Presentation slides for IPTcomm 2011 conference.
Abstract:
Existing standard protocols for the web and Internet telephony fail
to deliver real-time interactive communication from within a web
browser. In particular, the client-server web protocol over reliable
TCP is not always suitable for end-to-end low latency media path
needed for interactive voice and video communication. To solve
this, we compare the available platform options using the existing
technologies such as modifying the web programming language
and protocol, using an existing web browser plugin, and using a
separate host resident application that the web browser can talk to.
We argue that using a separate application as an adaptor is a
promising short term as well as long term strategy for voice and
video communications on the web.
Our project aims at developing the open technology and sample
implementations for web-based real-time voice and video
communication applications. We describe the architecture of our
project including a RESTful web communication API over HTTP
inspired by SIP message flows, a web-friendly metadata for
session description, and an UDP-based end-to-end media path.
All other telephony functions reside in the web application itself
and/or in web feature server. The adaptor approach allows us to
easily add new voice and video codecs and NAT traversal
technologies such as Host Identity Protocol. We want to
make web-based communication accessible to millions of web
developers, maximize the end user experience and security, and
preserve the huge investment in and experience from SIP systems
while adhering to web standards and development tools.
5. Modify browser Use existing plugin
Extend web protocols/languages Most existing web communication
1. Include SIP/RTP stack systems use Flash Player
2. Add device access, codecs and
e2e transport (IETF, W3C)
Web server proprietary
HTTP server
HTML HTML
HTML HTML FP FP
UDP proprietary
over UDP
Available Options
Build new plugin Use separate app
Just handles missing pieces Runs as a separate process/service
(device access, codec, transport)
Web
HTTP server
HTTP Web
server HTML
HTML HTML HTML HTML
plugin plugin
UDP UDP
App App
6. Modify browser Use existing plugin
No other dependencies Ubiquitous availability
Eventually a standard Browser agnostic
Numerous web developers Rich developer tools and experience
Reluctance to change One-to-one as well as group
Portable device access/sharing Transport is not enough (for SIP/RTP)
Time to ubiquitous availability Cannot install new codecs
Depends on vendor for updates
Available Options
Comparison Use separate app
1. With existing technologies Browser and app agnostic
2. Emerging standard protocols Any transport, language, codecs.
3. Allow walled garden Persistent/long lived state
4. Require new install Yet another install, slow adoption
5. App dies on page close Security and access control
6. Re-use web security means Video display needs plugin
7. Our Project
(1) Signaling API
Resource-based, SDP=>XML/JSON, Apache web server,
subscribe/notify, long-lived connection, PHP websocket server,
persistent vs transient data, access control MySQL database
DB
(2) Communication Widgets
Click to call, contact list,
conference object
(3) Media Application API
Transport, auth and media objects
8. Summary
Opportunities Available Options Our project
Motivation 1. Modify browser Signaling API
to move VoIP to web 2. Use existing plugin Communication widgets
3. Build new plugin Media path
4. Use separate application Demonstration
Project: http://sites.google.com/site/vvowproject/
Source: http://code.google.com/p/vvowproject/
Demo: http://gardo1.rice.iit.edu/webconf/
