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.
In Office Communications Server 2007 (OCS) there are 3 modes of voice calling:
As previously stated, direct interoperability via SIP/RTP with Telephony Service Providers: aka “SIP Trunking” is not supported in OCS 2007.
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”
The value proposition of SIP Trunking for an OCS customer is:
The benefit of providing a SIP Trunking feature for Microsoft OCG is:
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.
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