rtc-io / rtc Goto Github PK

The github README says to just install rtc module with npm. But node is complaining it can't find rtc-media module.

The rtc-media module has 3 errors.

Error: Cannot find module 'rtc-media'
at Function.Module._resolveFilename (module.js:440:15)
at Function.Module._load (module.js:388:25)
at Module.require (module.js:468:17)
at require (internal/module.js:20:19)
at Object. (/private/var/www/eyehawk.io/index.js:4:16)
at Module._compile (module.js:541:32)
at Object.Module._extensions..js (module.js:550:10)
at Module.load (module.js:458:32)
at tryModuleLoad (module.js:417:12)
at Function.Module._load (module.js:409:3)

We're using rtc.io in http://flowhub.io with http://noflojs.org to allow remote/live programming/debugging on devices from our webapp. Would like to use the same protocol for running automated test from a node.js CLI tool.

We're experimenting with the wrtc node.js library to support this. Would you be open to have support for the rtc.io offer-exchange for node.js clients in rtc itself? Or is it something that should stay separately?

I want to create a room based live video streaming website with integrated chat. Are there any modules in rtc.io that could help my purpose.

When I am implementing

http://rtc.io/tutorial-rtc-video-chat.html with my own local singanalling server as https://switchboard.rtc.io is probably down. I am getting error

TypeError: Failed to execute 'appendChild' on 'Node': parameter 1 is not of type 'Node'.

When doing localVideo.appendChild(rtc.local);

And also I am not able to call the init method on rtc

Thank you for these open source tools.
The switchboard reference implementation is in-memory not designed to be deployed at scale; can you recommend a signalling server for use at scale with the rest of the rtc stack?

Running a simple tests which creates two datachannels and sends between them shows:

Error: global leaks detected: Primus, RTCSessionDescription, RTCIceCandidate, isValid

ran using mocha.checkLeaks()

we wanted build a classroom app which only teacher can speak and student can only listen, is there a way to take advantage of webRTC and rtc.io to achieve this?

the only solution i can think of is: make nodejs server and teacher as Peer, and nodeJS server send video stream to students via websocket, which is not really P2P

I'd like to get this sorted before @silviapfeiffer's presentation today, as after that I think we should lock it in stone. My thoughts:

• The ALLCAPS for globals seems to be an "artifact of the crockfordian era", which I believe we are coming to an end of. After working in the liberated lands of effective JS module systems it feels a little clunky, especially when exposing other internals (i.e. RTC.quickconnect).
• Mixed caps is definitely used by some libraries (Backbone, Ember, et al) so could be an option.
• All lowercaps feels comfortable to me, and is used by more progressive libraries (such as angular, d3, etc).

I'd like to do a quick straw poll and get some additional input before locking this in...

The rtc package should definitely move to using rtc-captureconfig version 2. This can be achieved pretty easily by running:

npm install rtc-captureconfig@2 --save

And then doing a rebuild, minor version bump and publish...

The data channel example shown in the tutorial http://rtc.io/tutorial-rtc-text-chat.html has some inaccurate code which causes wrong example behaviour in case of 3 or more nodes in the chat. Particularly, when the second channel is created, the message.onkeyup function which was initialized before, is reinitialized by the just created function:

    messages.onkeyup = function () {
        channel.send(this.innerHTML);
    };

Such a bug in the tutorial makes imaging of your RTC package worse.

I suggest the following a little bit more precise code to replace a tutorial code. This code creates a channel registry which is updated by two different session channel-related events, and allows to create as many nodes in the chat, as simultaneous channels allowed in the RTC implementation.

    // Set RTC options
    var rtcOpts = {
        room: 'test-data',
        signaller: '//switchboard.rtc.io',
        capture: false,
    };

    // call RTC module
    var rtc = RTC(rtcOpts);

    // A contenteditable element to show our messages
    var messages = document.getElementById('messages');

    // Channels registry
    var channels = {};

    // Send message to every registered channel
    messages.onkeyup = function () {
        for(var i in channels) {
            channel = channels[i];
            channel.send(messages.innerHTML);
        }
    };

    // Bind to events happening on the data channel
    // and fill the channel registry
    function bindDataChannelEvents(id, channel, attributes, connection) {
        channels[id] = channel;

        // Receive message
        channel.onmessage = function (evt) {
            messages.innerHTML = evt.data;
        };
    }

    // Clean channel registry for closed channels
    function unbindDataChannelEvents(id, channel, name) {
        delete channels[id];
    }

    // Start working with the established session
    function init(session) {
        session.createDataChannel('chat', {
            ordered: true,
            maxRetransmits: 12
        });
        // Bind to important session events
        session.on('channel:opened:chat', bindDataChannelEvents);
        session.on('channel:closed:chat', unbindDataChannelEvents);
    }

    // Detect when RTC has established a session
    rtc.on('ready', init);

Hello RTC Team,
Is there any application / module that was developed for remotely controlling the mouse and keyboard strokes using RTC?

regards
Anish nema

In Firefox JavaScript Console there are messages like:

RTCIceServer.url is deprecated! Use urls instead.

And

Mandatory/optional in createOffer options is deprecated! Use {"offerToReceiveAudio":true,"offerToReceiveVideo":true} instead (note the case difference)!

The messages don't really bother me, but I suppose they might be dropped at some point. So should these be updated?

Does it supports Ios 64-bit support?

/private/var/www/eyehawk.io/node_modules/rtc-capture/index.js:5
navigator.getUserMedia = navigator.getUserMedia ||
^

ReferenceError: navigator is not defined
at Object. (/private/var/www/eyehawk.io/node_modules/rtc-capture/index.js:5:1)
at Module._compile (module.js:541:32)
at Object.Module._extensions..js (module.js:550:10)
at Module.load (module.js:458:32)
at tryModuleLoad (module.js:417:12)
at Function.Module._load (module.js:409:3)
at Module.require (module.js:468:17)
at require (internal/module.js:20:19)
at Object. (/private/var/www/eyehawk.io/node_modules/rtc-media/index.js:1:77)
at Module._compile (module.js:541:32)

Hello,

I have the following use-case I would like to implement:

• A set of users (e.g. 5 users) are conferencing in a room.
• For each user, I want to capture their video stream on the server and do processing (e.g. machine-learning analysis).

I can't find anything in the other rtc modules for allowing this. One idea I had was to set up another server that would be another "client" for the main rtc server, meaning that each room would have an extra client connected that is the video processing server. However, I'm not sure how that would work given that the modules were written for browser-based clients.

Thank you very much!

When the capture option is specified as a string, then this line fails. Replacing the last part with something like:

[].concat(capture || [])

should fix the problem...

As discussed internally, I'd like to deprecate rtc-media in favour of more concise, targeted implementations. Both the rtc-capture and rtc-attach modules are now functionally complete so I'd like to look at performing a migration to these modules and at the same time bump the major of rtc-quickconnect to 3.x

Happy to do this work unless there are other takers, and if not, I'll schedule it in the in the next few days.

https://www.npmjs.org/package/rtc says "bowser install rtc-io/rtc" which I think refers to "bower install rtc-io/rtc" (no Mario character required). But when I do so I get :

$ bower install rtc-io/rtc
bower rtc#* cached git://github.com/rtc-io/rtc.git#3.1.2
bower rtc#* validate 3.1.2 against git://github.com/rtc-io/rtc.git#*
bower rtc#~3.1.2 ENOTFOUND Package rtc not found

Is there something wrong with bower.json? (noob remark : could the newline removed in commit 90fd373 be the cause ?)

It looks like CDNjs is going through a little bit of a transition process right now with regards to it's autoupdate process (see cdnjs/cdnjs#3638), but once this is stable we should definitely get the various dist resources publishing.

Not urgent, just something to do :)

https://www.pkcsecurity.com/blog#webrtc-flow

This is a great resource, just wanted to share it

Seems to be a little bit of confusion around the fact that rtc can be used for dc only comms. While I'd recommend using rtc-quickconnect in this case, probably worth including an example that demonstrated the capture: false case.

Folks, I am trying to run the simple example in the rtc.io website https://rtc.io/tutorial-rtc-video-chat.html

I am getting the following error: Uncaught TypeError: Failed to execute 'appendChild' on 'Node': parameter 1 is not of type 'Node'.

And the problem is that acctually the rtc object does not have the remote and local attributes:

// Display local and remote video streams
localVideo.appendChild(rtc.local);
remoteVideo.appendChild(rtc.remote);

I used the last RTC release 3.3.0. Am I missing something here?

I have an extremely simple example and got following errors:

https://drive.google.com/file/d/16ICY28UJQM19veagpqJe4j_-Sr26zEeZ/view

Uncaught Error: Failed to execute 'createObjectURL' on 'URL': No function was found that matched the signature provided.TypeError: Failed to execute 'createObjectURL' on 'URL': No function was found that matched the signature provided. at attachToElement (bundle.js:30799) at module.exports (bundle.js:30832) at Step.attach.local (bundle.js:30849) at addKgoTask (bundle.js:30449) at localVideo (bundle.js:60) at createConference (bundle.js:53) at Object.2../config.js (bundle.js:89) at s (bundle.js:1) at e (bundle.js:1) at bundle.js:1 attachToElement @ bundle.js:30799 module.exports @ bundle.js:30832 attach.local @ bundle.js:30849 __kgoRunStep__ @ bundle.js:30556 runTask @ bundle.js:30638 run @ bundle.js:30649 (anonymous) @ bundle.js:30671 (anonymous) @ bundle.js:30634 Step.done @ bundle.js:30566 complete @ bundle.js:30548 handleCapture @ bundle.js:31065 bundle.js:35344 WebSocket connection to 'wss://switchboard.rtc.io/' failed: Error in connection establishment: net::ERR_CONNECTION_TIMED_OUT attemptNext @ bundle.js:35344 connect @ bundle.js:35406 signaller.connect @ bundle.js:33075 module.exports @ bundle.js:33212 module.exports @ bundle.js:32957 module.exports @ bundle.js:31210 createConference @ bundle.js:33 2../config.js @ bundle.js:89 s @ bundle.js:1 e @ bundle.js:1 (anonymous) @ bundle.js:1 bundle.js:35344 WebSocket connection to 'wss://switchboard.rtc.io/primus' failed: Error in connection establishment: net::ERR_CONNECTION_TIMED_OUT attemptNext @ bundle.js:35344

Does it relay streams ?
My goal is to create some video chat application with 20 users: and webrtc seems to have limits to 6 beacuse of its peer2peer technology.