The Unified Communications vision that is embodied in Office Communications Server 2007 R2 is built on Session Initiation Protocol (SIP), an industry-standard, application-layer signaling protocol that is used for starting, controlling, and ending communication sessions in an IP-based network. SIP is formally described in the Internet Engineering Task Force (IETF) reference specification Request for Comments (RFC) 3261. By using SIP, one user can explicitly invite another to join a conversation or multimedia session. A SIP session begins when the second user accepts a SIP INVITE request.
In Office Communications Server 2007 R2, SIP is used for instant messaging (IM), conferencing, presence subscriptions, video, and Voice over IP (VoIP), providing a common user experience across all these communication modes. Signaling for phone calls coming from the public switched telephone network (PSTN) is converted to SIP by the media (PSTN) gateway.
Although SIP sessions can include the sharing of real-time media, SIP itself does not handle the actual media data, such as audio, video, and application sharing. In practical terms, this separation means that SIP and various media protocols can evolve independently.
Another key protocol is Transport Layer Security (TLS), which enhances security and data integrity for communications over IP networks. By default, Office Communications Server 2007 R2 is configured to use TLS for client-to-server connections. Additionally, Office Communications Server uses mutual TLS (MTLS) for server-to-server connections.
Other key protocols that are used in Office Communications Server 2007 R2 include the following:
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Hypertext Transfer Protocol (HTTP).A standard Internet
protocol that is used for communication between the Focus and
conferencing servers. HTTP is used by the Address Book service,
Group Expansion service, and Device Update service. It is also used
for downloading meeting content to users.
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Centralized Conference Control Protocol (C3P).A custom
protocol for communicating conference creation and control commands
from clients to Office Communications Server 2007 R2. C3P commands
are carried as XML in SIP SERVICE or INFO messages. C3P commands
are also carried over HTTP Secure (HTTPS) to all conferencing
servers.
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PSOM (Persistent Shared Object Model).A custom protocol that
is used for transporting Web conferencing content.
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Secure Real-Time Transport Protocol (SRTP).An IETF standard
protocol that is used for securely transporting audio, video, and
application sharing content to various media devices. It is based
on the RTP protocol, which defines a packet format for carrying
audio and video over IP networks.
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Secure Real-Time Control Protocol (SRTCP).An IETF standard
protocol that is used in conjunction with RTP and SRTP to convey
information about the signal quality of an audio/video (A/V)
conferencing session to various media devices.
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Internet Connectivity Establishment (ICE).An IETF Draft
(soon to be an RFC) that is used in Office Communications Server to
traverse network address translations (NATs) and firewalls with
audio, video, and desktop sharing data. For details, see
ICE Protocol
Upgrade.
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Traversal Using Relay NAT (TURN) and Session Traversal Utilities
for NAT (STUN).Protocols that are used for audio, video, and
desktop sharing data transfer, and for clients and servers to
obtain STUN candidates and allocate TURN candidate transport
addresses on the A/V Edge service for traversal of NATs and
firewalls. STUN and TURN produce the candidates (on which ICE then
performs connectivity checks) to find the most preferred route for
data transmission across the A/V Edge service.
To download a complete list of Office Communications Server
protocols, see