This document discusses a Javascript port of the JAIN SIP stack over websockets called Javascript SIP. It was created to allow re-using SIP-based infrastructure in webRTC applications by defining SIP over websockets. The core JAIN SIP classes have been ported by hand to Javascript while maintaining the same architecture, API and naming conventions. This provides a Javascript SIP stack that has been tested with early SIP over websockets implementations. Next steps include adding higher level APIs and optimizations to improve performance and support additional features like IMS/RCS profiles.

1. Javascript SIP
A javascript port of
JAIN SIP stack over websocket
François Joseph Levée, Laurent Strullu, Orange Labs
2012 Mobicents Summit
Thanks to Yuemin Qin

2. Rationales
 WebRTC is gaining momentum
– WebRTC = standardized (W3C and IETF) Real Time Communications
in the browser
– Part of webRTC are already available in Google Chrome (Firefox,
opera, and IE are coming..)
 WebRTC does not standardize signaling….
– left to the developer’s choice : JSON, XMPP, SIP, …over HTTP, XHR,
WS…
– WebRTC focus on Media Path (API and protocols): codecs, NAT,
SDP…
 …but re-using already in place SIP based
infrastructure in webRTC would be definitively cool!
– need definition of usage of SIP over a web transport: SIP over WS
IETF draft
– need a Javascript SIP Stack based on a large community  Here we
are with Javascript JAIN SIP!
– need support of WS in SIP components (see Jean and Vladimir
2
presentation tomorrow) libre
diffusion Orange JAIN SIP Javascript port
– need a GW to bridge to already deployed SIP networks and access

3. WebRTC in a nutshell
SIP
SRTP,
Multiplexing,
ICE…
3 diffusion libre

4. Where are we? Availa
ble
 All core JAIN SIP classes today !
have been ported by !
hand (no ugly tool),
 JAIN SIP stack
architecture, API and
naming convention
 used, CORE (RFC3261) implementation supported
all SIP
 Tested with Mobicents SIP over WS early
implementation
 ~550Kb (compressed)
 not fully optimized yet (see next steps)
 no SDP Javascript library available (but SDP handling is
part of PeerConnection API)
 http://code.google.com/p/jain-sip/source/checkout?
4 repo=javascript
Javascript SIP diffusion libre Orange JAIN SIP Javascript port

5. Example
 JAIN SIP Client sample : SIP INVITE request creation and
emission for a JAIN SIP ware developer ;-)
 Higher level API for web developer is planned as
next step
5 diffusion libre Orange JAIN SIP Javascript port

6. Default application
• A basic WebRTC JAIN SIP softphone WebApp can be found
in the project Google code repository:
~50
lines
= of
code!!
 This application will be the default application in
MSS2.0, allowing to do SIP calls without a native SIP
phone !
6 diffusion libre Orange JAIN SIP Javascript port

7. Next Steps
 Very Short term (Q4 2012)
– Provide a higher level Communication Framework
– New features:
– Add WSS support (required for security and proxy
traversal)
– Add Keep Alive (RFC5626) (required for server side
connectivity info)
– Optimizations:
– multithreading using web workers for message parsing
– more library size minimization
 Short term (Q12013)
– Add IMS/RCS profile
7 diffusion libre Orange JAIN SIP Javascript port

8. References
 rtcweb IETF WG: http://tools.ietf.org/wg/rtcweb/
 WebRTC W3C: http://www.w3.org/TR/2012/WD-webrtc-
20120821/
 www.webrtc.org
 Javascript JAIN stack port:
http://code.google.com/p/jain-sip/source/checkout?
repo=javascript
8 Javascript SIP diffusion libre Orange JAIN SIP Javascript port

9. Questions ?
9 Javascript SIP diffusion libre

10. Thank you !