I am building a WebRTC app where two users are selected at random and then connect to each other to chat. Both clients keep an open WebSocket connection and I am planning to use this to exchange their offers/answers to signal a connection. The case I am trying to account for is when there is a peer that intentionally sends bad configuration information, and also when the peer might spontaneously disconnect in the middle of the signaling exchange.
My solution to the first case is have the server keep state of the exchange, so when the connection is first established I would expect that user A provide an offer and user B have an answer. Is this appropriate? or should this be implemented exclusively client side?
My solution to the second problem feels to me like a hack. What I am trying to do is notify the user that a match has been made and then the user will set a timeout say 20 seconds, if a connection hasn't been made in that amount of time then it should move on...
Are these appropriate solutions? How do you reliably establish a WebRTC when either peer can't be trusted? Should the signaling server be concerned with the state of the exchange?
Sounds like you're more concerned about call set up errors rather than being able to trust the identity of the remote peer. They are two very different problems.
Assuming it is the call set up errors you are concerned about you shouldn't be trying to avoid them you should be trying to make sure your application can handle them. Network connection issues are something that will always crop up and need to be handled.
Setting a timer for the establishment of a WebRTC call to complete is a logical solution. Displaying a warning to the user that the time limit is approaching also seems like a good idea. SIP is a signalling protocol and it has a defined timeout for the completion of a transaction and if it doesn't complete within that time it will generate an error response. You could use the same approach.
Related
I would like to send messages between two peers using WebRTC without a signaling server. I'm using QR codes to send over the connection descriptions, and I found an example implementing exactly this:
https://github.com/TomasHubelbauer/webrtc-qr-signaling-channel
However, it requires scanning the QR code both ways. I'm wondering if it's possible to only have to scan a QR code from the phone, but still establish a connection?
Conceptually, I'm thinking like this:
ClientA shows QR code, essentially saying "here's how to connect to me."
ClientB scans it, and now knows how to find ClientA.
ClientB sends "here's how to connect to me" to clientA.
They now both know how to find and talk to eachother.
However, this doesn't seem possible when looking through the WebRTC documentation. If this isn't possible to do - then why?
I strongly doubt it is possible.
During the offer/answer exchange, each of the peers generates a self-signed (D)TLS certificate and includes its cryptgraphic hash ("fingerprint") in the SDP. Without the answer's SDP, the offering peer would be unable to authentify the answering one.
You might be able to achieve the same user experience, however, by implementing a signalling server on one of the peers. The QR code would contain the URL of the signalling server together with authentication data, and once connected the two peers could perform a traditional offer/answer SDP, and even trickle ICE candidates.
Scenario:
I'm using WebRTC (Google's libjingle) on iOS and PeerConnection is setup using a TURN server and I'm waiting for all candidates to gather before I send them to the peer (I'm using SIP). The problem is that although all candidates are gathered in around 1-3 seconds (I can see it in the logs) the iceGatheringChanged() callback is not called with state GatheringComplete until after around a whole minute!
Any idea why that happens?
After analyzing the traffic using Google's AppRTCDemo for iOS it seems that for GatheringComplete to fire, the client needs to already have received the candidates from the remote side, and that because it seems to need to setup TURN Allocations and add Permissions on the new allocation so that data can be exchanged with the peer. Is that the case? If so why?
Best regards
Are you exchanging the candidates for both party in real time? You are right, TURN client requires the other party candidates to create permission in TURN server and also to make check lists to start ICE processing.
Signaling is not addressed by WebRTC (even if we do have JSEP as a starting point), but from what I understand, it works that way :
client tells the server it's available at X
server holds that information and maps it to an identifier
other client comes and sends an identifier to get connection information from the first client
other client uses it to create it's one connection information and sends it to the server
server sends this to first client
both client can now talk
This is all nice and well, but what happends if a 3rd client arrives ?
You have to redo the whole things. Which suppose the first two clients are STILL connected to the server, waiting for a 3rd client to signal itself, and start the exchanging process again so they can get the 3rd client connection information.
So does it mean you are required to have to sort of permanent link to the server for each client (long polling, websocket, etc) ? If yes, is there a way to do that efficiently ?
Cause I don't see the point of having webRTC if I have to setup nodejs or tornado and make it scales to the number of my users. It doesn't sound very p2pish to me.
Please tell me I missed something.
What about a chat system? Do you really need to keep a permanent link to the server for each client? Of course, because otherwise you have no way of keeping track of a user's status. This "permanent" link can be done different ways: you mentioned WebSocket and long polling, but simple periodic XHR polling works too (although this will affect the UX, depending on the interval).
So view it like a chat system, except that the media stream is P2P for reduced latency. Once a P2P WebRTC connection is established, the server may die and, of course, the P2P connection will be kept between the two clients. What I mean is: both users may always block your server once the P2P connection is established and still be connected together in the wild Internets.
Understand me well: once the P2P connection is established, your server will not be doing any more WebRTC signalling. The connection is only needed to keep track of the statuses.
So it depends on your application. If you want to keep the statuses of users and make them visible to others, then you're in the same situation as a chat system: you need to keep a certain link, somehow, to make sure their statuses are synced. Otherwise, your server exists to connect them together and is not needed afterwards. An example of the latter situation is: a user goes to a webpage, the webpage provides him with a new room URL, the user shares this URL to another peer by another mean, the other peer joins the room, server connects them together (manages WebRTC signalling) and then forgets them. They are now connected until one of them breaks the link. Just like this reference app.
Instead of a central server keeping one connection per client, a mesh network could also be considered, albeit difficult to implement.
I have a project coming up where I need to send and receive messages through a specific mobile operator, which only provides an SMPP interface. The whole project will be a hosted website. I have already read quite a lot, but I do not yet quite understand what is actually needed from my side to use the protocol.
Should my application try to maintain a constant connection to the smpp?
Can I simply connect, send a message and then disconnect?
Are receiving messages based on push or pull?
Thanks for the help.
SMPP is a peer-to-peer protocol. That should mean that SMS Gateway (your side) and SMSC (your mobile operator) need to have a proper bind/connection established. Even when there are no SMS or DLRs to send/receive, there is a continous exchange of smpp PDU (enquire_link/enquire-link_resp) that ensure that the bind is established.
In detail, if you send an enquire_link PDU and you get no response (enquire_link_resp) the bind is broken. Your sms won't be delivered (will remain enqueued in your gateway store), and you won't receive MOs (incoming sms) or DLRs (delivery report). To re-establish the connection you should re-initiate the connection.
So, my answer would be that you need a constant connection to SMSC.
You are stating you want to receive messages, as a result at least a bind_receiver is needed. Because you don't know when messages are going to come in, you will have to be constantly connected, rather than disconnecting after each event.
With regards to your question about "push or pull" this depends on how you solve the first problem. If you can build a solution that is constantly connected, the result will be a push (the carrier will push it to you as soon as they receive the message). If (for some reason) you cannot maintain a constant connection, you'll end up building a pull mechanism. You'll connect to the carrier ever X seconds to see if they have a message waiting for you.
I do need to highlight 2 pitfalls though:
A number of carriers in the world, do not store or even accept messages if you are not connected, therefore, depending on which carrier you interact with, you might be forced to use a continuous connection.
Most carriers do not allow you to open and close connections in quick succession. Once you disconnect, you can not reconnect for a time frame of X seconds.
Therefore a constant connection is really the way to go. Alternatively, you can look into a company like Nexmo, which will provide you with a HTTP Call every time a message arrives.
I'm not sure which language your developing your application in, but if you use any of the popular languages (Java, PHP, Perl) there are modules out there that handle basic SMPP Connectivity for you. A quick google search for your language and "SMPP Client" will give you a list of references.
Using techniques as hinted at in:
http://msdn.microsoft.com/en-us/library/system.servicemodel.servicecontractattribute.callbackcontract.aspx
I am implementing a ServerPush setup for my API to get realtime notifications from a server of events (no polling). Basically, the Server has a RegisterMe() and UnregisterMe() method and the client has a callback method called Announcement(string message) that, through the CallbackContract mechanisms in WCF, the server can call. This seems to work well.
Unfortunately, in this setup, if the Server were to crash or is otherwise unavailable, the Client won't know since it is only listening for messages. Silence on the line could mean no Announcements or it could mean that the server is not available.
Since my goal is to reduce polling rather than immediacy, I don't mind adding a void Ping() method on the Server alongside RegisterMe() and UnregisterMe() that merely exists to test connectivity of to the server. Periodically testing this method would, I believe, ensure that we're still connected (and also that no Announcements have been dropped by the transport, since this is TCP)
But is the Ping() method necessary or is this connectivity test otherwise available as part of WCF by default - like serverProxy.IsStillConnected() or something. As I understand it, the channel's State would only return Faulted or Closed AFTER a failed Ping(), but not instead of it.
2) From a broader perspective, is this callback approach solid? This is not for http or ajax - the number of connected clients will be few (tens of clients, max). Are there serious problems with this approach? As this seems to be a mild risk, how can I limit a slow/malicious client from blocking the server by not processing it's callback queue fast enough? Is there a kind of timeout specific to the callback that I can set without affecting other operations?
Your approach sounds reasonable, here are some links that may or may not help (they are not quite exactly related):
Detecting Client Death in WCF Duplex Contracts
http://tomasz.janczuk.org/2009/08/performance-of-http-polling-duplex.html
Having some health check built into your application protocol makes sense.
If you are worried about malicious clients, then add authorization.
The second link I shared above has a sample pub/sub server, you might be able to use this code. A couple things to watch out for -- consider pushing notifications via async calls or on a separate thread. And set the sendTimeout on the tcp binding.
HTH
I wrote a WCF application and encountered a similar problem. My server checked clients had not 'plug pulled' by periodically sending a ping to them. The actual send method (it was asynchronous being a server) had a timeout of 30 seconds. The client simply checked it received the data every 30 seconds, while the server would catch an exception if the timeout was reached.
Authorisation was required to connect to the server (by using the built-in feature of WCF that force the connecting person to call a particular method first) so from a malicious client perspective you could easily add code to check and ban their account if they do something suspicious, while disconnecting users who do not authenticate.
As the server I wrote was asynchronous, there wasn't any way to really block it. I guess that addresses your last point, as the asynchronous send method fires off the ping (and any other sending of data) and returns immediately. In the SendEnd method it would catch the timeout exception (sometimes multiple for the client) and disconnect them, without any blocking or freezing of the server.
Hope that helps.
You could use a publisher / subscriber service similar to the one suggested by Juval:
http://msdn.microsoft.com/en-us/magazine/cc163537.aspx
This would allow you to persist the subscribers if losing the server is a typical scenario. The publish method in this example also calls each subscribers on a separate thread, so a few dead subscribers will not block others...