diff --git a/README b/README index c3a7959..3786e76 100644 --- a/README +++ b/README @@ -1,104 +1,104 @@ SylkServer ---------- Copyright (c) 2010-2011 AG Projects http://ag-projects.com Authors: Adrian Georgescu, Denis Bilenko, Saul Ibarra 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/agpl-3.0.html Description ----------- SylkServer is a state of the art, extensible SIP Application Server SylkServer allows creation and delivery of rich multimedia applications accessed by SIP User Agents. The server supports SIP signaling over TLS, TCP and UDP transports, RTP and MSRP media planes, has built in capabilities for creating ad-hoc SIP multimedia conferences with Wideband Audio, IM and File Transfer and can be easily extended with other applications by using Python programming language. Features -------- SIP Signaling - TLS, TCP and UDP transports - INVITE and MESSAGE for sessions - SUBSCRIBE/NOTIFY Audio - Wideband codecs (G722 and Speex) - Narrow-band codecs (G711 and GSM) - sRTP encryption - RTP timeout - DTMF handling Instant Messaging - MSRP chat and SIP MESSAGE - CPIM envelope - Is-composing indicator - History buffer Conferencing - RTP wideband mixer - MSRP Switch with private messgaing - Conference event package - isfocus support Built-in Applications --------------------- 1. 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. SIP end-points that do not support MSRP chat can join the bridge by using audio only, they will receive the chat messages over the SIP signaling using SIP MESSAGE method, which is supported by many legacy end-points. Messages sent to the room using SIP MESSAGE will be dispatched by either SIP MESSAGE or through established MSRP sessions depending on how the end-points have joined the room. 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 later to those that have joined the conference at a later moment. Standards --------- The server implements relevant features from the following standards: * MSRP and its relay extension RFC4975, RFC4976 - * File Transfer over RFC5547 + * File Transfer over MSRP RFC5547 * 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 5.1. INVITE: Joining a Conference Using the Conference URI - Dial-In * MSRP switch draft-ietf-simple-chat-07