SylkServer

h1. Project Moonshot (a.k.a. WebSylk)

Adding WebRTC and multiparty video capability to SylkServer. This will allow anyone with basic computer knowledge to setup a zero-configuration meeting point that can be used to join a multi-party conference using standards-based clients like SIP, XMPP, IRC and Web-browsers. Also, it will allow third-parties to develop their own front-ends based on a web based API.

(Items in this list are not necessarily in order)

h2. Add VP8 support to SIP SIMPLE Client SDK

WebRTC mandates implementing both VP8 and H264, but the H264 part is getting quite some resistance. The current state (as of 12th of May 2015) is: Chrome does not implement H264, and in Firefox it has to be manually enabled through the advanced configuration menu. We need to add support for it to the SDK so we can transparently interoperate with WebRTC endpoints without the need for transcoding. Apparently libvpx already has VP9 support, which is currently behind a flag in Chrome, it's probably a good idea to try to add it too.

H2. Adding support for "transparent streams" to SIP SIMPLE Client SDK.

h2. Extend SylkServer with a web framework

Using such framework any web based application can tap into. Different applications might need to serve web content: screensharing images, conference web page, WebRTC endpoint, etc. There should be a single HTTP(S) endpoint where applications tap into and serve content.

Example URLs:

https://conference.sip2sip.info/sylk/conference/screen-sharing/

https://conference.sip2sip.info/sylk/conference/

https://conference.sip2sip.info/sylk/webrtcgateway/

When staring a conference room, SylkServer will advertise the following entry points:

 * A SIP address (all media possible)

 * An optional phone number (audio only)

 * An  XMPP address (chat and optional audio/video)

 * A WEB URL (all media is possible)

 * IRC room (text only)

h2. Design WebSylk

The WebSocket API for gatewaying WebRTC endpoints to the SIP world. It needs to cover:

h3. Phase 1

* Registration of multiple accounts

* Multiple sessions

* Audio / video sessions

h3. Phase 2

* Chat

* Presence

* Contacts management

h3. Phase 3

* File transfer

* Screen sharing

* Documents shared editing (may need additional resources)

h2. Implement the gateway skeleton (first alpha release)

It needs to cover:

* Multiple accounts

* Single session per account

* Audio and video

h2. Gateway phase 2

* Chat

* Presence

* Contact management

h1. Other

Random notes, thoughts.

h2. Write our own Janus SIP plugin

Based on the existing one, but using PJSIP instead. It will become clearer once we are further in the project. We'd use just the SIP signaling part, not any media capability, that's taken care of.

The WebSylk API would not change, but it should facilitate some features such as multiple sessions per account.

h2. Multi-party video

There is a Janus plugin for that, maybe we can leverage it. Adding support for this to our SDK is a no go, in it's current state. Basically we'd need a Session which doesn't really handle media, but it takes some SDP that got passed to it and deals with it. We'd use multiple Video streams in a single session, and some way to correlate them, probably piggibacking on what WebRTC already does with BUNDLE, for example. Not sure if Janus supports simulcast. (sending one big picture and the rest as thumbnails)

h2. Streaming

This can probably be offloaded to Janus, we will contribute the patches upstream.