Settings GUI.

To start it simple, we decided to not to add a different setting in the GUI for
each audio contraint. This may be added in the future, though (perhaps in a
dedicated Audio Settings tab)
This is related to #4873
2021-01-29 14:18:15 -03:00
Mario Jr
b753ef5d8d Fix wrong path in settings for audio constraints
Using now 'application' instead of old 'audio' path
2021-01-26 14:12:18 -03:00
Mario Jr
7858ba94ba Avoid setting empty/undefined constraint
This removes the console warning about invalid constraint
2021-01-25 23:45:27 -03:00
Mario Jr
97c76900cb Add setting for enabling/disabling microphone audio filters
As explained in #11143, disabling audio filters is desired in some scenarios.
This basically adds an option for user to disable default constraints.
When user doesn't change this value in Settings > Application, the default
value for each audio constraints is retrieved from settings.yml.
When user changes this value in Settings > Application, audio
filters (AGC, Noise Supression and Echo Cancellation) are all set to
true/false, according to the value selected in the Settings GUI.

To start it simple, we decided to not to add a different setting in the GUI for
each audio contraint. This may be added in the future, though (perhaps in a
dedicated Audio Settings tab)
This is related to #4873
2021-01-22 16:30:42 -03:00
Mario Jr
92708da447 Allow audio constraints to be changed in bbb-html5's settings.yml
In some scenarios, there's no need for the browser to apply such audio filters. For example, when user's microphone already does audio filtering (echo cancellation, noise supression ...).
This commit doens't change the current behavior (filters still follow browser's default config): admins need to uncomment/set these values if disabling/enabling specific filters if desired.
This is related to #4873
2021-01-12 14:42:27 -03:00
Anton Georgiev
c1ffced27d Merge branch 'v2.2.x-release' of github.com:bigbluebutton/bigbluebutton into develop 2020-12-17 21:14:29 +00:00
Mario Jr
6113113af9 Add option to disable traceSip logs
Audio client logs already cover audio session progress the way we need.
This avoids keepAlive and other unnecessary messages to be logged in browser's console.
If setting is not present, default value is set to false.
2020-12-11 00:31:10 -03:00
Mario Jr
c65fa2b350 Add keep alive message to audio's websocket
This was added as an option (websocketKeepAliveInterval), which is the interval to send keep alive messages.
Setting websocketKeepAliveInterval to 0 disables the keep alive, producing the same old behavior.
This helps avoid websocket disconnection due to socket inactivity, preventing it to unnecessarily reconnect.
Also, sometimes reconnect fails and error 1005 is triggered.
Fixes problems reported in #10985.
Also reduces occurrences of error 1005.
2020-12-10 23:48:01 -03:00
Anton Georgiev
244a239810 Merge branch 'v2.2.x-release' of github.com:bigbluebutton/bigbluebutton into 2020-12-01-merge 2020-12-01 20:02:50 +00:00
prlanzarin
1bef5f37a6 [sip.js] Annotate every audio-manager callback call with a bridge name identifier
Useful when parsing logs to split errors by bridge type
2020-12-01 18:23:14 +00:00
prlanzarin
338e8f8291 [listenonly] Rewrite listen only SFU/Kurento bridge
Fixed listen only reconnection handling

Added proper error handling; now all errors have proper mapped codes which are funneled through to audio-manager logger and should be easier to gauge types of errors

Fixed botched reconnection error rejection, audio modal shouldnt be stuck anymore when it fails

Remove every tie that listen only bridge had to kurento-extension
2020-12-01 18:19:31 +00:00
Mario Jr
214cd12c59 Fix typo on dtmf log 2020-11-26 00:01:58 -03:00
Mario Jr
370e3cb39d Use INFO message as default for sending dtmf on call transfer
Instead of sending using rfc4733 standard, we use INFO message for all transfers
INFO message was used in older SIP.js version. Although this is not a standard for sending DTMF tones, this has more reliability (once it sent over TCP)
This might reduce occurrences of 1008
2020-11-25 18:33:45 -03:00
Mario Jr
af48c8977d Reduce delay for the first reconnection attempt of audio's websocket
This is the same behavior we used to have on older sip.js version code
By doing this we reduce errors when user try to perform join/hangup during an websocket reconnection
2020-11-23 12:40:38 -03:00
Mario Jr
2b89dd7db2 Do not trigger reconnect when ICE connection terminates before hanging up process is finished
This could leave users to have your audio reconnected in the main room, while joining a breakout room
Some information can be found in #10528
2020-11-20 00:25:46 -03:00
Mario Jr
9a2fcd27e0 Revert "Prevent Error 1006 when user has two or more occurrences of ':' (colon) in your name"
This reverts commit 0a601359bb.
2020-11-20 00:23:43 -03:00
Mario Jr
0a601359bb Prevent Error 1006 when user has two or more occurrences of ':' (colon) in your name
This happens because FreeSWITCH is not able to parse the "From" header when it has multiple occurrences of ':'. So user is not able to join audio.
To fix, we now changed the "callerId" to use the base64 value of the user name, instead of directly using user's input (the callerId format keeps being a triple like this: <user_id>-bbbID-<base64_encoded_name>).
Once this callerIdName is encoded at the same point it is generated, there shouldn't be server side effects for changing this value; except for those places where the callerName is retrieved by splitting this triple (such as the voice talking-indicator, as described below).
Updated the talking-indicator to retrieve the username from User's object, instead of retrieving from the one username generated by splitting the callerId triple.
This problem also happens in versions <= 2.2.26.
2020-11-17 15:31:43 -03:00
prlanzarin
a1f0276b64 [html5/audio] Add hackViaWs to SIP.js and make it configurable in settings.yml, 1002 workaround
This adds the possibility to configure the SIP Via header to plain WS to allow reverse proxying from WSS to WS, internally, to work around a bug in freeswitch where the WSS stack would get deadlocked due to a still unidentified bug in there that has to do with SSL termination
2020-11-10 15:04:45 +00:00
Mario Jr
b948bea11b Force using plan-b as default sdpSemantics for chrome
Although Chrome's default is now unified plan, Chrome <-> FreeSWITCH ICE connection fails for some Chrome installations (specially those running on Windows).
FS ICE fails when Chromes's SDP has "a=mid:<index>" (instead of "a=mid:audio").
This fixes Error 1010 and situations where echo test takes too long.
This fixes #6414 regression, once we do the same older version of SIP.js used to do.
2020-11-09 21:58:16 -03:00
Mario Jr
46e0c263fe Use iceConnectionState to monitor ICE connection status
We now use both peer's connectionstatechange and iceconnectionstatechange to monitor ICE state for audio sessions.
The same way we did with old sip.js version, we leave iceconnectionstate trigger audio actions , such as connect, disconnect, reconnect.
We still listen for 'failed' state for connectionstatechange event, because chrome triggers this (tested on 86+).
This should reduce the audio error 1010 ocurrences, once some browsers (specially Chrome/Android) don't trigger connectionstatechangeevent.
This might reduce problems reported in #10708, which still needs more investigation though.
2020-11-08 22:43:52 -03:00
Mario Jr
3a689578c6 Monitor peer's iceconnectionstatechange event
This gives more information about ICE connection, combined with onconnectionstatechange event
2020-11-06 09:37:58 -03:00
Mario Jr
2585d957e8 Correctly map WebSocket error
Maps WebSocket's 1006 error to BBB's 1002, the same way it was done with old sip.js version
Set user agent's number of reconnection attempts to the same value as older sip.js version
2020-11-06 09:25:40 -03:00
Mario Jr
3e3b648040 Properly stops userAgent / peer when audio connection/reconnection fails
Changed the maximum attempts of the UserAgent reconnection (this should be changed when binding audio's websocket to meteor's connection state).
Added a log to monitor WS reconnect attempts.
2020-10-28 15:04:30 -03:00
Mario Jr
d1e5f189ba Prevent 1005 error log when user close/reload bbb's window/tab
When closing/reloading tab with active microphone, audio exits successfully but a wrong log-error (1005) is shown.
We now process closing/reloading tab the same way we do when user hangup the call.
2020-10-25 16:12:58 -03:00
Mario Jr
18c20261e1 Change default value of iceGatheringTimeout to current's SIP.js default
For some reason (still investigating), using turn/coturn on 443/tcp makes firefox's iceGathering process (during echo test) takes 12+ seconds (tested on webrtc's trickle page with multiple instances).
This was found when testing the current default (15s) on production with a private turn/coturn server on port 443/tcp. For default bbb setup (stun only), echo test still runs fast.
To avoid adding extra delay to iceGathering on this scenario (Firefox + turn on 443/tcp), i am just setting the default value back to the 5s (old default).
So , for those who wants to reduce the 1004 occurrences, increasing the iceGatheringTimeout could help (just be aware this adds delay on the mentioned scenario).
2020-10-24 08:58:25 -03:00
Mario Jr
a86ff72aa3 Increase default iceGatheringTimeout
Added a default 'MEDIA' option: iceGatheringTimeout. This option allows admin to set a higher ICE gathering timeout, which can help when getting ICE errors during audio negotiation (eg 1004)
Default value set to 15s (current default is 5s).
2020-10-23 11:21:20 -03:00
Mario Jr
993c3a5a8a Do not show reconnect/disconnect message when new ICE candidates are found
Sometimes, when user already joined audio session, RTCPeerConnection may
find new ICE candidates, which triggers 'connected' state for peer's
'onconnectionstatechange' event. When this happens we process this
new state the same way when user is not running an audio session, which
makes html5client popup an annoying 'Audio Connected' message.
The audio keeps working fine, but this can make user think that there's a
connection issue, or the audio is reconnecting, while audio is ok.
2020-10-23 11:20:08 -03:00
Mario Jr
df67d2e680 Better handling audio reconnection
When getting disconnected with 1001 ("websocket closed unexpectedly" error) we were creating a new SIP session, therefore a new FreeSWITCH channel.
While reconnecting the socket, instead of closing the SIP session, we keep it alive during reconnection (audio should keep working in the meantime). When reconnected we keep using this same session (avoiding the creation of an extra one).
We also better handle WebSocket error codes from SIP.js.
FF immediately closes websocket when unloading page, so we now to stop user agent when 'beforeunload' event is triggered, to avoid leaving open sessions in FreeSWITCH when user leaves page.
2020-10-15 11:24:23 -03:00
Mario Jr
e9e436378a Correctly set audio input/output devices
When refusing ("thumbs down" button) echo test, user is able to select a different input device. This should work fine for chrome, firefox and safari (once user grants permission when asked by html5client).
For output devices, we depend on setSinkId function, which is enabled by default on current chrome release (2020) but not in Firefox (user needs to enable "setSinkId in about:config page). This implementation is listed as (?) in MDN.
In other words, output device selection should work out of the box for chrome, only.
When selecting an outputDevice, all alert sounds (hangup, screenshare , polling, etc) also goes to the same output device.
This solves #10592
2020-10-06 20:37:55 -03:00
Mario Jr
e1b9ad3536 Map stun/turn servers into WebRTC's iceServers, when using fallback stun 2020-10-02 16:19:55 -03:00
Mario Jr
49bfe9f48d Set stun/turn server for audio's peer in html5 client
Latest SIP.js version sets this using peerConnectionConfiguration property instead of UserAgent option.
This solves #10569
2020-10-01 10:16:48 -03:00
Mario Jr
619ffa0ec1 Port SIP.js to 0.17.1 release
This considerably changes the way we process audio signaling and start audio elements in user's browser.
We now avoid using AudioContext element for both microphone and listenonly calls, once it is unstable for some iOS devices (cracky audio, user stops hearing audio after a while).
Increased default value for listenOnlyCallTimeout: this avoids activating FreeSWITCH's fallback when ICE negotiation takes longer than 15sec (tested on DO).
Increased listenonly logs.
This fixes #8133 #10388
2020-09-25 20:26:22 -03:00
Tainan Felipe
cb53b42c0e Merge remote-tracking branch 'upstream/develop' into issue-9723 2020-08-10 16:52:58 -03:00
Tainan Felipe
cc9662b6b8 Implement speaker change and add device update feature 2020-08-10 16:43:49 -03:00
prlanzarin
dac3259c48 Merge remote-tracking branch 'pedrobmarin/bbb-multiple-webcams' into upstream-2.2-vpeg-base 2020-08-05 14:20:58 +00:00
Pedro Beschorner Marin
ab31861544 Add a minimum socket validation to full-audio connection 2020-07-29 15:51:19 -03:00
Pedro Beschorner Marin
19e301e28e Add a minimum socket validation to full-audio connection 2020-07-29 13:10:17 -03:00
Tainan Felipe
0ba6ff5cf6 Merge remote-tracking branch 'upstream/develop' into issue-9723 2020-07-27 17:02:44 -03:00
Tainan Felipe
2c61d5ee75 Add input/output dynamic audio change 2020-07-27 16:49:26 -03:00
prlanzarin
1dbafffa26 audio: make listen only call timeout configurable 2020-07-09 18:02:18 +00:00
Joao Siebel
3e95ed0e4b Merge remote-tracking branch 'upstream/v2.2.x-release' into merge-2.2 2020-06-16 16:40:56 -03:00
Anton Georgiev
ddb54273c1 Added callerIdName in audio logs where possible 2020-06-12 21:13:49 +00:00
Anton Georgiev
c9e996de21 Merge branch 'v2.2.x-release' of github.com:bigbluebutton/bigbluebutton into merge-2.2-into-develop 2020-05-25 17:32:24 +00:00
prlanzarin
2cba85e866 html5: refactor STUN/TURN fetch, add fallback STUN, fix deadlock in audio join
Refactored STUN/TURN fetch to be done only once, when successful, per session and cache it in mem to avoid too many reqs. Current way is a bit dumb, this should increase reliability a bit more. The caching is configurable so folks who want to use very short lived TURN credentials can disable it

Add a fallback STUN config option to be used when the default STUN/TURN fetch fails

Clean the safari/no candidate generation pre flight check from 3rd party STUNs

Fix deadlock in audio join when STUN/TURN fetch failed
2020-05-21 04:35:17 +00:00
Anton Georgiev
8129468300 Merge branch 'v2.2.x-release' of github.com:bigbluebutton/bigbluebutton into merging 2020-03-04 23:36:21 +00:00
Chad Pilkey
5a678a4faf react to voice call state when connecting with FS LO 2020-03-04 10:25:54 -08:00
Chad Pilkey
ecfbe5e506 implmenent sip.js fallback for playing audio with web audio api 2020-02-29 00:38:30 +00:00
Anton Georgiev
3754d0ab6f Merge branch 'v2.2.x-release' of github.com:bigbluebutton/bigbluebutton into merge-2.2-into-master-feb-28-2020 2020-02-28 17:47:06 -05:00
Chad Pilkey
6a4ba7a300 watch for voice call state updates instead of DTMFs in the client 2020-02-18 14:03:06 -08:00
Anton Georgiev
4b0452d1fd Merge branch 'v2.2.x-release' of github.com:bigbluebutton/bigbluebutton into feb14-merge-2.2-into-master 2020-02-14 21:43:35 +00:00