SylkServer SIP/XMPP Gateway¶
Interoperability between XMPP and SIP protocols has become a necessity given that all closed and public Internet Services for Real Time Communications have migrated to a variant of these protocols, if not internally at least at their borders. Both protocols share a common feature-set, the most important one being the usage of Internet identifiers in the form of user@domain and replace the need for classic telephone numbers.
Various projects in the open source domain emerged to tackle the interoperability between XMPP and SIP but all of them focused on one-to-one Presence signaling or VoIP sessions, but never in the context of rich media interoperability and using standard user@domain Internet addressing.
SylkServer SIP/XMPP gateway application goal is to cover this gap and provide an Open Source implementation for interoperability of all relevant media types and multimedia conferencing for them. We consider the following as relevant media: Wide-band audio, Session based IM, File Transfer, Presence and Multiparty Conferencing facilities for all of them. We consider Screensharing and Video beyond the basic need of interoperability due to obvious reasons related to codecs, scalability or lack of standards.
The goal is to implement a bidirectional signaling and media gateway for Presence, Session based Instant Messaging, Multiparty Chat, Audio, and File Transfer for SIP and XMPP signaling protocols, RTP and MSRP media planes. This software will work with ZERO configuration, a simple command like "apt-get install sylkserver" will be enough for deployment of the gateway on a Debian Linux server.
This research and development is part of the WormHole Project sponsored by NLNet and AG Projects.
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).
Specifications and Standards¶
SIP Signaling and Media¶
- SIP (RFC3261) and related RFCs for DNS, SDP, ICE, RTP (too many to mention)
- MSRP Protocol http://tools.ietf.org/html/rfc4975
- MSRP Relay extension http://tools.ietf.org/html/rfc4976
- MSRP Alternative Connection Model (ACM) //tools.ietf.org/html/rfc6135
- Multi-party Chat Using MSRP http://tools.ietf.org/html/draft-ietf-simple-chat-14
XMPP Signaling and Media¶
- 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
- Multi-User Chat http://xmpp.org/extensions/xep-0045.html
- File Transfers http://xmpp.org/extensions/xep-0096.html
- Audio and Video Jingle http://xmpp.org/extensions/xep-0166.html
- Audio and Video Jingle RTP http://xmpp.org/extensions/xep-0167.html
- Audio and Video Jingle ICE (UDP) http://xmpp.org/extensions/xep-0176.html
- Delivering Conference Information to Jingle Participants http://xmpp.org/extensions/xep-0298.html
Protocol Translation Specifications¶
- XMPP - SIP Core Interworking
- XMPP - SIP Presence
- XMPP - SIP Message
- XMPP - MSRP Chat
- XMPP - MSRP Group Chat
Lessons learned during implementation and Suggestions for improvement of Protocol Translation Specifications
Completed items are operational on SIP2SIP which supports Chat and Presence to gmail.com an jabber.org domains and audio works with jit.si accounts. Progress announcements are made on SIP beyond VoIP mailing list
- Software Library Analysis
- Gateway Design
- SIP-XMPP Chat
- SIP-XMPP Presence
- SIP-XMPP Multiparty Chat
- SIP-XMPP Jingle Audio