opensource

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One of WebRTC’s benefits is that the source to it is all open source. Building WebRTC from source provides you the ultimate flexibility to do what you want with the code, but it is also crazy difficult for all but the small few VoIP stack developers who have been dedicated to doing this for years. What benefit does the open source code provide if you can’t figure out how to build from it?

As WebRTC matures into mobile, native desktop apps, and now into embedded devices as part of the Internet of Things, working with the lower-level source code is becoming increasingly common.

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As Reid previously introduced in his An Intro to WebRTC’s NAT/Firewall Problem post, NAT traversal is often one the more mysterious areas of WebRTC for those without a VoIP background. When two endpoints/applications behind NAT wish to exchange media or data with each other, they use “hole punching” techniques in order to discover a direct communication path that goes from one peer to another through intervening NATs and routers but not traversing any relays. “Hole punching” techniques will fail if both hosts are behind certain types of NATs (e.g. symmetric NATs) or firewalls. In those cases, a direct communication path cannot be found and it’s necessary to use the services of an intermediate host that acts as a relay for the media or data packets, which typically sits on the public Internet. The TURN (Traversal Using Relays around Nat) protocol allows an endpoint (the TURN client) to request that a host (the TURN server) act as a relay. So far TURN, along with ICE and STUN, has seen little deployment. Now that it is a fundamental piece of WebRTC, it is gaining some momentum. In fact, at the IETF we’re now starting a new effort that will focus on enhancements to TURN/STUN that will be applicable to WebRTC deployments. This new effort is called TRAM (Turn Revised And Modernized), and we’re currently discussing its charter.

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