This wiki page is meant for providing technical documentation and access to the version control source code repository.
SylkServer official web site is http://sylkserver.com
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 and MSRP media planes, WebRTC support and has built in capabilities for creating multiparty conferences with wideband Audio, IM and File Transfers and can be extended with other applications by using Python language.
SylkServer was born from the need of realizing ubiquitous connectivity between SIP and XMPP protocols without having to develop a multi-stack client. End-users typically use both type of clients for their best features. SIP and XMPP protocols have in common the same addressing scheme user@domain email style addresses. Using SylkServer, the best client of either protocol can be used to reach clients using the other protocol without having to configure anything in the network or clients. We call this function SIP/XMPP gateway, which is standardized by the XMPP foundation.
A complete description of the SIP/XMPP gateway standards and SylkServer design and implementation can be found here:
Going beyond simple translation of media, SylkServer is able to create multiparty conferences with clients from both worlds. Using SylkServer ad-hoc multiparty conferencing capabilities, both SIP and XMPP client can connect to the same address and exchange the participants list, have text chat and wide-band audio. Again, there is no need to configure anything to realize such complex task.
SylkServer is an application server that behaves like a SIP end-point and therefore must be deployed behind a SIP Proxy/Registrar server for registration, proxy, presence agent, accounting and authorization functions. The SIP Proxy must be configured to route certain flows to SylkServer and handle incoming requests from it as necessary.
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For the complete feature list look on the features page
- TLS, TCP and UDP transports
- INVITE and REFER methods
- DNS and Bonjour discovery
Voice over IP¶
- Wideband (Opus, G722, Speex)
- Narrowband (G711, GSM, iLBC)
- Encryption (sRTP)
- Chat Sessions (MSRP)
- Composing Indication
- Delivery reports
- Wideband audio
- IM and File Transfers
- Participants Info
- Presence and IM
- Multi-party conferencing (MUC)
- Jingle audio (RTP)
- SIP account registration
- Audio / video calling
- Codec agnostic