I was very often asked what e.g. is the meaning of STUN or RT.
Also, which RFC is behind this protocol. If further information are available, I have posted this info too.
Therefore, here it come:
STUN - (Simple Traversal of User Datagram Protocol (UDP) - Through Network Address Translators (NATs))
This is protocol used on the Edge server, where UDP data is passed through the NAT. It contains information about the external (public) IP address where the client is hidden behind and the internal (private) IP address the client has assigned.
STUN (Session Traversal Utilities for NAT)
URI Scheme for the Session Traversal Utilities for NAT protocol
NAT Behavior Discovery Using Session Traversal Utilities for NAT (STUN)
TURN - Traversal Using Relay NAT
TURN is a design part of the ICE process, but can be used also without ICE. It is responsible for NATed client supporting "direct" communication. in Lync/ Skype for business, this protocol is server related.
ICE - Interactive Connectivity Protocol
IT determines all possible UDP and TCP port involved in a SIP communication. Necessary for the client negotiation process which is the best possible path for communication. This protocol is client related. But need ICE award servers.
MRAS - Media Relay Authentication Service
This is an authentication protocol used with Lync and Skype for business. MRAS initiates Token for authentication. It can be seen more as a component, rather then a protocol. It involves in the SIP authentication.
I haven't found official information about the MRAS Server/ Service, but it is most best describe the Audio/ Video Authentication description.
PSOM - Shared Object Messaging Protocol
It a Microsoft proprietary protocol used for Web Conferencing. PSOM is the media protocol for data collaboration. PSOM will use TLS as the underlying transport. PSOM can be used by conferencing clients to establish media channels with the Web Conferencing Server to negotiate or transfer media.
C3P - Centralized Conference Control Protocol (CCCP)
The Centralized Conference Control Protocol (C3P) activates, modifies, deactivates, and controls conferences. It utilize SIP standards for conferencing.
RTP/ RTCP - Real-Time Transport Protocol
RTP/RTCP is the standard protocol for the transport of real-time data, including audio and video.
SRTP - Secure Real-Time Transport Protocol
SIP - Session Initiation Protocol
Session Initiation Protocol (SIP) is the industry standard protocol described in IETF RFC 3261 that defines a standard way for session setup, termination, and media negotiation between two parties. It is widely used for Voice over IP (VoIP) call signaling.
SDP - Session Description Protocol
Session Description Protocol (SDP) is the industry standard protocol described in IETF RFC 4566 that defines a standard way to convey media details, transport addresses, and other session description metadata to the participants when initiating multimedia teleconferences, Voice over IP calls, streaming video, or other session
MTLS is nearly the same as TLS, but can contain multiple session with in a TLS connection setup. That's why Lync and Skype for Business use it between the Server-to-Server communication.
General information:Signaling and Control Protocol
SIP, as specified in RFC 3261, is used for session setup and termination in Office Communications Server. SIP messages use TCP or TLS as the underlying transport layer for client-to-server communications and TLS with mutual authentication (MTLS) for server-to-server communications. Conferences and call control are established within the context of existing SIP sessions using C3P protocol. C3P commands are sent using SIP INFO messages. A separate SUBSCRIBE/NOTIFY dialog is used to subscribe to conference packages, state change notifications, and the conference participant list.
The Web Conferencing Server uses PSOM as the media protocol for data collaboration. PSOM uses TLS as the underlying transport. As the client for the Web Conferencing Server, Live Meeting functionality also relies on PSOM.
RTP and RTCP are used to provide audio/video functionality. Secure Real-time Transport Protocol (SRTP) and Secure Real-time Transport Control Protocol (SRTCP) are used to provide secure, encrypted audio/video functionality.) RTP/RTCP uses TCP or User Datagram Protocol (UDP) as the underlying transport.
RTA/RTAudio - Realt-Time Audio
RTV/RTVideo - Real-Time Video
G711/ G729/ G722
This is the SKYPE codec used with Skype and Skype for Business. The new version is only used between clients and client to server. The Mediation Server in Skype for Business will not make use of SILK.
A new codec with will be used also in the open communication program with Polycom new phone. But will not be supported with in Skype for Business.