Upperside WebRTC Conference - Mobicents, HTML5 and SIP over WebSockets
Upcoming SlideShare
Loading in...5

Like this? Share it with your network


Upperside WebRTC Conference - Mobicents, HTML5 and SIP over WebSockets






Total Views
Views on SlideShare
Embed Views



6 Embeds 222

http://www.scoop.it 186
https://twitter.com 17
https://si0.twimg.com 8
http://www.linkedin.com 7
http://twitter.com 2
http://www.verious.com 2



Upload Details

Uploaded via as Adobe PDF

Usage Rights

© All Rights Reserved

Report content

Flagged as inappropriate Flag as inappropriate
Flag as inappropriate

Select your reason for flagging this presentation as inappropriate.

  • Full Name Full Name Comment goes here.
    Are you sure you want to
    Your message goes here
Post Comment
Edit your comment

Upperside WebRTC Conference - Mobicents, HTML5 and SIP over WebSockets Presentation Transcript

  • 1. Mobicents, HTML5 WebRTCSIP Over WebSockets Jean Deruelle - TeleStax, Inc12th October 2012, UpperSide WebRTC Conference
  • 2. Questions ??? Dont Wait til the end, interrupt is mandatory !!!
  • 3. HTML5 WebRTC Signaling and Media● WebRTC is independent of WebSockets● Can use anything for signalling including Ajax, server push or plain HTTP● Media is peer to peer and can handle both audio and video
  • 4. SIP Over WebSocketsTypical Flow WebSocket Browser Browser Server HTTP GET HTTP 200 OK SIP REGISTER SIP OK Other server SIP INVITE
  • 5. SIP Over WebSockets Flow Detailedhttp://tools.ietf.org/html/draft-ietf-sipcore-sip-websocket-04 : Still a draft Browser WebSocket ● Regular HTTP request with Server Upgrade header ● Switch to normal mode ○ No HTTP any more, just plain subprotocol ○ ..except its masked so plaintext cant be misinterpreted and avoid security issues ● SIP Messages carried in WebSocket Data ● New SIP Transports : WS or WSS (for Secure using TLS) ○ Addresses advertised by browsers are invalid => literally "df7jal23ls0d.invalid" ○ Via, Contact, everything
  • 6. Peer to Peer ? Browser Another browser Browser to HTTP GET Browser cant be done HTTP 200 OK through HTTP, SIP REGISTER really need a Server ! SIP OK SIP INVITE
  • 7. Use Mobicents asThe Server of Choice● Deliver support for reusable applications that dont care about transport● Applications see the real addresses instead of the invalid ones● Applications can still determine the transport type● Transparent B2BUA, UAC, UAS and Proxy
  • 8. Implemented inside JAIN SIP Stack● Automatically adds WebSocket support to any JAIN SIP based server (SIP Stack used by Mobicents and Google) ○ SIP Servlets http://dev.telestax.com/sipservlets/ ○ JAIN SLEE SIP RA http://dev.telestax.com/jain-slee/ ○ standalone JAIN SIP http://dev.telestax.com/jain-sip/● Doesnt add new dependencies But a huge thank you to Netty.io
  • 9. NAT Concerns● Since the socket is reused there will be no NAT issues when clients are behind the firewall.● If the server is behind firewall its still a bit difficult, but manageable.● The RTP is the most important NAT problem, but it is browser responsibility to fix this ○ They are doing a great job at this ■ STUN/ICE is a built-in and mandatory ■ Chrome to Chrome interop is practically guaranteed
  • 10. Thank you ! http://telestax.com/