Which stun server

This IP is not returning a Binding Request: Google seems to be broadcasting this IP for stun This has resolved the issue for us, but would be great if Google can stop broadcasting that IP. Hi bud, how you test these server address? Just paste them in the web browser address? Skip to content. Sign in Sign up. Instantly share code, notes, and snippets. Created Sep 18, Code Revisions 1 Stars Forks Embed What would you like to do?

Embed Embed this gist in your website. Share Copy sharable link for this gist. Learn more about clone URLs. Download ZIP. STUN server list. This file contains bidirectional Unicode text that may be interpreted or compiled differently than what appears below. To review, open the file in an editor that reveals hidden Unicode characters. Learn more about bidirectional Unicode characters. This comment has been minimized. This enables server-sent events—data sent from a web server to a browser client through HTTP.

EventSource is designed for one-way messaging, but it can be used in combination with XHR to build a service for exchanging signaling messages. A signaling service passes a message from a caller, delivered by XHR request, by pushing it through EventSource to the callee. WebSocket is a more-natural solution, designed for full duplex client—server communication—messages that can flow in both directions at the same time.

One advantage of a signaling service built with pure WebSocket or server-sent events EventSource is that the backend for these APIs can be implemented on a variety of web frameworks common to most web-hosting packages for languages such as PHP, Python, and Ruby. TLS should be used for all connections to ensure messages cannot be intercepted unencrypted and also to reduce problems with proxy traversal.

It is also possible to handle signaling by getting WebRTC clients to poll a messaging server repeatedly through Ajax, but that leads to a lot of redundant network requests, which is especially problematic for mobile devices. Even after a session has been established, peers need to poll for signaling messages in case of changes or session termination by other peers. Although a signaling service consumes relatively little bandwidth and CPU per client, signaling servers for a popular app may have to handle a lot of messages from different locations with high levels of concurrency.

WebRTC apps that get a lot of traffic need signaling servers able to handle considerable load. You don't go into detail here, but there are a number of options for high-volume, high-performance messaging, including the following: eXtensible Messaging and Presence Protocol XMPP , originally known as Jabber—a protocol developed for instant messaging that can be used for signaling Server implementations include ejabberd and Openfire.

JavaScript clients, such as Strophe. The following is code for a simple web app that uses a signaling service built with Socket. The design of Socket. This example is not designed to scale as a production-grade signaling service, but is simple to understand for a relatively small number of users.

It has been ported to various backends, but is perhaps best known for its Node version used in this example. There's no WebRTC in this example. It's designed only to show how to build signaling into a web app. View the console log to see what's happening as clients join a room and exchange messages. You don't need to learn about node-static for this.

It just happens to be used in this example. To run this app on localhost, you need to have Node, Socket. IO, and node-static installed. Node can be downloaded from Node. To install Socket. IO and node-static, run Node Package Manager from a terminal in your app directory:. From your browser, open localhost Open a new tab or window in any browser and open localhost again. To see what's happening, check the console. Whatever approach you choose for signaling, your backend and client app—at the very least—need to provide services similar to this example.

If you don't want to roll your own, there are several WebRTC signaling servers available, which use Socket. This is now somewhat obsolete, but it's worth looking at the code if you're considering something similar. Encryption is mandatory for all WebRTC components. However, signaling mechanisms aren't defined by WebRTC standards, so it's up to you to make signaling secure.

If an attacker manages to hijack signaling, they can stop sessions, redirect connections, and record, alter, or inject content. Also, be careful not to broadcast signaling messages in a way that they can be accessed by other callers using the same signaling server.

For metadata signaling, WebRTC apps use an intermediary server, but for actual media and data streaming once a session is established, RTCPeerConnection attempts to connect clients directly or peer-to-peer. In a simpler world, every WebRTC endpoint would have a unique address that it could exchange with other peers in order to communicate directly. In reality, most devices live behind one or more layers of NAT , some have antivirus software that blocks certain ports and protocols, and many are behind proxies and corporate firewalls.

ICE tries to find the best path to connect peers. It tries all possibilities in parallel and chooses the most efficient option that works. You do not need to configure anything, and you do not need to worry about vending out TURN credentials to authorized clients.

Search Results for. This implies the following preferred precedence of network communication protocols: UDP - Direct between source and target of media flow. UDP - Indirect relayed between source and target of media flow. Here, the traffic must be relayed through the TURN server to bypass restrictive firewall rules, and the preference is UDP over TCP because TCP's guaranteed ordered delivery of packets implies overhead that is undesirable for real-time communications.

There are four types of candidates: "host" candidates - Gathered directly from the local network adapter, host candidates are the internal IP address and port of a computer on a LAN. They can only route between peers on the same subnet.

When signaled to a remote peer they can be used to route traffic over the internet providing that traffic is not blocked by firewall rules.

In this case, these candidates are gathered directly by peers after they have established a connection. A peer reflexive candidate could be used after connectivity has been established if ongoing connectivity checks determine the candidate is routable, and connectivity on the currently active candidate is failing.

This is useful when a firewall does not allow direct routing via srflx candidates. In this case, the remote peer has signaled a candidate it can use to route traffic to the relay server, which can then in turn relay that traffic to the peer behind the restrictive firewall.

The firewall allows the TURN server to route traffic to the peer because the peer made the initial request to the server. In this way, the TURN server acts as a "man-in-the-middle" that the initiating peer uses to circumvent its firewall.

From the discussion above, you know that communication over relay candidates is the least desirable option. Candidates are tested in order of rank. The first to establish connectivity is the winner and becomes the "active" candidate. Ongoing connectivity checks are performed on candidates. If the active candidate fails connectivity checks then a different candidate is used.