Applications
Sylk Suite consists of different software packages. This page describes them as well as the inbuild applications in Sylk Server.
Bundled applications in Sylk Server
WebRTC
The webrtc application has multiple components. To develop a web endpoint for this application, you must use sylkrtc.js library. The functions for the application are:
Conferencing
This component allows WebRTC enabled browsers to organize ad-hoc video conferences including screen and file sharing.
Sylk Server 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.
Support media and features: audio/video, file-sharing, chat, screen-sharing, remote speaker and mute control and account-less anonymous join using web link.
The backend is provided by Janus server.
An overview about this application is available here
SIP/WebRTC gateway
This component can be used to bridge one-to-one audio and video calls between SIP clients and WebRTC endpoints.
Any SIP service and SIP compatible device can be used for the SIP side.
The application supports transparently any audio/video codec negotiated by the end-points.
See https://webrtc.sipthor.net for a working example.
Screensharing is also supported depending on the web browser capabilities.
SIP/WebRTC Messaging Server
Sylk Server provides online and offline messaging services for standard SIP end-points and WebRTC clients that support SylkRTC API.
You can find the details in the SIP/WebRTC Messaging documentation.
SIP conference
Sylk Server allows SIP end-points to create ad-hoc conference rooms by sending INVITE with SDP content 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 servers is interoperable with XMPP MUC (see SIP/XMPP gateway section below).
A detailled description can be found in the SIP Conference documentation.
SIP/XMPP gateway
This application implements all relevant standards produced by STOX IETF Working Group that has been chartered to standardise SIP/XMPP protocol interoperability.
Audio, Chat and Presence media are supporeted including the multi-party scenario.
By pointing the correspondent DNS records for SIP or XMPP services of a given Internet domain to the address of this gateway, any packet sent to or connection established to the gateway by one of the protocols is transparently translated into the other. The server is designed in such way that it requires zero-configuration (except of course for the DNS domains configuration). The domain must be set in xmppgateway.ini configuration file.
For more information see SIP/XMPP gateway documentation
IRC gateway
This application can be used to bridge from SIP and XMPP worlds to IRC conference rooms.
Echo
The echo application can be used to test audio calls, incoming audio is sent back to the caller with a small delay.
Playback
Sylk Server can be set to answer calls for certain addresses, play a wav file and the hangup. Video playback is also supported using H.264 and VP8 codecs.
External applications
Mobile Push notifications
Sylk Pushserver was designed to act as a central dispatcher for mobile push notifications inside RTC provider infrastructures. Both the provider and the mobile application customer, in the case of a shared infrastructure, can easily audit problems related to the processing of push notifications.