How to Validate pair in the ICE protocol? - webrtc

Related WebRTC, ICE protocol gives the which pair of addresses will work for direct media transfer between the pairs.
Let A and B are two endpoints
To choose which address will work for direct communication between A and B, Person A first gather candidates, encode candidate attribute, encode the SDP offer message, and send it to another endpoint.
When B get offer message from A,then person B gather candidates, encode the SDP answer message with its own list of candidates and send it to person A.
At this end of this process, each agent has a complete list of local candidates and Remote candidates. Its pairs them up, resulting in CANDIDATE PAIRS. To see, which pair work, each agent performs the connectivity checks using STUN req/resp.
How many connectivity checks are performed, to nominate valid candidate pair?
What are the remaining ICE connectivity checks are performed regarding webRTC call?
To develop ICE module for webRTC call, I have to follow each step in RFC5245 or any thing else?

How many connectivity checks are performed, to nominate valid
candidate pair?
The number of candidate pairs are the number of connectivity checks done by each side.
What are the remaining ICE connectivity checks are performed regarding
webRTC call?
There are no extra ICE connectivity checks for webRTC.
To develop ICE module for webRTC call, I have to follow each step in
RFC5245 or any thing else?
You have to implement or use existing implementation of DTLS protocol, RFC5763 and RFC5764. DTLS implementation can be found on OpenSSL library.
All these seems a lot of work but if you use openssl then its easy enough.

Related

In WebRTC, can ICE candidates be re-used across different RTCPeerConnections?

I am working on setting up group calls involving up to 8 peers using WebRTC.
Let's say a peer needs to set up 7 RTCPeerConnections to join a group call. Instead of relying on onicecandidate event for every single RTCPeerConnection, I was wondering if I can track the client's icecandidates in a central location and reuse it for each new RTCPeerConnection. (e.g. Signaling Server will keep track of a peer's full ICE candidates, and share them with other peers as soon as they need them).
I am unsure what the average number of 'icecandidates' each client will have, but with ice trickle process, it seems that many duplicate http or websocket calls will need to be made to a Signaling Server in oder to exchange ice candidates between any 2 peers.
So I was wondering if I could just "accumulate" ice candidates locally and reuse them when new RTCPeerConnection will need to be made with new peer.
You can not. ICE candidates are associated with the peerconnection and its ice username fragment and password.
There is a feature called ice forking that would allow what you ask for but it is not implemented yet. https://bugs.chromium.org/p/webrtc/issues/detail?id=11252#c3 has some details.

Send offers with WebRTC only

I want to create something similar like chat roulette:
There are two peers. Both peers send an SDP offer to the signaling server asking it to get connected with someone. The signaling server uses the offer of peer A to send it as an answer to peer B and vice versa.
Both peers could setLocalDescription() and setRemoteDescription() without using createAnswer().
Could they now go to the next step and exchange candidates? Or is it necessary that at least one is sending a real answer created with createAnswer()?
No. An offer is not an answer. An answer builds on an offer, it's a refinement, an iteration on it, a negotiation.
The offer-answer exchange is inherently asymmetric, so your peers will be in incompatible states if they've both sent out offers.
Instead, solve discovery (pairing) of A and B first, then do WebRTC from A to B.

WebRTC - TURN and ICE functions

I'm trying to understand a concept of WebRTC. As I found in some descriptions (for example here http://www.innoarchitech.com/content/images/2015/02/webrtc-complete-diagram.png), there is such a way of making a connection:
Call STUN, to get your IP:port address.
Get some channel from TURN - with that channel you can send info to other peer.
Send to other peer ICE candidates.
Accept ICE candidates with other peers- start a call.
The question is, what do we need ICE candidates for? We know our IP, we can send it to TURN therefore to other peer, and on TURN we have a nice connection with other peer- so we don't have to scary about NATs. Why except that we are sending ICE candidates (why many?), and why we need to use them?
We have 3 main concepts here:
ICE
TURN
STUN
The ICE negotiation is not that simple...
To execute ICE, UAs have to identify all address candidates, transport addresses. Transport addresses are a combination of IP address and port for a particular transport protocol. There are three types of candidates:
Host Candidate – transport address associated with a UA’s local interface
Relayed Candidate – transport address associated with a TURN server (can only be obtained from a TURN server)
Server Reflexive Candidate – translated address on the public side of the NAT (obtained from either a STUN server or a TURN server)
After UA1 has gathered all of its candidates, it arranges them in order of priority from highest to lowest and sends them to UA2 in attributes in an SDP offer message. UA2 performs the same candidate gathering and sends a SDP response with it’s list of candidates. Each UA takes the two lists of candidates and pairs them up to make candidate pairs. Each UA gathers these into check lists and schedules connectivity checks, STUN request/response transaction, to see which pairs work. Figure 3 shows the components of the candidate pairs that make up the UA check list.
ICE assigns one of the agents as the “Controlling Agent” and the other as the “Controlled Agent”. The controlling agent used the valid candidate pairs to nominate a pair to use for the media. There are two nomination methods that can be used:
Regular Nomination The checks continue until there is at least one valid candidate pair. The controlling agent picks from the valid pairs and sends a second STUN request on that pair with a flag to tell the peer that this is the one that is nominated for use.
Aggressive Nomination The nomination flag is sent with every STUN request, once the first check succeeds ICE processing for that media stream is finished and a second STUN request is not needed.
Each candidate pair in the check list has a state associated with it. The state is assigned by the UA once the check list has been computed. There are five possible states:
Frozen This pair can only be checked after being put in the waiting state. To enter the waiting state some other check must succeed first.
Waiting As soon as this is the highest priority pair in the check list a check will be performed.
In-Progress A check has been sent for this pair and the transaction is in progress
Succeeded Successful result from pair check.
Failed Failed result from pair check.
The link below includes more information and diagrams of the ICE flow.
Reference:
http://www.vocal.com/networking/ice-interactive-connectivity-establishment/
RFC https://www.rfc-editor.org/rfc/rfc5245#page-9
TURN is typically used only as a fallback when a direct peer-to-peer connection cannot be established. The latter is the hard part, and is what ICE is for.
Always using TURN, is an option, but a bit of an edge-case.

ICE connectivity in a WebRTC call

In a Webrtc call, I am using sip signalling and sdp for media parameter negotiation.
Before call start, I do a stun-bind transaction and get reflexive candidates. I have put those reflexive candidates in sdp in addition to base and host candidates.
As soon as we get 200 OK for Invite, we need to start media. For media start, I need to know which candidate pair I need to use.
I hope to determine which candidate pair I need to use, we need to do connectivity check. I am not sure how to do connectivity check (like which message to send.. etc).
Can somebody help me in this to understand.
Also is there an open source (c, linux based), that gives ice/stun/turn support.
This information is given on RFC 5245. You need to read this RFC for implementing ICE. For your query about doing ICE connectivity check, read this section of the RFC.
Also is there an open source (c, linux based), that gives
ice/stun/turn support.
Search google for this and you will get your answer.

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

I newly wrote a simple chat application, but I didn'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 ?
the answer from #Ichigo is correct, but it is a litte bit bigger. Every ICE contains 'a node' of your network, until it has reached the outside. By this you send these ICE's to the other peer, so they know through what connection points they can reach you.
See it as a large building: one is in the building, and needs to tell the other (who is not familiar) how to walk through it. Same here, if I have a lot of network devices, the incoming connection somehow needs to find the right way to my computer.
By providing all nodes, the RTC connection finds the shortest route itself. So when you would connect to the computer next to you, which is connected to the same router/switch/whatever, it uses all ICE's and determine the shortest, and that is directly through that point. That your collegue got less ICE candidates has to do with the ammount of devices it has to go through.
Please note that every network adapter inside your computer which has an IP adress (I have a vEthernet switch from hyper-v) it also creates an ICE for it.
ICE stands for Interactive Connectivity Establishment , its a techniques used in NAT( network address translator ) for establishing communication for VOIP, peer-peer, instant-messaging, and other kind of interactive media.
Typically ice candidate provides the information about the ipaddress and port from where the data is going to be exchanged.
It's format is something like follows
a=candidate:1 1 UDP 2130706431 192.168.1.102 1816 typ host
here UDP specifies the protocol to be used, the typ host specifies which type of ice candidates it is, host means the candidates is generated within the firewall.
If you use wireshark to monitor the traffic then you can see the ports that are used for data transfer are same as the one present in ice-candidates.
Another type is relay , which denotes this candidates can be used when communication is to be done outside the firewall.
It may contain more information depending on browser you are using.
Many time i have seen 8-12 ice-candidates are generated by browser.
Ichigo has a good answer, but doesn't emphasise how each candidate is used. I think MarijnS95's answer is plain wrong:
Every ICE contains 'a node' of your network, until it has reached the outside
By providing all nodes, the RTC connection finds the shortest route itself.
First, he means ICE candidate, but that part is fine. Maybe I'm misinterpreting him, but by saying 'until it has reached the outside', he makes it seem like a client (the initiating peer) is the inner most layer of an onion, and suggests the ICE candidate helps you peel the layers until you get to the 'internet', where can get to the responding peer, perhaps peeling another onion to get to it. This is just not true. If an initiating peer fails to reach a responding peer through the transport address, it discards this candidate and will try a different candidate. It does not store any nodes anywhere in the candidate. The ICE candidates are generated before any communication with the responding peer. An ice candidate does not help you peel the proverbial NAT onion. Also regarding the second quote I made from his answer, he makes it seem like ICE is used in a shortest path algorithm, where 'shortest' does not show up in the ICE RFC at all.
From RFC8445 terminology list:
ICE allows the agents to discover enough information
about their topologies to potentially find one or more paths by which
they can establish a data session.
The purpose of ICE is to discover which pairs of addresses will work. The way that ICE does this is to systematically try all possible pairs (in a carefully sorted order) until it finds one or more that work.
Candidate, Candidate Information: A transport address that is a
potential point of contact for receipt of data. Candidates also
have properties -- their type (server reflexive, relayed, or
host), priority, foundation, and base.
Transport Address: The combination of an IP address and the
transport protocol (such as UDP or TCP) port.
So there you have it, (ICE) Candidate was defined (an IP address and port that could potentially be an address that receives data, which might not work), and the selection process was explained (the first transport address pair that works). Note, it is not a list of nodes or onion peels.
Different users may have different ice candidates because of the process of "gathering candidates". There are different types of candidates, and some are obtained from the local interface. If you have an extra virtual interface on your device, then an extra ICE will be generated (I did not test this!). If you want to know how ICE candidates are 'gathered', read the 2.1. Gathering Candidates