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SylkServer
----------
A State of the art, extensible RTC Application Server
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 allows creation and delivery of rich multimedia applications
accessed by SIP Clients, XMPP endpoints and Web applications. The server
supports SIP and XMPP signaling, RTP, MSRP and WebRTC media planes, has
built in capabilities for creating multiparty conferences with Audio and
Video, IM/ File Transfers and can be extended with custom applications by
using Python language.
Deployment Scenarios
--------------------
SylkServer is typically deployed behind a SIP Proxy that is designed to
route the inbound and outbound traffic, handle the authentication,
authorization and accounting.
-SylkServer can also be deployed as multimedia SIP conference server on a
-private network to serve SIP clients on the same LAN by using bonjour mode.
-Blink for MacOSX can be used for automatic discovery of SylkServer instances
-in the neighborhood.
+SylkServer can be deployed as multimedia SIP conference server on a private
+network to serve SIP clients on the same LAN by using bonjour mode. Blink
+for MacOSX can be used for automatic discovery of SylkServer instances in
+the neighborhood.
-SylkServer can also be deployed as a standalone video conference server.
-The client side can be a standalone application (like the companion Sylk
-client) or a standard web browser with WebRTC support.
+SylkServer can be deployed as a standalone video conference server. The
+client side can be a standalone application (like the companion Sylk client)
+or a standard web browser with WebRTC support (like Chrome or Firefox).
Features
--------
SIP Signaling
- TLS, TCP and UDP transports
- INVITE and REFER
- SUBSCRIBE/NOTIFY
- Bonjour mode
NAT Traversal
- SIP Outbound
- ICE clients
- MSRP Relay clients
- MSRP ACM clients
Audio
- Wideband (Opus, G722 and Speex)
- Narrowband (G711 and GSM)
- SRTP encryption (SDES and ZRTP key-exchanges)
- Hold/Unhold
- RTP timeout
- DTMF handling
Video
- H.264 and VP8 codecs
- - SRTP encryption (SDES and ZRTP key-exchanges)
+ - SRTP encryption (SDES and ZRTP key-exchange)
Instant Messaging
- MSRP protocol
- CPIM envelope
- Is-composing
- Delivery reports
File Transfer
- MSRP protocol
- Progress reports
- Conference-info extension
- Conference room persistent
-Conferencing
+Audio/Chat conferencing
- Wideband RTP mixer
- MSRP switch
- XMPP MUC
- Multiparty screensharing
- Conference event package
Video conferencing
- WebRTC
- - Encryption
+ - Encryption (TLS, sRTP)
- Ad-hoc conferencing
- H.264 and VP8 video codecs
- Opus wideband audio
XMPP Gateway
- Server to Server mode
- IM (MSRP sessions and SIP Messages)
- Presence (SIMPLE and XMPP)
WebRTC Gateway
See README.webrtc file.
Applications
------------
When a request arrives at SylkServer, an application is selected depending
on the Request URI. The selection mechanism is described in detail in the
sample configuration file config.ini.sample.
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.
Using an extension to MSRP chat protocol, the server provides also
multi-party screen sharing capability.
XMPP Gateway
SylkServer can act as a transparent inter-domain gateway between SIP and
XMPP protocols. This can be used by a SIP service provider to bridge out to
external XMPP domains or to receive incoming chat messages and Jingle audio
sessions from remote XMPP domains to its local SIP users. In a similar
fashion, a XMPP service provider can use the gateway to bridge out to
external SIP domains and handle incoming chat requestes from SIP domains to
the XMPP users it serves.
A media session or a presence session initiated by an incoming connection on
the XMPP side is translated into an outgoing request on the SIP side and the
other way around. To make this possible, proper SIP or XMPP records must
exists into the DNS zone for the domain that needs the gateway service.
WebRTC gateway
This application can be used to bridge audio/video calls between SIP clients
and Web applications. Any SIP service can be used and a simple to use
client API is provided for developing web pages that include such
functionality. This application supports transparently any audio/video
codec negotiated by the end-points, however WebRTC has standardized
particular codecs for the use on the web, therfore the SIP clients must
support the same set.
See https://webrtc.sipthor.net for a working example.
WebRTC video conference
This application allows WebRTC enabled end-points to organize ad-hoc video
conferences. See Sylk client for an example of how to use this application.
Standards
---------
The server implements relevant features from the following standards:
- SIP (RFC3261) and related RFCs for DNS, ICE and RTP
- MSRP protocol RFC4975
- MSRP relay extension RFC4976
- MSRP File Transfer RFC5547
- MSRP switch RFC7701
- 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
- XMPP core (RFC 6120) http://xmpp.org/rfcs/rfc6120.html
- XMPP extensions http://xmpp.org/xmpp-protocols/xmpp-extensions
- Instant Messaging and Presence http://xmpp.org/rfcs/rfc6121.html
- Interworking between the Session Initiation Protocol (SIP) and the
Extensible Messaging and Presence Protocol (XMPP):
- Presence: RFC7248
- IM: RFC7572
- Chat: RFC7573
- Multi-party chat: RFC7702
- WebRTC standards http://www.w3.org/TR/webrtc/
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
Credits
-------
Special thanks to our sponsors:
- NLnet Foundation http://nlnet.nl
- SIDN Fonds https://sidnfonds.nl
Authors: Saul Ibarra, Tijmen de Mes
Contributors: Denis Bilenko, Dan Pascu
Mentorship: Adrian Georgescu

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