1. It can use either. By default, preference is given to UDP, but depending on the firewall(s) in between the peers connecting it may only be able to connect with TCP. You can use [Wireshark](http://www.wireshark.org/) to capture packets and verify whether TCP or UDP is being used. In Chrome you can also see details on the selected candidate (`googActiveConnection`) by going to `chrome://webrtc-internals`.

2. &quot;Reliability mode&quot; probably refers to the reliability mode of the [DataChannel](http://www.w3.org/TR/2013/WD-webrtc-20130910/#rtcdatachannel), which can be configured to run in reliable or unreliable mode. DTLS refers to the currently optional, but [soon to be default method](http://blog.vline.com/post/57544771647/webrtc-digest-week-of-7-29-ietf-berlin-and-vp8) of exchanging encryption keys (the other deprecated mode is SDES). Firefox only supports DTLS, so for browser interop, you&#39;ll currently need to [enable it in Chrome](http://www.webrtc.org/interop).

3. The RTCPeerConnection (media) will use TCP or UDP, while the DataChannel uses SCTP. The SCTP implementation used by Firefox is implemented on top of UDP: https://code.google.com/p/sctp-refimpl/.

4. It&#39;s possible to filter out TCP or UDP ICE candidates before adding them with [`addIceCandidate`](http://www.w3.org/TR/2013/WD-webrtc-20130910/#widl-RTCPeerConnection-addIceCandidate-void-RTCIceCandidate-candidate-VoidFunction-successCallback-RTCPeerConnectionErrorCallback-failureCallback). Generally, you should not try to force the transport used since WebRTC will just &quot;do the right thing&quot;. The browser does not limit the number of TCP connections used by WebRTC beyond any limit on the RTCPeerConnection or DataChannel (i.e., if you can have 10 PeerConnections, they can each use TCP without any problem).

Just a quick question:  
How do you clear the screen in shell?
I&#39;ve seen ways like:

    import os
    os.system(&#39;cls&#39;)

This just opens the windows cmd, clears the screen and closes but
I want the shell window to be cleared  
(PS: I don&#39;t know this helps, but I&#39;m using version 3.3.2 of Python)  
Thank you :)

Clear screen in shell

I have cart in my top menu, using `push-right` and my problem is that the dropdown bubble falls out of the page. I am trying to align the bubble right, so that it right-aligns with &#39;click&#39;

like this 

![enter image description here][1]

[1]: http://i.stack.imgur.com/MtT4f.png


HTML:

&lt;!-- language: lang-html --&gt;
   
    &lt;div class=&quot;cart pull-right&quot;&gt;
      &lt;ul class=&quot;nav&quot;&gt;
        &lt;li class=&quot;dropdown&quot;&gt;
          &lt;a class=&quot;dropdown-toggle&quot; data-toggle=&quot;dropdown&quot; href=&quot;#&quot;&gt;CLICK&lt;/a&gt;
           &lt;div class=&quot;dropdown-menu&quot;&gt;
           STUFF
           &lt;/div&gt;
        &lt;/li&gt;
      &lt;/ul&gt;
    &lt;/div&gt;

bootstrap dropdown bubble align right (not push-right)

This sounds like a very basic question, but I need a confirmation

 1. Does WebRTC use TCP or UDP as its peer-to-peer transport? How do I
    know ?
 2. I read that there are reliability mode and DTLS agreement, how does
    they affect?
 3. Is this transport the same for both Media and DataChannel?
 4. How do I switch between TCP and UDP?

I ask this because I know that browsers have a limit on the number of parallel connections  (I think they talk about TCP), and maybe UDP connection is not limited.

Does WebRTC use TCP or UDP?

<p>This sounds like a very basic question, but I need a confirmation</p>
<ol>
<li>Does WebRTC use TCP or UDP as its peer-to-peer transport? How do I
know ?</li>
<li>I read that there are reliability mode and DTLS agreement, how does
they affect?</li>
<li>Is this transport the same for both Media and DataChannel?</li>
<li>How do I switch between TCP and UDP?</li>
</ol>
<p>I ask this because I know that browsers have a limit on the number of parallel connections  (I think they talk about TCP), and maybe UDP connection is not limited.</p>



I was trying to use adb over TCP/IP. I have followed these steps

    adb tcpip 5555
    
    adb connect 194.68.0.100:5555


I have used my device for 2 days and now I am unable to connect to my IP address like when I do

    adb tcpip 5555

it doesn&#39;t respond anything. Anyone knows what could be the scenario.


adb connection over tcp not working now

I have installed and web application which is running on port 8080 on RHEL (centOS). I only have command line access to that machine. I have tried to access that application from my windows machine from which I am connected to server via command-line, but it is giving connection time out error.

Then I have tried to open port 8080. I have added following entry into the iptables.

&gt;-A INPUT -m state --state NEW -m tcp -p tcp --dport 8080 -j ACCEPT

After adding this into the iptables I have restarted it with - 
`/etc/init.d/iptables restart`

But still I am not able to access that application from my windows machine.

Am I doing any mistake or missing something?

How to open port in Linux

I&#39;m having trouble wrapping my head around the `pipe` function shown in several Node.js examples for the [net][1] module.

    var net = require(&#39;net&#39;);
    
    var server = net.createServer(function (socket) {
      socket.write(&#39;Echo server\r\n&#39;);
      socket.pipe(socket);
    });

Can anyone offer an explanation on how this works and why it&#39;s required?


  [1]: http://nodejs.org/api/net.html

Using pipe() in Node.js net

WebSockets [have the option][1] of sending pings to the other end, where the other end is supposed to respond with a pong.

&gt; Upon receipt of a Ping frame, an endpoint MUST send a Pong frame in
   response, unless it already received a Close frame.  It SHOULD
   respond with Pong frame as soon as is practical.

TCP [offers something similar][2] in the form of keepalive:

&gt;  [Y]ou send your peer a keepalive probe packet with no data in it and the ACK flag turned on. You can do this because of the TCP/IP specifications, as a sort of duplicate ACK, and the remote endpoint will have no arguments, as TCP is a stream-oriented protocol. On the other hand, you will receive a reply from the remote host (which doesn&#39;t need to support keepalive at all, just TCP/IP), with no data and the ACK set.

I would think that TCP keepalive is more efficient, because it can be handled within the kernel without the need to transfer data up to user space, parse a websocket frame, craft a response frame, and hand that back to the kernel for transmission. It&#39;s also less network traffic.

Furthermore, WebSockets [are explicitly specified][3] to always run over TCP; they&#39;re not transport-layer agnostic, so TCP keepalive is always available:

&gt; The WebSocket Protocol is an independent TCP-based protocol.

So why would one ever want to use WebSocket ping/pong instead of TCP keepalive?

  [1]: https://www.rfc-editor.org/rfc/rfc6455#section-5.5.2
  [2]: http://tldp.org/HOWTO/TCP-Keepalive-HOWTO/overview.html
  [3]: https://www.rfc-editor.org/rfc/rfc6455#section-1.7

WebSockets ping/pong, why not TCP keepalive?

Is it possible to show the `adb logcat` for a `tcpip` device in Android Studio? I am writing a application that utilizes the USB port and I cannot use it for `adb usb`.

Currently, I am using the following instruction to view the logcat in Command Prompt, but it would be nice if this was integrated into AS:
http://developer.android.com/tools/help/adb.html#wireless

Use adb over tcpip in Android Studio?

For some reason I am having a hard time sending and receiving data from the same socket. Anyways here is my client code:


    var client = new UdpClient();
    IPEndPoint ep = new IPEndPoint(IPAddress.Parse(&quot;127.0.0.1&quot;), 11000); // endpoint where server is listening (testing localy)
    client.Connect(ep); 

    // send data
    client.Send(new byte[] { 1, 2, 3, 4, 5 }, 5);

    // then receive data
    var receivedData = client.Receive(ref ep);  // Exception: An existing connection was forcibly closed by the remote host

**Basically I want to create a protocol where I send a udp packet and then I expect a response. Just like the HTTP protocol for every request there is a response.** This code works if the server is on a different computer. There might be the case where the server and client are on the same computer though.

Here is the server:

    UdpClient udpServer = new UdpClient(UDP_LISTEN_PORT);
            
    while (true)
    {
        var groupEP = new IPEndPoint(IPAddress.Any, 11000); // listen on any port
        var data = udpServer.Receive(ref groupEP);
        udpServer.Send(new byte[] { 1 }, 1); // if data is received reply letting the client know that we got his data          
    }


-----------------------------------

Edit 
---

the reason why I cannot use tcp is because sometimes the client is behind a NAT (router) and it is simpler to do UDP hole punching than TCP.



-------
Solution:
----

thanks to markmnl answer here is my code:

Server:

    UdpClient udpServer = new UdpClient(11000);

    while (true)
    {
        var remoteEP = new IPEndPoint(IPAddress.Any, 11000); 
        var data = udpServer.Receive(ref remoteEP); // listen on port 11000
        Console.Write(&quot;receive data from &quot; + remoteEP.ToString());
        udpServer.Send(new byte[] { 1 }, 1, remoteEP); // reply back
    }


Client code:

    var client = new UdpClient();
    IPEndPoint ep = new IPEndPoint(IPAddress.Parse(&quot;127.0.0.1&quot;), 11000); // endpoint where server is listening
    client.Connect(ep);

    // send data
    client.Send(new byte[] { 1, 2, 3, 4, 5 }, 5);

    // then receive data
    var receivedData = client.Receive(ref ep);

    Console.Write(&quot;receive data from &quot; + ep.ToString());

    Console.Read();



Simple UDP example to send and receive data from same socket

How do I forward a UDP port from my Docker container to the host machine?

How do I publish a UDP Port on Docker?

I newly wrote a simple chat application, but I didn&#39;t really understand the background of ICE Candidates.

When the peer create a connection they get ICE Candidates and they exchange them and set
them finally to the peerconnection.

So my question is, where do the ICE Candidates come from and how are they used and are they all really used ?

I have noticed that my colleague got less candidates when he executes the application on his machine, what could be the reason for different amount of Candidates ?


What are ICE Candidates and how do the peer connection choose between them?

I am testing the WebRTC AGC but I must be doing something wrong because the signal just passes through unmodified.

Here&#39;s how I create and initialize the AGC:

    agcConfig.compressionGaindB = 9;
    agcConfig.limiterEnable = 1;
    agcConfig.targetLevelDbfs = 9;   /* 9dB below full scale */

    WebRtcAgc_Create(&amp;agc);
    WebRtcAgc_Init(agc, minLevel, maxLevel, kAgcModeFixedDigital, 8000);
    WebRtcAgc_set_config(agc, agcConfig);

And then for each 10ms sample block I do the following:

    WebRtcAgc_Process(agc, micData, NULL, 80, micData, NULL, micLevelIn, &amp;micLevelOut, 0, &amp;saturationWarning);

Where `micLevelIn` is set to 0. 

Can somebody tell me what I&#39;m doing wrong? 

I expected that a full scale sine tone would be attenuated to the target DBFS level; and a low level sine tone (i.e. -30dBFS) would be amplified to match the target DBFS level. But that&#39;s not what I&#39;m seeing.

WebRTC AGC (Automatic Gain Control)

How to use a &quot;custom&quot; video capturer to create a VideoTrack and provide frames?

The classic approach to create a VideoTrack is:

**1 - Get a VideoCapturer instance**

    VideoCapturer capturer = VideoCapturer.create(name);
**2 - Create a VideoSource**

    VideoSource videoSource = peerconnectionFactory.createVideoSource(capturer,	videoConstraints);

**3 - Create a VideoTrack using the video source**

    VideoTrack videoTrack = peerconnectionFactory.createVideoTrack(&quot;Label&quot;, videoSource);

**4 - Add the track to the MediaStream**
___

I was wondering if there is a way to change **step one**. Instead of using the native Capturer instance, use an Android one, and provide the frames to the VideoTrack using the callback:

    public void onPreviewFrame(byte[] data, Camera camera) {
       // provide the frames to the VideoTrack
    }

Any suggestions? 

Create a WebRTC VideoTrack with a &quot;custom&quot; Capturer on Android with libjingle

I am capturing audio data using `getUserMedia()` and I want to send it to my server so I can save it as a Blob in a MySQL field.

This is all I am trying to do. I have made several attempts to do this using WebRTC, but I don&#39;t even know at this point if this is right or even the best way to do this.

Can *anybody* help me?

Here is the code I am using to capture audio from the microphone:

	navigator.getUserMedia({
		video:false,
		audio:true,
	},function(mediaStream){
		
		// output mediaStream to speakers:
		var mediaStreamSource=audioContext.createMediaStreamSource(mediaStream);
		mediaStreamSource.connect(audioContext.destintion);
		
		// send mediaStream to server:
		
        // WebRTC code? not sure about this...
        var RTCconfig={};
        var conn=new RTCPeerConnection(RTCconfig);
        
		// ???
		
	},function(error){
		console.log(&#39;getUserMedia() fail.&#39;);
		console.log(error);
	});

How can I send this `mediaStream` up to the server?

After Googling around I&#39;ve been looking into `WebRTC`, *but this seems to be for just peer to peer communication* - actually, now I&#39;m looking into this more, I think this is the way to go. It seems to be the way to communicate from the client&#39;s browser up to the host webserver, but nothing I try even comes close to working.

I&#39;ve been going through [the W3C documentation][1] (which I am finding way too abstract), and I&#39;ve been going thru [this article on HTML5 Rocks][2] (which is bringing up more questions than answers). Apparently I need a signalling method, can anyone advise which signalling method is best for sending mediaStreams, XHR, XMPP, SIP, Socket.io or something else?

What will I need on the server to support the receiving of WebRTC? My web server is running a basic LAMP stack.

Also, is it best to wait until the `mediaStream` is finished recording before I send it up to the server, or is it better to send the `mediaStream` as its being recorded? I want to know if I am going about doing this the right way. I have written file uploaders in javascript and HTML5, but uploading one of these `mediaStreams` seems hellishly more complicated and I&#39;m not sure if I am approaching it right.

Any help on this would be greatly appreciated.

  [1]: http://www.w3.org/TR/webrtc/
  [2]: http://www.html5rocks.com/en/tutorials/webrtc/basics/


Sending a MediaStream to host Server with WebRTC after it is captured by getUserMedia

I am trying to figure out how to test whether a STUN/TURN server is alive and properly responding to connections. Ideally this test would be performed from an external machine, just in case the STUN/TURN machine is down for this case should also be reported by the connectivity test.

Has anyone looked into this case in the past? What solutions would be recommended?

STUN/TURN server connectivity test

After (briefly) reviewing the Go language spec, effective Go, and the Go memory model, I&#39;m still a little unclear as to how Go channels work under the hood.

What kind of structure are they? They act kind of like a thread-safe queue /array.

Does their implementation depend on the architecture?

How are Go channels implemented?

I was reading some go code and say a few different ways to pass go channels. Maybe they are the same but I was wondering if there is any difference since I couldn&#39;t find documentation online:

1)

    func serve(ch &lt;-chan interface{}){ //do stuff }

2)

    func serve(ch chan&lt;- interface{}){ //do stuff }

3)

    func serve(ch chan interface{}){ //do stuff }


4)

    func server(ch *chan interface{}){ //do stuff}

I was wondering what the difference between them were and if they were just equivalent ways to do the same thing: pass a channel around different goroutines.

NOTE: I am aware that there is no reason to pass a pointer to a chan, map, or slice, or function value, since those are all reference types which internally contain a pointer (the exception would be if you want the callee to change the reference type header). The only reason I provided it is for completeness (i.e. to really provide every way a channel could be attempted to be passed as a parameter and to make on question that hopefully, references all ways to do this and compares them).




Different ways to pass channels as arguments in function

How to know a buffered channel is full? I don&#39;t know to be blocked when the buffered channel is full, instead I choose to drop the item sent to the buffered channel.

How to know a buffered channel is full

Golang newbie here. 

Is there a functional difference between

    func randomNumberGenerator() &lt;-chan int {

and 

    func randomNumberGenerator() chan int {

I&#39;ve tried using both and they seem to work fine for me.

I&#39;ve seen the former used by Rob Pike (one of Go creators) in his Go Concurrency Patterns talk at Google IO 2012. I&#39;ve also seen it used in Go&#39;s official website. Why add 2 extra characters (&quot;&lt;-&quot;) when you can omit it? I&#39;ve tried looking for the difference on the web, but couldn&#39;t find it.

What&#39;s the difference between &quot;&lt;-chan&quot; and &quot;chan&quot; as a function return type?

I&#39;m working on a concurrent Go library, and I stumbled upon two distinct patterns of synchronization between goroutines whose results are similar:

[**Waitgroup**][1]

```
package main

import (
	&quot;fmt&quot;
	&quot;sync&quot;
	&quot;time&quot;
)

var wg sync.WaitGroup

func main() {
	words := []string{&quot;foo&quot;, &quot;bar&quot;, &quot;baz&quot;}

	for _, word := range words {
		wg.Add(1)
		go func(word string) {
			time.Sleep(1 * time.Second)
			defer wg.Done()
			fmt.Println(word)
		}(word)
	}
	// do concurrent things here

	// blocks/waits for waitgroup
	wg.Wait()
}
```

[**Channel**][2]
```
package main

import (
	&quot;fmt&quot;
	&quot;time&quot;
)

func main() {
	words := []string{&quot;foo&quot;, &quot;bar&quot;, &quot;baz&quot;}
	done := make(chan bool)
	// defer close(done)
	for _, word := range words {
		// fmt.Println(len(done), cap(done))
		go func(word string) {
			time.Sleep(1 * time.Second)
			fmt.Println(word)
			done &lt;- true
		}(word)
	}
	// Do concurrent things here

	// This blocks and waits for signal from channel
	for range words {
		&lt;-done
	}
}

```

I was advised that `sync.WaitGroup` is slightly more performant, and I have seen it being used commonly. However, I find channels more idiomatic. What is the real advantage of using `sync.WaitGroup` over channels and/or what might be the situation when it is better?


  [1]: https://play.golang.org/p/ZYPLlcp16TZ
  [2]: https://play.golang.org/p/CglhQg0eVjL

Content Type	Original Author	Original Content on Stackoverflow
Question	onmyway133	View Question on Stackoverflow
Solution 1 - Tcp	tomtheengineer	View Answer on Stackoverflow

Does WebRTC use TCP or UDP?

Tcp Problem Overview

Tcp Solutions

Solution 1 - Tcp

bootstrap dropdown bubble align right (not push-right)

Clear screen in shell

Attributions