WebRTC Expert Feature

March 14, 2013

WebRTC Signaling - Making the Right Choices

WebRTC uses an RTCPeerConnection mechanism to communicate streaming data between peers (as WebRTC can involve much more than browsers, we will use the "peer" word), but also needs a signaling process to coordinate communication and send control messages.  Signaling is not specified in the RTCPeerConnection API, so the choice of signaling protocol is left up to the Web applications developer. This decision to not tightly define the signaling protocol is both a stroke of brilliance and a challenge for developers. Having a range of signaling options allow the developer to choose the signaling method best suited to the applications, but it also requires consideration into the implementation decision phase.

WebRTC app developers are left to choose the messaging protocol for their implementation, such as SIP or XMPP, and any appropriate duplex (two-way) communication channel.  The apprtc.appspot.com demonstration uses XHR and the Channel API as the signaling mechanism.  A number of early developers have used WebSocket to carry the signaling traffic. The developer could use an HTTP request mechanism as well.

Making the decision as to the appropriate signaling is highly applications dependent. If the application is integrating to an existing VoIP solution or SIP trunks, using a SIP like format may be best. If it is an extension of an IM based system XMPP may be a better choice. In a pure website application such as a social, WebSocket may be a good choice. Or not in all of the cases. In fact, there is no clear definition of the right signaling mechanism to use, it just depends. The key is matching your applications to the right choice in signaling mechanism.

At WebRTC Conference and Expo, there will be a dedicated session on the topic of signaling. The speakers will define the choices and how to make decisions and the potential pitfalls of the options. The session will give attendees a clear view of how to decide the best signaling form their applications and deployment environment.

WebRTC Conference and Expo is the place to be to really understand WebRTC and how to implement and deploy applications using this new capability. For developers, WebRTC Conference and Expo has two and a half days of information packed sessions and panels. In addition to the right content, this is an opportunity to meet the standards leaders and leading developers in WebRTC. Planned sessions in the Developer Track are:

  • WebRTC Tutorial and Training
  • Google WebRTC Workshop and Application demonstrations
  • Standards Process and Review
  • Key Standards Issues
  • Making the Right Signaling Choices
  • Using the Data Channel
  • Group Video Chat Issues
  • ICE, STUN, Turn & Security
  • Deploying WebRTC to the Cloud
  • NodeJS and MongoDB
  • Implementing Mobile WebRTC

Edited by Brooke Neuman


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