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.
Related
Not sure the title makes a lot of sense. To add some context, we are building a WebRTC infrastructure and to so do we have a few STUN servers up and and running.
We sometimes have users complaining of call taking too long to connect therefore we would like to get some analytics on the calls. Because we provide a list of STUN IPs (including some public STUN as backup), we would like to detect the STUN server that successfully initiated the call.
We have collected a bunch of information thanks to RTCPeerConnection.getStats but there is nothing related to the STUN itself. So for my questions:
is there any JS API that allow us to retrieve the STUN used?
is there any tool that I am not aware of that could do the job?
do the SDP contains any information related to STUN?
Hope all of this is clear, thanks for your kind replies
The statistics do contain a server url:
https://w3c.github.io/webrtc-stats/#dom-rtcicecandidatestats-url
However, that is not implemented and since STUN servers are not involved in the actual call that information is unlikely to be useful.
For TURN servers you can get the active candidate pair and the IP of any relay involved from getStats. See https://webrtc.github.io/samples/src/content/peerconnection/constraints/ for a sample that shows how to determine the active candidate pair.
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.
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.
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
I'm using a TK102 GPS localizer. Along with it, I got only simple end-user docs. No API, dev specs or similar for writing code that will use this localizer.
I was told that it uses UDP. So I wrote a simple PHP listener. But either localizer is not using UDP or something is wrong in communication between it and server. Listener works fine (gets UDP packets from other clients) and localizer is sending something (I'm being charge by GSM operator for GPRS transmission), but the data it sends, doesn't reach server.
I asked about server or networking issues on Unix/Linux and SuperUser. Here I would only ask, if someone knows any API/dev-specs for this localizer, so I can check, if it really uses UDP or if I haven't made any other error (in configuration for example).
The localizer and its clones
We're talking about Xexun TK102 Tracker here. The original one, because there are many clones under other companies from China, selling similar GPS localizer, with the same cover and logo, but with:
less performance electronics on-board (for example -- able to report location once per 20 or 30 seconds, not once per 5 seconds like in original one),
the ones that are sending lesser information (lack of direction/bearing, altitude, number of satelites used for location fix and many more),
units using different format of data or non-standard transmission protocol for sending it (for example, cheaper units are unable to use UDP protocol and are transmiting data through TCP protocol, using packets that not always follows standards or definictions.
Coban and Kintech are only two of many clones sold on eBay and in e-shops, claiming to be original Xexun trackers.
On the other hand, original Xexun and some clones (like Coban for example) are harder to control from own script, because they require a correct answer from the server, where data is sent over GPRS. If unit does not receive such reply, it breaks connection. The cheapes unit does not have this checking and will always sent location data to specified IP address over provided port.
Product description
Here is product description of original Xexun localizer (and here is a clone under Kintech name).
Possible buyer must be very careful (and should secure return policy, for which buying directly in China is not recommended) as there are many reports about sellers claiming to sell original Xexun device and sending a clone actually.
Though this device is five years old, it is still sold at many places (including eBay), but even at theses sources it is very hard to get anything worth for developers, except some simple, very basic user guide.
I have confirmed information (from two different sources) that there is no official API available for this device. The only option is to Google around, ask other users or use forums (see below).
If you own original Xexun localizer, you may try to contact company international departament and ask their technicians to include some changes to device source code and to send you updated firmware, with your changes - wow! That was confirmed by company itself.
Forum
I found a perfect forum for TK102 device, with a lot of questions and answers:
here is a general forum on TK102 device (kept alive for 4,5 year with 171 pages and 2000+ posts!),
here you'll find more specific topic on receiving data from this localizer,
this forum is also about TK102 unit, but it is entirely in French.
There are many other devices dissussed and in general, this is the biggest forum in the world, with topics for localizers and simillar information.
GPRS Protocol Specs
In general, any TK102 related devices is opening a socket for a direct TCP transmission (original one can be switched to use UDP protocol). Data is being transsmited over port specified by user, in configuration and using GPRS only (requires SIM card with enabled GPRS, there is no way to use WiFi).
Sending frequency, format and amount of data being send, entirely depends on kind of device is being used -- it is more extensive and more configurable in original one than in clones.
Using FileDropper I shared GPRS Protocol Specification for TK102 Geolocalizer. It contains basic information on how to setup TK102 (and possible all its clones) to send location over GPRS. And what sort of data you should except to receive from in, on server side. This could be useful for someone.
BTW: If links goes dead, contact me for a reupload or sending it over e-mail
Correct server response problem
Make sure, if you're using correct data transmission protocol! Many (really many) cheap clones uses TCP, while only original TK102 allows switching to UDP. This is convenient, because you need really basic server configuration to handle TCP connections, while you have to use specific server-side software (like node.js) or specific configuration (open to certain ports) to handle UDP. But the key thing is to determine correct protocol, as listening to TCP data, while your localizer sends UDP, will most certainly fail.
Take into consideration, that many TK102 clones requires a correct response from the server after each data, it send. It breaks connection after sending some welcome garbage UDP packet, as it does not receive response, it waits for.
It is quite hard (quite impossible?) to find any guide to many of these clones, on what kind of responses server should sent. This often leads into situation of developer being unable to estabilish two-way communication between server and localizer. Many localizers are sold to be used only via SMS communication or throughs paid services that had signed and agreement with producer and received protocol specification that contains valid responses server should generate for particular TK102 clone.
Double check, if this is not source of problem, if you can't communiacte with your localizer from your app.
You can check some models protocol specs here:
http://www.traccar.org/docs/protocol.jsp