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 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 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-exchange) Instant Messaging - MSRP protocol - CPIM envelope - Is-composing - Delivery reports File Transfer - MSRP protocol - Progress reports - Conference-info extension - Conference room persistent Audio/Chat conferencing - Wideband RTP mixer - MSRP switch - XMPP MUC - Multiparty screensharing - Conference event package Video conferencing - WebRTC media - Encryption (TLS, sRTP) - H.264 and VP8 video codecs - Opus wideband audio - Selective Forwarding Unit 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. SylkServer implements Selective Forwarding Unit (SFU) functionality that uses little resources on the server side allowing for handling much more load than classic MCUs. For scalling up beyond one server AG Projects provides SIP Thor, which provides horizontal scalability for all components present in a RTC platform. Sylk WebRTC client is provided as a sample client and other clients can be developed using sylkrtc API. 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/ - RTP Topologies RFC7667 3.7 Selective Forwarding Middlebox 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