SylkServer
----------

Copyright (c) 2010-2011 AG Projects
http://ag-projects.com

Authors: Adrian Georgescu, Denis Bilenko, Saul Ibarra
Home page: http://sylkserver.com


License
-------

SylkServer is licensed under GNU General Public License version 3. A copy of
the license is available at http://www.fsf.org/licensing/licenses/gpl-3.0.html


Description
-----------

SylkServer is a SIP application server that can be programmed to perform SIP
end-point applications.

The server supports SIP signaling over TLS, TCP and UDP transports, RTP and
MSRP media planes, has built in capabilities for creating ad-hoc SIP
multimedia conferences with HD Audio, IM and File Transfer and can be easily
extended with other applications by using Python language.


Deployment
----------

SylkServer must be typically deployed behind a SIP Proxy as it is not
designed to route packets, handle authorization or accounting.

SylkServer is horizontally scalable with the amount of available hardware by
using SIP Thor, a P2P based self-organizing network technology.


Features
--------

SIP Signaling

 - TLS, TCP and UDP transports
 - INVITE and REFER
 - SUBSCRIBE/NOTIFY
 - Protocol tracing

NAT Traversal

 - SIP Outbound
 - ICE and STUN
 - MSRP Relay
 - MSRP ACM

Audio

 - Wideband (G722 and Speex)
 - Narrowband (G711 and GSM)
 - sRTP encryption
 - Hold/Unhold
 - RTP timeout
 - DTMF handling
 
Instant Messaging

 - MSRP protocol
 - CPIM envelope
 - Is-composing
 - Delivery reports

File Transfer

 - MSRP protocol
 - Progress reports
 - Conference-info extension
 - Conference room persistent

Conferencing

 - Wideband RTP mixer
 - MSRP switch
 - Conference event package
 - Add/remove participants


Applications
------------

Conference

SylkServer allows SIP end-points to create ad-hoc conference rooms by
sending INVITE to a random username at the hostname or domain where the
server runs.  Other participants can then join by sending an INVITE to the
same SIP URI used to create the room.  The INVITE and subsequent re-INVITE
methods may contain one or more media types supported by the server.  Each
conference room mixed audio, instant messages and uploded files are
dispatched to all participants.

One can remove or add participants by sending a REFER method to the
conference URI.

If a participant sends a file to the SIP URI of the room, the server will
accept it, store it for the duration of the conference and offer it to all
participants either present at that moment, or offer it on demand to those
that have joined the conference at a later moment.


Standards
---------

The server implements relevant features from the following standards:

 - MSRP protocol RFC4975
 - MSRP relay extension RFC4976
 - MSRP File Transfer RFC5547
 - MSRP switch draft-ietf-simple-chat-08
 - MSRP Alternative Connection Model RFC6135
 - Indication of Message Composition RFC3994
 - CPIM Message Format RFC3862
 - Conference event package RFC4575
 - A Framework for Conferencing with SIP RFC4353
 - Conferencing for User Agents RFC4579
 * Conferencing for User Agents RFC4579
   5.1  INVITE: Joining a Conference Using the Conference URI - Dial-In
   5.2  INVITE: Adding a Participant by the Focus - Dial-Out
   5.5  REFER: Requesting a Focus to Add a New Resource to a Conference
   5.11 REFER with BYE: Requesting a Focus to Remove a Participant from a Conference


Support
-------

The project is developed and supported by AG Projects. The support is
provided on a best-effort basis. "best-effort" means that we try to solve
the bugs you report or help fix your problems as soon as we can, subject to 
available resources.

To request support you must the use SIP Beyond VoIP mailing list:

http://lists.ag-projects.com/mailman/listinfo/sipbeyondvoip

For commercial support contact AG Projects http://ag-projects.com