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Thread: VOIP for SILC




VOIP for SILC
user name
2006-07-10 06:35:42
Hello,

I ressurrected the media API and added VOIP support for
SILC.  The patch 
and files are at http://iki.fi/priikone/v
oip/ and include the new media 
API which is based on Gstreamer 0.10.x.  If you want to
commit it, it 
effectively adds secure VOIP to Gaim (with SILC).  I already
sent this 
email yesterday but apparently the attachments were too big
and the email 
didn't get through, so they are at that address now.

Some background on multimedia support in SILC.  In SILC
protocol MIME is 
employed to deliver any kind of media in the network.  It
can deliver 
audio, video and any other type of media that MIME can
represent. 
Because of this it also is able to encapsulate practically
any kind of 
multimedia protocol into SILC.  This means multimedia
sessions can be set 
up with SIP, H.323, Jingle, etc.  We have however selected
as default 
protocol the SDP (Session Description Protocol) to describe
multimedia 
sessions and simple exchange using SILC message packets to
set up the 
session (based on RFC 3264).  The SDP is also used by SIP
protocol.  In 
case of SILC, SDP is the simplest method to set up
multimedia session, and 
it is this protocol that the patch is using.  Personally I
have no plans 
adding SIP or any other support, for now.

The protocol allows setting up four basic types of
multimedia sessions: 
Peer-to-peer with other client, peer-to-server for group
conferencing, 
direct conferencing with client inside SILC network and
group conferencing 
inside SILC network (bypassing NATs).  The patch supports
currently only 
peer-to-peer with another client.  The current SILC Toolkit
doesn't bend 
very well to the others, but this will be rectified in SILC
Toolkit 1.1 
which specifically includes optimizations and features for
multimedia 
applications.  The peer-to-peer was hard enough to glue into
the Toolkit. 
Currently also only TCP is supported, UDP will come later.

It's been tested, but can't hurt by testing more.  Patch
against trunk. 
Tested with Gstreamer 0.10.8, all plugins installed.  The
files voice.[ch] 
and sdpmessage.[ch] go into src/protocols/silc/.  Any chance
this might 
get into 2.0.0?

 	Pekka
____________________________________________________________
____________
  Pekka Riikonen                                 priikone at
silcnet.org
  Secure Internet Live Conferencing (SILC)       http://silcnet.org/


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