‘SIP Trunking’ in Office Communications Server

Author: Russell Bennett

Publication date: May 2008

Product version: Office Communications Server 2007 R2

In the Office Communications Group (OCG) white paper “Integrating Telephony with Office Communications Server 2007” we stated that “SIP Trunking” was out of scope for that release but was “under consideration” for future releases.  We are always reluctant to commit features to releases in advance of the official release announcement; however it is reasonable to say that the process of consideration is underway.

OCS 2007 Voice Connectivity

In Office Communications Server 2007 (OCS) there are 3 modes of voice calling:

  1. OCS point to point: User A calls User B – call is VoIP end to end.
    • The endpoint could be Office Communicator (OC) or an IP phone (e.g. Polycom CX700)
    • Users could be either inside the network or roaming
    • Applies equally to multi-modal communications (i.e. IM, Video, Collaboration)
  2. OCS Federation-a User in domain A calls a User in domain B -   the call is  VoIP end to end over the public internet.
    • Federated calls still carry the same capabilities as point to point  calls described above – roaming, video, etc.
  3. PSTN to OC: a telephony user (PSTN, mobile, PBX) calls an OC user (or vice versa).
    • OCS Co-existence – PBX phone and OC are integrated via “dual-forking” mechanism between PBX and OCS, such as Nortel’s Converged Office
    • OCS Stand-alone – OC accesses telephony via integration using the PBX or a Media Gateway
    • Remote Call Control – OC applies 3rd Party Call Control to a PBX phone

As previously stated, direct interoperability via SIP/RTP with Telephony Service Providers: aka “SIP Trunking” is not supported in OCS 2007.

What is “SIP Trunking”?

What do we mean by “SIP Trunking”?  OCG’s definition was laid out in the white paper referenced above.  The very fact that we needed to define it shows that there are many interpretations of this term.  To further complicate matters, OCG is not the only Microsoft Business Unit engaged in offering this feature – the ResponsePoint group (see: http://www.microsoft.com/responsepoint/default.aspx) is also shipping a SIP Trunk feature in their product and that has a slightly different technical specification to the one we are considering.
Jonathan Rosenberg has formally defined a “SIP Trunk” here: (http://tools.ietf.org/id/draft-rosenberg-sipping-siptrunk-00.txt ) in at least 4 different guises (and who am I to dispute that  ?)  However, as a vendor of equipment that at some time in the future will support this function, it is incumbent on OCG to define what that function might be.
For Microsoft OCG, “SIP Trunking” is the use of SIP and RTP to pass telephony traffic from the enterprise network edge to a network service provider over an IP connection (i.e. without traversing TDM or circuit networks).  For cases where OCS is connecting to a Gateway or IP-PBX, as we qualify through the Unified Communications Open Interoperability Program (http://technet.microsoft.com/UCOIP), we use the term “Direct SIP”

Why SIP Trunking?

The value proposition of SIP Trunking for an OCS customer is:

  1. Not having to deploy, maintain, and operate IP-PSTN gateways on-premise either regionally or at remote offices
  2. Consolidation of data/voice networks (recurring costs)
  3. Reduction of call degradation by reducing the number of conversions of the call from IP to TDM (and back): note that most calls today are carried on a long distance network over IP transports.

The benefit of providing a SIP Trunking feature for Microsoft OCG is:

  • In the short term, offering OCS customers the choice of how to connect to the PSTN
  • The long term value of SIP Trunking is ultimately about creating a roadmap of federated multi-media communications via managed networks

The value proposition of SIP Trunking for Telephony Service Provider is to bring new value to their IP customers and to define a services-based UC value-proposition.  IP-centric service providers, on the other hand, are hoping to access a new channel for their services.

Is SIP Trunking defined in a standard?

A technical recommendation for “IP PBX / Service Provider Interoperability” was created by the SIPconnect technical working group of the SIP Forum in 2006.  In many ways, this was a useful document, but it has not been broadly supported by equipment vendors or the network service providers.  Indeed, the actual uptake of SIP Trunking services has lagged far behind the apparent demand for such a service: the voice traffic traversing a SIP Trunk is currently a tiny proportion of total global trunked traffic.
In an attempt to address the adoption issue, the SIP Forum has launched a new initiative to revise the SIPconnect specification.  The Board of the SIP Forum recognized that Microsoft, as a leading vendor of unified communications solutions, could be a positive proponent of this effort.  In parallel, we realized that the number of service providers (wireline, IP and mobile) around the world was significantly greater than the number of PBX vendors.  We also came to realize that a minority of these Service Providers were currently offering SIP Trunking, and those who did were not necessarily compliant with the SIP RFCs.  Thus, the easiest way for us to address the issue of there being no defacto standard was to work with the SIP Forum to help define a standard that all vendors and service providers can support.
 The natural outcome of this mutual realization was that, at the invitation of the Board of the SIP Forum, Microsoft OCG has submitted a base specification for SIPconnect 1.1 (see: http://www.sipforum.org/component/option,com_docman/task,cat_view/gid,45/Itemid,75/ ) and, at the time of writing, the document has been downloaded 300 times.  As of May 7th, the Technical Working Group has started work on the effort, lead by Rich Shockey of Neustar.  The timelines for completion have not been finalized, but we hope that a final draft of SIPconnect 1.1 will be ready by the end of the year

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