ICE

All posts tagged ICE

webrtcH4cKS: ~ Am I behind a Symmetric NAT?

NATs can be a nuisance for VoIP, particularly Symmetric NATs . Fortunately WebRTC includes tools for dealing with them. Image source: http://pinktentacle.com/

WebRTC establishes peer-to-peer connections between web browsers. To do that, it uses a set of techniques known as Interactive Connectivity Establishment or ICE. ICE allows clients behind certain types of routers that perform etwork Address Translation, or NAT, to establish direct connections. (See the WebRTC glossary entry for a good introduction.) One of the first problems is for a client to find what its public IP address is. To do so, the client asks a STUN server for its IP address.

NATs are boxes (physical or virtual) that connect our local private networks to the public internet. They do so by translating the internal IP addresses we use to public ones. They work differently from one another, which ends up requiring WebRTC to rely on both STUN and TURN in order to connect calls. For background on these, check out some of our past posts on this topic like this one and this one.

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slack webrtc2

Dear Slack,

There has been quite some buzz this week about you and WebRTC.

WebRTC… kind of. Because actually you only do stuff in Chrome and your native apps:

I’ve been there. Launching stuff only for Chrome. That was is late 2012. In 2016, you need to have a very good excuse to launch something with WebRTC and not support Firefox like this:
 

Maybe you had your reasons. As usual, I tried to get a dump from chrome://webrtc-internals to see what is going on. Thanks to Dag-Inge Aas for providing one. The most interesting bit is the call to setRemoteDescription:

I would like to note that you reply to Chrome’s offer of UDP/TLS/RTP/SAVPF with a profile of RTP/SAVPF. While that is still tolerated by browsers, it is improper.
Your a=msid-semantic line looks very interesting. “WMS janus”. Sounds familiar, this is meetecho’s janus gateway (see Lorenzo’s post on gateways here). Which by the way works fine with Firefox from what I hear.

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The “IP Address Leakage” topic has turned into a public relations issue for WebRTC. It is a fact that the WebRTC API’s can be used to share one’s private IP address(es) without any user consent today. Nefarious websites could potentially use this information to fingerprint individuals who do not want to be tracked. Why is this an issue? Can this be stopped? Can I tell when someone is trying to use WebRTC without my knowledge? We try to cover those questions below along with a walkthrough of a Chrome extension that you can install or modify for yourself that provides a notification if WebRTC is being used without your knowledge.

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This is the next decode and analysis in Philipp Hancke’s Blackbox Exploration series conducted by &yet in collaboration with Google. Please see our previous posts covering WhatsApp and Facebook Messenger for more details on these services and this series. {“editor”: “chad hart“}

FaceTime is Apple’s answer to video chat, coming preinstalled on all modern iPhones and iPads. It allows audio and video calls over WiFi and, since 2011, 3G too. Since Apple does not talk much about WebRTC (or anything else), maybe we can find out if they are using WebRTC behind the scenes?

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Unnatural shrinkage. Photo courtesy Flikr user Ed Schipul

Unnatural shrinkage. Photo courtesy Flikr user Ed Schipul

One evening last week, I was nerd-sniped by a question Max Ogden asked:

That is quite an interesting question. I somewhat dislike using Session Description Protocol (SDP)  in the signaling protocol anyway and prefer nice JSON objects for the API and ugly XML blobs on the wire to the ugly SDP blobs used by the WebRTC API.

The question is really about the minimum amount of information that needs to be exchanged for a WebRTC connection to succeed.

 WebRTC uses ICE and DTLS to establish a secure connection between peers. This mandates two constraints:

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For the last year and a half I’ve been working with a number of customers helping them to understand what WebRTC is about, supporting them in the definition of new products, services, and in some cases even developing WebRTC prototypes/labs for them. I’ve spent time with Service Providers, Enterprise and OTT customers and the very first time I demoed WebRTC to them, after the initial ‘wow moment’ almost all of them complained about the ‘call setup delay’, as in some cases represented tens of seconds.

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The WebRTC NAT/Firewall traversal trifecta

Most folks that set out to write an application, or build an architecture, begin with nothing but features and functionality in mind.  Many might start out assuming they will be traversing flat, reliable, and secure networks.  Inevitably, reality sets in as one starts to demo or prototype much beyond the friendly confines of the lab, and suddenly you begin finding scenarios not working properly, quality issues cropping up, or your stuff gets hacked.  “Phase 2” of the design emerges, backing in all the necessary tools to cover the gaps.

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