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87 Commits

Author SHA1 Message Date
Sergiu Toma ce1fe3ca8d Merge pull request 'Update 'app.js'' (#33) from sergiu-patch-1 into develop
Reviewed-on: #33
Reviewed-by: bmamihai <mihai.bozieru@safemobile.com>
2023-06-07 12:51:12 +00:00
Sergiu Toma 96b770d9c5 Update 'app.js' 2023-06-07 12:50:24 +00:00
Sergiu Toma 4b1f4fab70 Merge pull request 'LH-284: Update duplex documentation workflow diagrams' (#32) from LH-284-update-and-centralize-diagrams-for-video-calls into develop
Reviewed-on: #32
Reviewed-by: Adriana <adriana.epure@safemobile.com>
2023-05-30 11:27:09 +00:00
Sergiu Toma ba7c2186af LH-284: Update duplex documentation workflow diagrams 2023-05-30 11:57:02 +03:00
bmamihai 636e8a9fab Merge pull request 'added build commit no' (#31) from build-commit-no into develop
Reviewed-on: #31
Reviewed-by: Cristi Ene <cristi.ene@safemobile.com>
2023-04-10 23:36:34 +00:00
bmamihai 5da2eb6927 update build.sh after review 2023-04-05 01:57:58 +03:00
bmamihai 97948a5d8c added build commit no 2023-04-02 01:17:18 +03:00
bmamihai b94b1bff86 Merge pull request 'docker' (#18) from docker into develop
Reviewed-on: #18
Reviewed-by: Cristi Ene <cristi.ene@safemobile.com>
2023-02-26 22:11:02 +00:00
bmamihai 5c4f2bf13e Merge branch 'develop' into docker 2023-02-26 22:10:34 +00:00
Sergiu Toma df7f1987f6 Merge pull request 'LH-276: Add close-producer event handler; Update client' (#30) from LH-276-close-producer into develop
Reviewed-on: #30
Reviewed-by: Cristi Ene <cristi.ene@safemobile.com>
2023-02-23 09:46:41 +00:00
Sergiu Toma 7842953faf LH-276: Refactor consumer-resume 2023-02-22 18:57:57 +02:00
Sergiu Toma 4591617b1e LH-276: Fix video consume when initiator is not set yet(at start) 2023-02-22 18:50:37 +02:00
Sergiu Toma 9b3f2f94c8 LH-276: Fix audio when initiator is not set yet(at start) 2023-02-22 18:46:01 +02:00
Sergiu Toma 29a4cd7227 LH-276: Format code 2023-02-22 18:40:39 +02:00
Sergiu Toma c80265fe25 LH-276: Format code 2023-02-22 18:36:50 +02:00
Sergiu Toma abb1533c9b LH-276: Format code 2023-02-22 18:32:41 +02:00
Sergiu Toma 6e3ce9fbb3 LH-276: Format code 2023-02-22 18:30:28 +02:00
Sergiu Toma 4d8adf9eac LH-276: Refactor consume(consumeAudio/consumeVideo); Format code 2023-02-22 18:21:53 +02:00
Sergiu Toma bde02fe250 LH-276: Add close-producer event handler; Update client 2023-02-21 02:41:48 +02:00
Sergiu Toma 0ddb43b4b5 Merge pull request 'LINXD-2342-dispatcher-audio' (#29) from LINXD-2342-dispatcher-audio into develop
Reviewed-on: #29
Reviewed-by: Cristi Ene <cristi.ene@safemobile.com>
2023-02-15 09:14:57 +00:00
Sergiu Toma f02d6af726 LINXD-2342: Added documentation for kind 2023-02-15 11:14:02 +02:00
Sergiu Toma 7ee93d7963 LINXD-2342: Update video client 2023-02-15 10:06:39 +02:00
Sergiu Toma 98212a78cb Update client to consume audio correctly 2023-02-11 21:32:53 +02:00
Sergiu Toma 33ea44ac02 Update client to consume audio correctly 2023-02-11 21:30:21 +02:00
Sergiu Toma 6b822142d0 Add the kind of producer on new-producer event 2023-02-11 21:13:57 +02:00
Sergiu Toma 50e95b93a4 Update client 2023-02-08 19:37:32 +02:00
Sergiu Toma 6eca0808c9 Merge pull request 'Update web client to work with dev' (#27) from update-web-client into develop
Reviewed-on: #27
Reviewed-by: bmamihai <mihai.bozieru@safemobile.com>
2023-01-23 20:37:27 +00:00
Sergiu Toma f728b23b7f Update web client to work with dev 2023-01-23 20:45:08 +02:00
Sergiu Toma 33c0234fb0 Merge pull request 'LINXD-2303: Added client-to-client workflow; Added client-mediadoup workflow' (#26) from LINXD-2303-video-documentation-mobile-mobile into develop
Reviewed-on: #26
Reviewed-by: Cristi Ene <cristi.ene@safemobile.com>
2023-01-19 14:44:24 +00:00
Sergiu Toma f81d5b2dfa LINXD-2303: Update diagrams Mediasoup connect & produce; Update diagram Client-Client workflow 2023-01-19 16:43:46 +02:00
Sergiu Toma ae87b5bd71 LINXD-2303: Added client-to-client workflow; Added client-mediadoup workflow 2023-01-19 16:31:16 +02:00
Sergiu Toma 09e82fb224 Merge pull request 'Moved 'new-producer' in 'transport-produce' handler' (#25) from new-producer-in-transport-produce into develop
Reviewed-on: #25
Reviewed-by: Cristi Ene <cristi.ene@safemobile.com>
2023-01-19 06:54:53 +00:00
Sergiu Toma 1da0170261 Moved 'new-producer' in 'transport-produce' handler 2023-01-18 11:49:02 +02:00
Privileged User aca0a6eac0 improved build to get also the git log to know what is on server 2023-01-17 16:00:52 +00:00
Privileged User 23ca56c8c6 rever build.sh 2023-01-17 15:51:55 +00:00
Sergiu Toma a2d0b6771b Merge pull request 'Add new-producer event; Update client to consume when receives new-producer event' (#24) from LAPI-675-generate-new-producer-in-mediasoup-and-client into develop
Reviewed-on: #24
Reviewed-by: Cristi Ene <cristi.ene@safemobile.com>
2023-01-16 15:40:06 +00:00
Sergiu Toma 9382fd11fb LAPI-675: Remove commented code 2023-01-16 17:39:15 +02:00
Sergiu Toma fb99fd57f8 LAPI-675: Refactor transport-connect new-producer 2023-01-16 17:32:31 +02:00
Sergiu Toma 0f4342777a LAPI-675: Update bundle 2023-01-13 20:12:59 +02:00
Sergiu Toma 69c167e4e9 Add new-producer event; Update client to consume when receives new-producer event 2023-01-13 20:11:24 +02:00
Cristi Ene ad4c1f27e7 Merge pull request 'LAPI-674: Parse transport-recv-connect and consume param' (#23) from LAPI-674-change-events-for-consumer-in-mediasoup-server into develop
Reviewed-on: #23
Reviewed-by: Cristi Ene <cristi.ene@safemobile.com>
2023-01-13 07:52:27 +00:00
Bozieru Mihai b2128c4754 fix build 2023-01-12 16:42:44 +00:00
Bozieru Mihai af53dc1610 fix build 2023-01-12 16:30:39 +00:00
Sergiu Toma f48fe4f9aa LAPI-674: Parse transport-recv-connect and consume param 2023-01-12 16:23:30 +02:00
Sergiu Toma 6fd6ca9755 Merge pull request 'comsume-undefined' (#22) from comsume-undefined into develop
Reviewed-on: #22
Reviewed-by: Cristi Ene <cristi.ene@safemobile.com>
2023-01-12 13:31:46 +00:00
Sergiu Toma 1b72941a32 Moved callId in consumer-resume 2023-01-12 15:09:21 +02:00
Sergiu Toma c037240f47 Refactor consumer-resume 2023-01-12 14:32:48 +02:00
Sergiu Toma 9716bddd71 Added log 2023-01-12 12:50:40 +02:00
Sergiu Toma e6c08a2c1d Refactor consumer-resume v3 2023-01-12 12:46:46 +02:00
Sergiu Toma 1342e812e6 Refactor consumer-resume 2023-01-12 12:45:23 +02:00
Sergiu Toma d8405eccc7 Refactor consumer-resume 2023-01-12 12:44:11 +02:00
Sergiu Toma 5bf31d452f Check for initiatorConsumerAudio before resume() 2023-01-12 12:38:40 +02:00
Sergiu Toma c034610471 Rollback 2023-01-12 12:38:01 +02:00
Sergiu Toma df5b3eab90 Check for initiatorConsumerAudio before resume() 2023-01-12 12:31:52 +02:00
Cristi Ene b0293230b6 Merge pull request 'add-diagrams-doc' (#21) from add-diagrams-doc into develop
Reviewed-on: #21
Reviewed-by: Cristi Ene <cristi.ene@safemobile.com>
2023-01-10 17:51:05 +00:00
Sergiu Toma 391e8eb6f0 Added diagram for close call before accept/reject 2023-01-10 13:17:57 +02:00
Sergiu Toma 3139a625a2 Added workflow diagrams 2023-01-10 12:05:29 +02:00
bmamihai 48eee903a5 added command for limiting cpu/memory 2023-01-05 14:31:51 +02:00
bmamihai d5bc0cd1d3 run under host network 2023-01-05 13:18:13 +02:00
bmamihai 1c353d7c88 Merge branch 'develop' into docker 2023-01-05 01:14:56 +02:00
Sergiu Toma 9d43b7ec0c Merge pull request 'LINXD-2270: Allow server and web client to have full duplex' (#20) from LINXD-2270-full-duplex into develop
Reviewed-on: #20
Reviewed-by: Cristi Ene <cristi.ene@safemobile.com>
2022-12-27 11:12:38 +00:00
Sergiu Toma 342d09c3e6 LINXD-2270: Removed console logs; Update commented code 2022-12-27 13:11:06 +02:00
Sergiu Toma b3409de3ba LINXD-2270: Allow server and web client to have full duplex 2022-12-19 19:31:33 +02:00
bmamihai 3e31ba21bd replace individual copy with copy all 2022-12-15 13:20:22 +02:00
bmamihai cdf02756d3 fix start 2022-12-14 01:32:17 +02:00
bmamihai b2f9f5affa remove pm2 and watchify 2022-12-14 00:48:14 +02:00
bmamihai 5b9bfeaa01 improve docker 2022-12-10 03:21:59 +02:00
bmamihai e3bef9b3e5 New dockerfile 2022-12-10 02:38:09 +02:00
Cristi Ene 8a9c370f02 Merge pull request 'LH-265-audio' (#17) from LH-265-audio into develop
Reviewed-on: #17
Reviewed-by: Cristi Ene <cristi.ene@safemobile.com>
2022-12-06 12:45:08 +00:00
Sergiu Toma 652019b07d LH-265: Update doc; Update bundle 2022-11-29 15:35:28 +02:00
Sergiu Toma 09c4a4b90e LH-265: Added audio on client and server 2022-11-29 14:19:02 +02:00
Bozieru Mihai 75d0e3aee7 exe right for build.sh 2022-10-31 22:26:08 +00:00
bmamihai 30ac997634 Merge pull request 'added build.sh' (#14) from temp-build into develop
Reviewed-on: #14
2022-10-31 22:22:20 +00:00
bmamihai 5aea138f6a added build.sh 2022-10-31 12:17:07 +02:00
Cristi Ene 5b01ddc2a8 Merge pull request 'LH-259-mediasoup-always-return-a-callback-response-to-clients' (#13) from LH-259-mediasoup-always-return-a-callback-response-to-clients into develop
Reviewed-on: #13
2022-10-25 16:18:54 +00:00
Sergiu Toma 084ff36ebe LH-259: Refactor createRoom 2022-10-24 22:38:06 +03:00
Sergiu Toma f4ebf92783 LH-259: Added callback from transport-produce 2022-10-24 22:35:22 +03:00
Sergiu Toma b59a157b18 LH-259: Comment callback from transport-produce 2022-10-24 22:19:37 +03:00
Sergiu Toma 9f8347bec5 LH-259: Update createRoom callback 2022-10-24 22:16:47 +03:00
Sergiu Toma 24390c98e5 LH-259: Added callbacks 2022-10-24 22:11:14 +03:00
Sergiu Toma 1a7371fe18 Parse RTC_MIN_PORT and RTC_MAX_PORT 2022-10-18 18:27:02 +03:00
Sergiu Toma be5f97762a Merge pull request 'LH-253: Added callId for transportclose and producerclose events' (#12) from LH-253-mediasoup-handle-callid-undefined into master
Reviewed-on: #12
2022-10-18 07:53:33 +00:00
Sergiu Toma 03a11126c4 LH-253: Check if we have callId in closeCall 2022-10-18 10:51:20 +03:00
Sergiu Toma fafbee6e4c LH-253: Added callId for transportclose and producerclose events 2022-10-18 02:05:22 +03:00
Sergiu Toma bbf23c33d4 Merge pull request 'LH-252: Update .env variables' (#11) from LH-252-mediasoup-add-a-config-file-with-keys-and-ports into master
Reviewed-on: #11
2022-10-09 06:50:45 +00:00
Sergiu Toma 5c2808e75a LH-252: Update .env variables 2022-10-06 15:21:54 +03:00
Sergiu Toma 2aea7497cc Merge pull request 'added log for dtls transport-connect' (#10) from LH-249-debug-for-i-os-dtls-problems into master
Reviewed-on: #10
2022-10-06 06:41:07 +00:00
21 changed files with 1742 additions and 1431 deletions

2
.dockerignore Normal file
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node_modules
doc

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.env
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PORT=3000
IP=0.0.0.0 # Listening IPv4 or IPv6.
ANNOUNCED_IP=185.8.154.190 # Announced IPv4 or IPv6 (useful when running mediasoup behind NAT with private IP).
RTC_MIN_PORT=2000
RTC_MAX_PORT=2020
SERVER_CERT="./server/ssl/cert.pem"
SERVER_KEY="./server/ssl/key.pem"

1
.gitignore vendored
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/node_modules
/dist

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@ -1,11 +1,25 @@
FROM ubuntu
FROM ubuntu:22.04
WORKDIR /app
RUN apt-get update && \
apt-get install -y build-essential pip net-tools iputils-ping iproute2 curl
RUN curl -fsSL https://deb.nodesource.com/setup_16.x | bash -
RUN curl -fsSL https://deb.nodesource.com/setup_18.x | bash -
RUN apt-get install -y nodejs
RUN npm install -g watchify
EXPOSE 3000
EXPOSE 2000-2020
EXPOSE 10000-10100
COPY . /app/
RUN npm install
EXPOSE 3000/tcp
EXPOSE 2000-2200/udp
CMD node app.js
#docker build -t linx-video .
# docker run -it -d --restart always -p 3000:3000/tcp -p 2000-2200:2000-2200/udp linx-video
#Run under host network
# docker run -it -d --network host --restart always -p 3000:3000/tcp -p 2000-2200:2000-2200/udp linx-video
#https://docs.docker.com/config/containers/resource_constraints/
#docker run -it -d --network host --cpus="0.25" --memory="512m" --restart always -p 3000:3000/tcp -p 2000-2200:2000-2200/udp linx-video

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@ -22,18 +22,20 @@
2. Run the `npm start:prod` command to start the server in production mode.
(To connect to the terminal, use `pm2 log video-server`)
---
### Web client
- The server will start by default on port 3000, and the ssl certificates will have to be configured
- The web client can be accessed using the /sfu path
ex: http://localhost:3000/sfu/?assetId=1&&accountId=1&producer=true&assetName=Adi&assetType=linx
ex: https://HOST/sfu/?assetId=1&&accountId=1&producer=true&dest_asset_id=75&assetName=Adi
assetId = asset id of the unit on which you are doing the test
accountId = account id of the unit on which you are doing the test
producer = it will always be true because you are the producer
(it's possible to put false, but then you have to have another client with producer true)
assetName = asset name of the unit on which you are doing the test
assetType = asset type of the unit on which you are doing the test
dest_asset_id= the addressee with whom the call is made
- To make a call using this client, you need a microphone and permission to use it
- For any changes related to the client, the command `npm run watch' will have to be used to generate the bundle.js used by the web client
### Demo project
The demo project used initially and then modified for our needs `https://github.com/jamalag/mediasoup2`

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app.js

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build.sh Executable file
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#/!bin/bash
## FUNCTIONS
function getGitVersion(){
version=$(git describe)
count=$(echo ${version%%-*} | grep -o "\." | wc -l)
if (( $count > 1 )); then
version=${version%%-*}
elif (( $count == 0 ));then
echo -e "Error: Git version \"${version%%-*}\" not respecting Safemobile standard.\n Must be like 4.xx or 4.xx.xx"
version="0.0.0"
else
if [[ "$1" == "dev" ]];then
cleanprefix=${version#*-} # remove everything before `-` including `-`
cleansuffix=${cleanprefix%-*} # remove everything after `-` including `-`
version="${version%%-*}.${cleansuffix}"
else
version="${version%%-*}.0" # one `%` remove everything after last `-`, two `%%` remove everything after all `-`
fi
fi
}
function addVersionPm2(){
file_pkg="package.json"
key=" \"version\": \""
if [ -f "$file_pkg" ] && [ ! -z "$version" ]; then
versionApp=" \"version\": \"$version\","
sed -i "s|^.*$key.*|${versionApp//\//\\/}|g" $file_pkg
text=$(cat $file_pkg | grep -c "$version")
if [ $text -eq 0 ]; then
echo "Version couldn't be set"
else
echo "Version $version successfully applied to App"
fi
fi
}
## PREBUILD PROCESS
# check dist dir to be present and empty
if [ ! -d "dist" ]; then
## MAKE DIR
mkdir "dist"
echo "Directory dist created."
else
## CLEANUP
rm -fr dist/*
fi
if [ -d "node_modules" ]; then
rm -fr node_modules
fi
# Install dependencies
#npm install
## PROJECT NEEDS
echo "Building app... from $(git rev-parse --abbrev-ref HEAD)"
#npm run-script build
cp -r {.env,app.js,package.json,server,public,doc,Dockerfile} dist/
#cp -r ./* dist/
# Generate Git log
dateString=$(date +"%Y%m%d-%H%M%S")
git log --pretty=format:"%ad%x09%an%x09%s" --no-merges -20 > "dist/git-$dateString.log"
# Get Git version control
getGitVersion $1
# Add version control for pm2
cd dist
addVersionPm2
## POST BUILD
cd -

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module.exports = {
hubAddress: 'https://hub.dev.linx.safemobile.com/',
mediasoupAddress: 'https://video.safemobile.org/mediasoup',
// mediasoupAddress: 'http://localhost:3000/mediasoup',
mediasoupAddress: 'https://testing.video.safemobile.org/',
}

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@ -34,6 +34,9 @@
<body>
<body>
<div id="video">
<legend>Client options:</legend>
<input type="checkbox" id="produceAudio" name="produceAudio">
<label for="produceAudio">Produce audio</label><br>
<table>
<thead>
<th>Local Video</th>
@ -43,12 +46,24 @@
<tr>
<td>
<div id="sharedBtns">
<video id="localVideo" autoplay class="video" ></video>
<video
id="localVideo"
class="video"
autoplay
muted
playsinline
></video>
</div>
</td>
<td>
<div id="sharedBtns">
<video id="remoteVideo" autoplay class="video" ></video>
<video
id="remoteVideo"
class="video"
autoplay
muted
playsinline
></video>
</div>
</td>
</tr>
@ -60,34 +75,11 @@
</td>
<td>
<div id="sharedBtns">
<button id="btnRecvSendTransport">Consume</button>
<!-- <button id="btnRecvSendTransport">Consume</button> -->
<button id="remoteSoundControl">Unmute</button>
</div>
</td>
</tr>
<!-- <tr>
<td colspan="2">
<div id="sharedBtns">
<button id="btnRtpCapabilities">2. Get Rtp Capabilities</button>
<br />
<button id="btnDevice">3. Create Device</button>
</div>
</td>
</tr>
<tr>
<td>
<div id="sharedBtns">
<button id="btnCreateSendTransport">4. Create Send Transport</button>
<br />
<button id="btnConnectSendTransport">5. Connect Send Transport & Produce</button></td>
</div>
<td>
<div id="sharedBtns">
<button id="btnRecvSendTransport">6. Create Recv Transport</button>
<br />
<button id="btnConnectRecvTransport">7. Connect Recv Transport & Consume</button>
</div>
</td>
</tr> -->
</tbody>
</table>
<div id="closeCallBtn">

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@ -10,147 +10,201 @@ const ASSET_NAME = urlParams.get('assetName') || null;
const ASSET_TYPE = urlParams.get('assetType') || null;
let callId = parseInt(urlParams.get('callId')) || null;
const IS_PRODUCER = urlParams.get('producer') === 'true' ? true : false
let remoteVideo = document.getElementById('remoteVideo')
remoteVideo.defaultMuted = true
let produceAudio = false
console.log('[URL] ASSET_ID', ASSET_ID, '| ACCOUNT_ID', ACCOUNT_ID, '| callId', callId, ' | IS_PRODUCER', IS_PRODUCER)
let socket
hub = io(config.hubAddress)
console.log('🟩 config', config)
const connectToMediasoup = () => {
socket = io(config.mediasoupAddress, {
reconnection: true,
reconnectionDelay: 1000,
reconnectionDelayMax : 5000,
reconnectionAttempts: Infinity
})
socket.on('connection-success', ({ _socketId, existsProducer }) => {
console.log(`[MEDIA] ${config.mediasoupAddress} | connected: ${socket.connected} | existsProducer: ${existsProducer}`)
if (!IS_PRODUCER && existsProducer && consumer === undefined) {
goConnect()
// document.getElementById('btnRecvSendTransport').click();
produceAudioSelector = document.getElementById('produceAudio');
produceAudioSelector.addEventListener('change', e => {
if(e.target.checked) {
produceAudio = true
console.log('produce audio');
} else {
produceAudio = false
}
if (IS_PRODUCER && urlParams.get('testing') === 'true') { getLocalStream() }
})
}
if (IS_PRODUCER === true) {
hub.on('connect', async () => {
console.log(`[HUB] ${config.hubAddress} | connected: ${hub.connected}`)
connectToMediasoup()
hub.emit(
'ars',
JSON.stringify({
ars: true,
asset_id: ASSET_ID,
account_id: ACCOUNT_ID,
})
)
hub.on('video', (data) => {
const parsedData = JSON.parse(data);
if (parsedData.type === 'notify-request') {
console.log('video', parsedData)
originAssetId = parsedData.origin_asset_id;
// originAssetName = parsedData.origin_asset_name;
// originAssetTypeName = parsedData.origin_asset_type_name;
callId = parsedData.video_call_id;
console.log('[VIDEO] notify-request | IS_PRODUCER', IS_PRODUCER, 'callId', callId);
getLocalStream()
}
if (parsedData.type === 'notify-end') {
console.log('[VIDEO] notify-end | IS_PRODUCER', IS_PRODUCER, 'callId', callId);
resetCallSettings()
}
})
})
hub.on('connect_error', (error) => {
console.log('connect_error', error);
});
hub.on('connection', () => {
console.log('connection')
})
hub.on('disconnect', () => {
console.log('disconnect')
})
} else {
connectToMediasoup()
}
});
let socket, hub
let device
let rtpCapabilities
let producerTransport
let consumerTransport
let producer
let producerVideo
let producerAudio
let consumer
let originAssetId
// let originAssetName = 'Adi'
// let originAssetTypeName = 'linx'
let consumerVideo // local consumer video(consumer not transport)
let consumerAudio // local consumer audio(consumer not transport)
const remoteSoundControl = document.getElementById('remoteSoundControl');
remoteSoundControl.addEventListener('click', function handleClick() {
console.log('remoteSoundControl.textContent', remoteSoundControl.textContent);
if (remoteSoundControl.textContent === 'Unmute') {
remoteVideo.muted = false
remoteSoundControl.textContent = 'Mute';
} else {
remoteVideo.muted = true
remoteSoundControl.textContent = 'Unmute';
}
});
// https://mediasoup.org/documentation/v3/mediasoup-client/api/#ProducerOptions
// https://mediasoup.org/documentation/v3/mediasoup-client/api/#transport-produce
let params = {
// mediasoup params
let videoParams = {
encodings: [
{
rid: 'r0',
maxBitrate: 100000,
scalabilityMode: 'S1T3',
},
{
rid: 'r1',
maxBitrate: 300000,
scalabilityMode: 'S1T3',
},
{
rid: 'r2',
maxBitrate: 900000,
scalabilityMode: 'S1T3',
},
{ scaleResolutionDownBy: 4, maxBitrate: 500000 },
{ scaleResolutionDownBy: 2, maxBitrate: 1000000 },
{ scaleResolutionDownBy: 1, maxBitrate: 5000000 },
{ scalabilityMode: 'S3T3_KEY' }
],
// https://mediasoup.org/documentation/v3/mediasoup-client/api/#ProducerCodecOptions
codecOptions: {
videoGoogleStartBitrate: 1000
}
}
const streamSuccess = (stream) => {
console.log('[streamSuccess]');
localVideo.srcObject = stream
const track = stream.getVideoTracks()[0]
params = {
track,
...params
let audioParams = {
codecOptions :
{
opusStereo : true,
opusDtx : true
}
}
setTimeout(() => {
hub = io(config.hubAddress)
const connectToMediasoup = () => {
socket = io(config.mediasoupAddress, {
reconnection: true,
reconnectionDelay: 1000,
reconnectionDelayMax : 5000,
reconnectionAttempts: Infinity
})
socket.on('connection-success', ({ _socketId, existsProducer }) => {
console.log(`[MEDIA] ${config.mediasoupAddress} | connected: ${socket.connected} | existsProducer: ${existsProducer}`)
if (!IS_PRODUCER && existsProducer && consumer === undefined) {
goConnect()
}
if (IS_PRODUCER && urlParams.get('testing') === 'true') { getLocalStream() }
})
socket.on('new-producer', ({ callId, kind }) => {
console.log(`🟢 new-producer | callId: ${callId} | kind: ${kind} | Ready to consume`);
connectRecvTransport();
})
socket.on('close-producer', ({ callId, kind }) => {
console.log(`🔴 close-producer | callId: ${callId} | kind: ${kind}`);
if (kind === 'video') {
consumerVideo.close()
remoteVideo.srcObject = null
}
else if (kind === 'audio') consumerAudio.close()
})
}
if (IS_PRODUCER === true) {
hub.on('connect', async () => {
console.log(`[HUB]! ${config.hubAddress} | connected: ${hub.connected}`)
connectToMediasoup()
hub.emit(
'ars',
JSON.stringify({
ars: true,
asset_id: ASSET_ID,
account_id: ACCOUNT_ID,
})
)
hub.on('video', (data) => {
const parsedData = JSON.parse(data);
if (parsedData.type === 'notify-request') {
console.log('video', parsedData)
originAssetId = parsedData.origin_asset_id;
// originAssetName = parsedData.origin_asset_name;
// originAssetTypeName = parsedData.origin_asset_type_name;
callId = parsedData.video_call_id;
console.log('[VIDEO] notify-request | IS_PRODUCER', IS_PRODUCER, 'callId', callId);
getLocalStream()
}
if (parsedData.type === 'notify-end') {
console.log('[VIDEO] notify-end | IS_PRODUCER', IS_PRODUCER, 'callId', callId);
resetCallSettings()
}
})
})
hub.on('connect_error', (error) => {
console.log('connect_error', error);
});
hub.on('connection', () => {
console.log('connection')
})
hub.on('disconnect', () => {
console.log('disconnect')
})
} else {
connectToMediasoup()
}
}, 1600);
const streamSuccess = (stream) => {
console.log('[streamSuccess] device', device);
localVideo.srcObject = stream
console.log('stream', stream);
const videoTrack = stream.getVideoTracks()[0]
const audioTrack = stream.getAudioTracks()[0]
videoParams = {
track: videoTrack,
...videoParams
}
audioParams = {
track: audioTrack,
...audioParams
}
console.log('[streamSuccess] videoParams', videoParams, ' | audioParams', audioParams);
goConnect()
}
const getLocalStream = () => {
console.log('[getLocalStream]');
navigator.mediaDevices.getUserMedia({
audio: false,
audio: produceAudio ? true : false,
video: {
width: {
min: 640,
max: 1920,
},
height: {
min: 400,
max: 1080,
}
qvga : { width: { ideal: 320 }, height: { ideal: 240 } },
vga : { width: { ideal: 640 }, height: { ideal: 480 } },
hd : { width: { ideal: 1280 }, height: { ideal: 720 } }
}
})
.then(streamSuccess)
.catch(error => {
console.log(error.message)
})
navigator.permissions.query(
{ name: 'microphone' }
).then((permissionStatus) =>{
console.log('🟨 [PERMISSION] permissionStatus', permissionStatus); // granted, denied, prompt
// It will block the code from execution and display "Permission denied" if we don't have microphone permissions
})
}
const goConnect = () => {
@ -167,7 +221,6 @@ const goCreateTransport = () => {
// server side to send/recive media
const createDevice = async () => {
try {
console.log('[createDevice]');
device = new mediasoupClient.Device()
// https://mediasoup.org/documentation/v3/mediasoup-client/api/#device-load
@ -178,7 +231,8 @@ const createDevice = async () => {
})
console.log('Device RTP Capabilities', device.rtpCapabilities)
console.log('[createDevice] device', device);
// once the device loads, create transport
goCreateTransport()
@ -207,18 +261,20 @@ const getRtpCapabilities = () => {
}
const createSendTransport = () => {
console.log('[createSendTransport');
// see server's socket.on('createWebRtcTransport', sender?, ...)
// this is a call from Producer, so sender = true
socket.emit('createWebRtcTransport', { sender: true, callId }, ({ params }) => {
socket.emit('createWebRtcTransport', { sender: true }, (value) => {
console.log(`[createWebRtcTransport] value: ${JSON.stringify(value)}`);
const params = value.params;
// The server sends back params needed
// to create Send Transport on the client side
if (params.error) {
console.log(params.error)
return
}
console.log(params)
// creates a new WebRTC Transport to send media
// based on the server's producer transport params
// https://mediasoup.org/documentation/v3/mediasoup-client/api/#TransportOptions
@ -244,10 +300,10 @@ const createSendTransport = () => {
})
producerTransport.on('produce', async (parameters, callback, errback) => {
console.log(parameters)
console.log('[produce] parameters', parameters)
try {
// tell the server to create a Producer
// Tell the server to create a Producer
// with the following parameters and produce
// and expect back a server side producer id
// see server's socket.on('transport-produce', ...)
@ -270,21 +326,45 @@ const createSendTransport = () => {
}
const connectSendTransport = async () => {
// we now call produce() to instruct the producer transport
console.log('[connectSendTransport] producerTransport');
// We now call produce() to instruct the producer transport
// to send media to the Router
// https://mediasoup.org/documentation/v3/mediasoup-client/api/#transport-produce
// this action will trigger the 'connect' and 'produce' events above
producer = await producerTransport.produce(params)
// Produce video
let producerVideoHandler = await producerTransport.produce(videoParams)
console.log('videoParams', videoParams);
console.log('producerVideo', producerVideo);
producer.on('trackended', () => {
producerVideoHandler.on('trackended', () => {
console.log('track ended')
// close video track
})
})
producer.on('transportclose', () => {
producerVideoHandler.on('transportclose', () => {
console.log('transport ended')
// close video track
})
})
// Produce audio
if (produceAudio) {
let producerAudioHandler = await producerTransport.produce(audioParams)
console.log('audioParams', audioParams);
console.log('producerAudio', producerAudio);
producerAudioHandler.on('trackended', () => {
console.log('track ended')
// close audio track
})
producerAudioHandler.on('transportclose', () => {
console.log('transport ended')
// close audio track
})
}
const answer = {
origin_asset_id: ASSET_ID,
@ -294,7 +374,7 @@ const connectSendTransport = async () => {
origin_asset_type_name: ASSET_TYPE,
origin_asset_name: ASSET_NAME,
video_call_id: callId,
answer: 'accepted', // answer: 'rejected'
answer: 'accepted', // answer: accepted/rejected
};
console.log('SEND answer', answer);
@ -306,11 +386,13 @@ const connectSendTransport = async () => {
// Enable Close call button
const closeCallBtn = document.getElementById('btnCloseCall');
closeCallBtn.removeAttribute('disabled');
createRecvTransport();
}
const createRecvTransport = async () => {
console.log('createRecvTransport');
// see server's socket.on('consume', sender?, ...)
// See server's socket.on('consume', sender?, ...)
// this is a call from Consumer, so sender = false
await socket.emit('createWebRtcTransport', { sender: false, callId }, ({ params }) => {
// The server sends back params needed
@ -320,15 +402,15 @@ const createRecvTransport = async () => {
return
}
console.log(params)
console.log('[createRecvTransport] params', params)
// creates a new WebRTC Transport to receive media
// Creates a new WebRTC Transport to receive media
// based on server's consumer transport params
// https://mediasoup.org/documentation/v3/mediasoup-client/api/#device-createRecvTransport
consumerTransport = device.createRecvTransport(params)
// https://mediasoup.org/documentation/v3/communication-between-client-and-server/#producing-media
// this event is raised when a first call to transport.produce() is made
// This event is raised when a first call to transport.produce() is made
// see connectRecvTransport() below
consumerTransport.on('connect', async ({ dtlsParameters }, callback, errback) => {
try {
@ -345,7 +427,8 @@ const createRecvTransport = async () => {
errback(error)
}
})
connectRecvTransport()
// We call it in new-rpoducer, we don't need it here anymore
// connectRecvTransport()
})
}
@ -353,7 +436,8 @@ const resetCallSettings = () => {
localVideo.srcObject = null
remoteVideo.srcObject = null
consumer = null
producer = null
producerVideo = null
producerAudio = null
producerTransport = null
consumerTransport = null
device = undefined
@ -361,40 +445,97 @@ const resetCallSettings = () => {
const connectRecvTransport = async () => {
console.log('connectRecvTransport');
// for consumer, we need to tell the server first
// For consumer, we need to tell the server first
// to create a consumer based on the rtpCapabilities and consume
// if the router can consume, it will send back a set of params as below
await socket.emit('consume', {
rtpCapabilities: device.rtpCapabilities,
callId
}, async ({ params }) => {
if (params.error) {
console.log('Cannot Consume')
return
}
// then consume with the local consumer transport
// which creates a consumer
consumer = await consumerTransport.consume({
id: params.id,
producerId: params.producerId,
kind: params.kind,
rtpParameters: params.rtpParameters
})
// destructure and retrieve the video track from the producer
const { track } = consumer
}, async ({videoParams, audioParams}) => {
console.log(`[consume] 🟩 videoParams`, videoParams)
console.log(`[consume] 🟩 audioParams`, audioParams)
console.log('[consume] 🟩 consumerTransport', consumerTransport)
let stream = new MediaStream()
stream.addTrack(track)
// stream.removeTrack(track)
remoteVideo.srcObject = stream
socket.emit('consumer-resume')
console.log('consumer', consumer);
// Maybe the unit does not produce video or audio, so we must only consume what is produced
if (videoParams) {
console.log('❗ Have VIDEO stream to consume');
stream.addTrack(await getVideoTrask(videoParams))
} else {
console.log('❗ Don\'t have VIDEO stream to consume');
}
if (audioParams) {
console.log('❗ Have AUDIO stream to consume');
let audioTrack = await getAudioTrask(audioParams)
stream.addTrack(audioTrack)
} else {
console.log('❗ Don\'t have AUDIO stream to consume');
}
socket.emit('consumer-resume')
remoteVideo.srcObject = stream
remoteVideo.setAttribute('autoplay', true)
remoteVideo.play()
.then(() => {
console.log('remoteVideo PLAY')
})
.catch((error) => {
console.error(`remoteVideo PLAY ERROR | ${error.message}`)
})
})
}
const getVideoTrask = async (videoParams) => {
consumerVideo = await consumerTransport.consume({
id: videoParams.id,
producerId: videoParams.producerId,
kind: videoParams.kind,
rtpParameters: videoParams.rtpParameters
})
return consumerVideo.track
}
const getAudioTrask = async (audioParams) => {
consumerAudio = await consumerTransport.consume({
id: audioParams.id,
producerId: audioParams.producerId,
kind: audioParams.kind,
rtpParameters: audioParams.rtpParameters
})
consumerAudio.on('transportclose', () => {
console.log('transport closed so consumer closed')
})
const audioTrack = consumerAudio.track
audioTrack.applyConstraints({
audio: {
advanced: [
{
echoCancellation: {exact: true}
},
{
autoGainControl: {exact: true}
},
{
noiseSuppression: {exact: true}
},
{
highpassFilter: {exact: true}
}
]
}
})
return audioTrack
}
const closeCall = () => {
console.log('closeCall');
@ -416,6 +557,30 @@ const closeCall = () => {
resetCallSettings()
}
// const consume = async (kind) => {
// console.log(`[consume] kind: ${kind}`)
// console.log('createRecvTransport Consumer')
// await socket.emit('createWebRtcTransport', { sender: false, callId, dispatcher: true }, ({ params }) => {
// if (params.error) {
// console.log('createRecvTransport | createWebRtcTransport | Error', params.error)
// return
// }
// consumerTransport = device.createRecvTransport(params)
// consumerTransport.on('connect', async ({ dtlsParameters }, callback, errback) => {
// try {
// await socket.emit('transport-recv-connect', {
// dtlsParameters,
// })
// callback()
// } catch (error) {
// errback(error)
// }
// })
// connectRecvTransport()
// })
// }
btnLocalVideo.addEventListener('click', getLocalStream)
btnRecvSendTransport.addEventListener('click', goConnect)
// btnRecvSendTransport.addEventListener('click', consume)
btnCloseCall.addEventListener('click', closeCall)