Upperside WebRTC Conference - Mobicents, HTML5 and SIP over WebSockets

telestax
telestaxOpen Source Cloud Communications
Mobicents, HTML5
      WebRTC
SIP Over WebSockets
     Jean Deruelle - TeleStax, Inc
12th October 2012, UpperSide WebRTC
              Conference
Questions ???


   Don't Wait 'til the end, interrupt is mandatory !!!
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
SIP Over WebSockets
Typical Flow

                           WebSocket
  Browser                                Browser
                            Server

            HTTP GET


            HTTP 200 OK



            SIP REGISTER


            SIP OK                     Other server



            SIP INVITE
SIP Over WebSockets
 Flow Detailed
http://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 it's masked so plaintext
                                                                 can't 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
Peer to Peer ?

  Browser                 Another browser
                                            Browser to
            HTTP GET                        Browser can't
                                            be done
            HTTP 200 OK
                                            through HTTP,
            SIP REGISTER
                                            really need a
                                            Server !
            SIP OK


            SIP INVITE
Use Mobicents as
The Server of Choice
● Deliver support for reusable applications that
  don't 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
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/
● Doesn't add new dependencies But a huge thank you to
  Netty.io
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 it's 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
Thank you !


 http://telestax.com/
1 of 10

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Upperside WebRTC Conference - Mobicents, HTML5 and SIP over WebSockets

  • 1. Mobicents, HTML5 WebRTC SIP Over WebSockets Jean Deruelle - TeleStax, Inc 12th October 2012, UpperSide WebRTC Conference
  • 2. Questions ??? Don't 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 WebSockets Typical Flow WebSocket Browser Browser Server HTTP GET HTTP 200 OK SIP REGISTER SIP OK Other server SIP INVITE
  • 5. SIP Over WebSockets Flow Detailed http://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 it's masked so plaintext can't 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 can't be done HTTP 200 OK through HTTP, SIP REGISTER really need a Server ! SIP OK SIP INVITE
  • 7. Use Mobicents as The Server of Choice ● Deliver support for reusable applications that don't 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/ ● Doesn't 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 it's 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/