so i started to work in this summer and the first task they have given to me is to use wireshark to understand why an application that uses webRTC doesn't use the turn server.
Can you guys help me out, to understand which steps should i do to understand better where is the problem.
I already run the wireshark and only get protocols STUN, that bind to a UDP connection.
TURN is a STUN extension so you will only see STUN packets in Wireshark.
You can easily test WebRTC+TURN in isolation using this sample from the WebRTC project. Remove the default stun server and add the url and credentials for your own TURN server.
Fire up wireshark, start capturing.
Click the "gather candidates" button on that page. You should see candidates with host type at least. You should, if the browser can reach the TURN server usually also see candidates with a srflx type.
If the TURN server is working and your credentials are valid, then you will get candidates with type relay. But you probably wouldn't be asking then.
Now go back to wireshark. Set the display filter to 'stun'. You should see some packets sent to the ip address of the TURN server. Right-click on one of them, 'follow' and 'udp stream'. That should show you all the packets between the browser and the TURN server.
You should be seeing binding requests (message_type=0x01) as well as binding success responses (message_type=0x101) from the server. If you don't see those, your turn server is not responding or something is blocking the client. You will also not get srflx candidates on the candidate gathering demo page.
You should also see packets wireshark interprets as 'allocate request udp' (the message type is 0x101). These are the important ones for TURN.
You should see an error from the TURN server with a message type 0x113 and an error code 401 (unauthorized) because in the first packets, there is no username attribute. In response to those the browser will start sending allocate requests that contain both a username and a message-integrity.
If things go well, those should be answered with an allocate success response (message type=0x103) indicating a xor-relayed-address.
If not and you see more 401 errors that usually means your username and password is wrong.
You might also find the articles on using wireshark to reverse-engineer Amazon Mayday and Whatsapp on WebRTChacks useful -- both use Wireshark.
The WebRTC project has some notes on Wireshark, too.
Related
I'm hosting a video call site on Heroku here. My coturn server is Docker-containerized and runs locally behind a router. I opened all the needed ports (3478, 40000-65535) so that I am sure that the corturn server works over the Internet because I checked it using Trickle ICE.
Everything is fine except the video is not transferred when two devices are not on the same network. Any help?
Are you sure your code is using the TURN server?
You need to pass the iceServers list with the TURN URL and correct credentials. See for example https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/setConfiguration
Then when you start the ICE gathering, you should see in coturn's log whether the Allocate Request was correctly handled and a XOR-RELAYED-ADDRESS returned.
But you can also see it on the client side (the laptop where the browser runs) by launching a tool like Wireshark and selecting stun as filter.
In addition, when the call is up, check in chrome://webrtc-internals whether ICE gathering has provided relay candidates.
I am trying to get a browser client to connect with my C++ linux application using WebRTC. So my environment is not the typical triangle WebRTC where 2 browsers setup a WebRTC call thru a server. Instead, the browser client side is typical, but my application is acting as the server and the remote client, so it does the signalling and also streams the SRTP media using gstreamer.
I am successful up to a point. I have successfully exchanged the ice candidates and the offer/answer SDP exchange is also successful. The browser ICE connection state successfully goes to "checking" and at that point I am stuck.
Question: Is the server or remote browser involved in the ice checking operations? That is, does the browser do the ICE checking with the STUN server or with the actual candidate address from the remote end. That would then imply that my C++ application has to be involved in that checking process.
Thanks,
-Andres
your server needs to respond to STUN binding requests at least which are sent as part of ICE.
If your server always has a public IP, using ice-lite (see RFC 5245) will make your life a lot easier.
I'm playing around with WebRTC, and what I'd like to achive is:
User1 opens the browser at 192.168.x.x
User2 opens the browser at 192.168.x.x
The same page
User 1 clicks call, user2 displays the stream on his screen.
I've created a signaling server with node and socket.io and I'm able to exchange messages betweeen users using socket.io rooms.
The steps I'm following are:
Get User Media
Create peerconnection1 - no ice servers
add the stream on peerconnection
create the offer
send offer via sockets
Receive the offer and create peerconnection2 - no ice servers
sending the answer
I've also put some logging in "onicecandidate" and "onaddstream" to see when they are called, and on "onaddstream" I create the videoelement.
When I press the call button I see on the other computer that the video element becomes black but I dont see any video neither audio.
For sure I'm missing some vital steps,
Could someone tell me the steps I have to do to make a correct call and exchange all the necessary data to display the stream on the other side?
Thank you very much
A STUN server is used to get an external network address.
TURN servers are used to relay traffic if direct (peer to peer) connection fails.
see this image describes how peerconnection works
webRTC Basics
You should still have at least a stun server referenced for one of your ICE servers. I would use 'stun:stun.l.google.com:19302' || 'stun:23.21.150.121', even though you do not technically need one.
But if you do not have ICE servers, you do not need to worry about gathering candidates. A couple of things that could be happening.
Make sure you Add your streams to each connection BEFORE creating your offer and creating your answer, it says you get the user media but not that you add it to your peerconnection
You are not setting your local and remote descriptions
Offering computer should set their local description when creating it
Answering computer should set their remote description with the offering description and set their local description with the one they create
Make sure you send the answer sdp to the initial offering computer and that that offering computer sets it as their remote description.
Streams WILL NOT send to each other unless you add the needed streams, create your descriptions, and then set their local and remote descriptions accordingly.
If that does not work, you should probably post your code for the page.
What I want is, basically, to create a connection between two different computers on same local network. But i want to do this by computers' local IP's. (like 192.168.2.23 etc)
This must be a totally local connection. no TURN or STUN Servers. I am not sure if this is possible. Because there are not much documentation/example/information about WebRTC.
So, how can I create a connection from my computer to another one just passing its local IP as parameter?
Update: To be more clear; imagine there is an html page contains some code that activates my camera and audio services. and another -almost same- page is open in other computer. Waiting a connection request... And there is a textbox in my page to type an IP belongs to other computer on my local network. type 192.168.2.xx and bingo! i have connection between me and other computer.
I want this process as IP based, because there may be more than 2 devices on the network. And all of them are possible devices to create connection. So i need to reach them by their IP's.
Any example code or explanation would be great! even if it tells that this is not possible.
Thanks
Peer discovery is a vital part in any WebRTC application. It's an expensive term for saying: "Hi, I'm computer 4 and I want to talk to you!".
See it as calling a friend over the phone. You need to dial his number first.
This part is not defined in the WebRTC standards. You need to implement this logic in your application. Once you know who you want to call, you need a way of exchanging vital information. This is called signaling, like flo850 put in his answer.
Signaling is needed before any peer-to-peer connection can be set up.
To come up with an idea for your use case of 7 devices in a LAN.
If you have these devices connected to for example a WebSockets server and are in the same channel.
The WebSockets server can be written to route messages to specific receivers.
Devices connected to the channel often are identified with some kind of ID, imagine you use the device's IP.
When you want to talk to computer 4 with IP 192.168.0.4 you send the exchange messages (signaling) on the channel to the receiver with ID, the IP of the device you want to connect with.
How to send the signaling (offer, answer) is described here with example code.
Hope this helps
Users usually sit behind NATs; that's why ICE concept implemented in WebRTC.
If both users are sitting behind same NAT; you can skip ICE servers by passing "NULL" parameter value over "RTCPeerConnection" constructor:
var peer = new [webkit|moz]RTCPeerConnection ( null );
Now, browser will use "host" candidates, also known as "local" candidates.
you still need a signaling server. During the ICE candidate search, your clients will exchange their local ip through this signaling server
Basically, I want to check to see if a game server is online/offline but the server is hosted through a port, how would I go about pinging it to check this. For testing purposes the server I want to ping is fr7.mooshroom.net:25667, however eventually I will be importing the server IP and port from an online plist so could the ip address and port be separated
If you could give me a step by step guide on how to do this, that would be much appreciated.
I am using the Reachability by tonymillion because the apple one doesnt work with iOS 5.
First, it depends on if you are talking about TCP or UDP.
If it's TCP, then the answer may be simple: try to open a TCP connection to the server at that port. If the connection opens, drop the connection and report success. Otherwise, failure. This is the simplistic view. It's possible if there are load balancers or firewalls in front of the server, the TCP connection may open but the backend server is down.
UDP would be harder. There is no way to know when you send UDP data that the server got the data unless it sends you some kind of response. It's possible that if the UDP server is down and you sent a message to the port, your computer might get an ICMP Error message back. That would definitely let you know the server is down. But firewalls may block this message getting back to your device, so that might not be reliable.
Otherwise, you need to send a properly formatted message to the server to get some kind of response. This is protocol-dependent but is the most reliable.
The traditional "ping" message is an ICMP echo query and response message. As such, there is no "port" associated with ICMP.