SIP Trunking: What Is It? Why Do I Need It? How Do I Deploy It?
Published May 20 2019 02:17 PM 2,502 Views
Brass Contributor
First published on TECHNET on Jun 13, 2010

A SIP trunk is a direct connection between your organization and an ITSP. It enables you to extend VoIP telephony beyond your organization's firewall without the need for an IP-PSTN gateway. Additionally, SIP trunks can carry instant messages, multimedia conferences, user presence information, Enhanced 9-1-1 (E9-1-1) emergency calls, and other SIP-based, real-time communications services. Because it is easier to configure and less expensive to design, operate, maintain, and upgrade, and because ITSPs deliver services at substantial savings, your investment in SIP trunking can give a quick and substantial return on investment.

Author: Dr. Rez

Original publication date: February 2010

Product version: Office Communications Server 2007 R2



What Is It? Why Do I Need It? How Do I Deploy It?

A SIP trunk is a direct connection between your organization and an Internet telephony service provider (ITSP). It enables you to extend voice over IP (VoIP) telephony beyond your organization's firewall without the need for an IP-PSTN gateway. This simpler configuration is easier and less expensive to design, operate, maintain, and upgrade. And because ITSPs deliver services (notably long distance) at substantial savings, your investment in SIP trunking can give a quick and substantial return on investment.

In addition to VoIP calls, SIP trunks can also carry instant messages, multimedia conferences, user presence information, Enhanced 9-1-1 (E9-1-1) emergency calls, and other SIP-based, real-time communications services. Microsoft Office Communications Server 2007 R2 users can place local and long distance calls with caller ID and call hold over SIP trunks. As SIP-based communication technologies advance and SIP trunking becomes commonplace, additional services are likely to become available.

With all of the benefits that SIP trunking provides (especially the substantial cost savings), it's no wonder that SIP trunking is a hot topic in unified communications. This article gives an overview of this popular new technology and its benefits, information about how it's deployed, and answers to questions about SIP trunking that people frequently ask Microsoft product managers and engineers.

What is SIP trunking?

This paper uses SIP trunking to describe using the Session Initiation Protocol (SIP) and Real-Time Protocol (RTP) to connect Office Communications Server 2007 R2 directly to an ITSP over an Internet Protocol (IP) connection for the purpose of originating and terminating telephone calls.

SIP is used to create and control the communications sessions that are the basis of VoIP telephony. SIP is also used for instant messaging, presence updates, multimedia, conferencing, and other real-time services that can traverse a SIP trunk. RTP manages the actual voice data in a VoIP calls.

The "trunk" in SIP trunking is a term from circuit-switched telecommunications. It refers to a dedicated physical line that connects switching equipment. A SIP trunk is similar to its circuit-switched predecessor in that it is a connection between telecommunications systems, but it is different because the connection it describes is a virtual connection. This connection can be made over a line that is used only for SIP trunking, over a dedicated line that carries SIP trunking with other IP traffic, or over the Internet on a virtual private network (VPN).

Why do I need SIP Trunking? What are the benefits?

Deploying SIP trunking can be a big step towards simplifying your organization's telecommunications and towards preparing for the latest real-time communications enhancements, but the biggest motivation for most organizations is immediate and substantial cost savings.


  • Save money on long distance service
    Long distance service typically costs significantly less with a SIP trunk connection.

  • Eliminate IP-PSTN gateways (or even your entire PBX)
    Because SIP trunks connect directly to your ITSP without traversing the publicly switched telephone network, you can get rid of IP-PSTN gateways and their attendant cost and complexity. Some ITSPs will even host a PBX for you, taking over both the PBX hardware and user administration if you choose, with substantial cost savings from reduced complexity, maintenance, and administration.

  • Eliminate a redundant network
    Deploying SIP trunking is a logical step towards the goal of having a single, IP-based network, rather than redundant telephone and data networks.

  • Eliminate BRI and PRI subscription fees
    With a SIP trunk connected directly to an Internet telephony service provider, you can dispense with costly BRIs and PRIs, replacing them with service that can cost significantly less. Furthermore, with SIP trunking you don't need to buy lines in blocks of 24 or 32. Instead, you can buy the bandwidth you need, in smaller increments, and at better prices.


  • Extend the capabilities of Office Communications Server with new services from ITSPs
    With a SIP trunk in place, you can extend existing capabilities with additional services, such as E9-1-1 emergency calling. In the future, we expect ITSPs will to new services, such as greater integration with mobile phones and presence information on devices that are not running Office Communicator.



How Do I Deploy SIP Trunking?

A SIP trunk can connect your organization to an ITSP through either an IP-PBX or a Mediation Server. A Mediation Server performs encrypting, decrypting, and data translation between an Office Communications Server deployment on one side and the world beyond on the other.

This paper addresses only connecting through a Mediation Server. For details about deploying a SIP trunk with an IP-PBX, refer to the documentation provided by your PBX manufacturer and your ITSP.

Deploying and Configuring a Mediation Server with SIP Trunking

Each Mediation Server has two interfaces (network adapters): an internal interface and an external interface. The external interface is commonly called the gateway interface because until recently it was usually connected to an IP-PSTN gateway (or an IP-PBX). When you deploy SIP trunking, however, the external interface is connected to the SIP trunk.

Deploying and configuring a Mediation Server is discussed in the Deploy a Mediation Server section of the Office Communications Server 2007 R2 documentation at http://go.microsoft.com/fwlink/?linkid=161881 . For the procedure for deploying a Mediation Server with SIP trunking, see Deploying SIP Trunking at http://go.microsoft.com/fwlink/?linkid=178627 .

Determining which Type of Connection to Use for your SIP Trunk

A SIP trunk can be deployed over the Internet or over a line that is dedicated to your organization and that may or may not share bandwidth with other IP traffic. A dedicated line (sometimes called a "leased line"-typically a T1 or fiber optic line) between your organization and the ITSP usually costs the most, but it can typically carry the most simultaneous calls and offers the highest security and reliability. If your organization has lower call volumes or less stringent security and availability requirements, another connection type might make sense for you. The following table highlights the pros and cons of each connection type.

Table 1. Comparison of SIP Trunking Connection Types

























Connection type



Advantages



Disadvantages



Dedicated line ("leased line") with no other traffic



Most reliable


More secure


VPN not required


Highest call-carrying capacity



Most expensive



Dedicated line ("leased line") shared with other IP traffic, often using Multiprotocol Label Switching (MPLS)



More secure


VPN not required



Excessive IP traffic can interfere with VoIP operation unless VoIP traffic is given priority



Public (Internet)



Least expensive



Least secure


VPN required


Least reliable


Lowest call-carrying capacity


For details about the security implications of the connection type that you choose, see SIP Trunking Drilldown: Security Considerations at http://go.microsoft.com/fwlink/?linkid=180918 .

Recommendations for Setting up SIP Trunking Network Connections

To improve security, we recommend the following practices when you set up SIP trunking network connections with Office Communications Server 2007 R2:


  • Set up a VLAN with static routing between the Mediation Server and the router. If you use a VPN server, set up a VLAN between the VPN server and the Mediation Server.

  • Do not allow broadcast or multicast packets to be transferred from the router to the VLAN. If you use a VPN server, do not allow broadcast or multicast packets to be transferred from the VPN server to the VLAN.

  • Block any routing rules that route traffic from the router to anywhere but the Mediation Server. If you use a VPN server, block any routing rules that route traffic from the VPN server to anywhere but the Mediation Server.

  • If you use a VPN server, encrypt data on the VPN by using Generic Routing Encapsulation (GRE).

Frequently Asked Questions

Microsoft product managers and engineers frequently hear the following questions about SIP trunking.

Can Office Communications Server 2007 R2 interoperate with SIP trunking providers?

Given the lack of widely supported certification and testing programs for SIP trunking, Microsoft has developed its own testing and certification program to ensure interoperability between Office Communications Server 2007 R2 and SIP trunking providers. For details about the program, see the Microsoft Unified Communications Open Interoperability Program Web site at http://go.microsoft.com/fwlink/?LinkId=178632 .

Which ITSPs are qualified for Office Communications Server 2007 R2?

The SIP Trunking Services Qualified for Microsoft Office Communications Server 2007 R2 Web page at http://go.microsoft.com/fwlink/?LinkId=178632 lists the ITSPs that have been independently qualified to meet the requirements of the Unified Communications Open Interoperability Program.

Do I need to deploy a session border controller in front of my Mediation Server before I set up SIP trunking?

Session border controllers (SBCs) should not be deployed in front of Mediation Servers, because they can prevent Office Communications Server from working properly. SBCs are usually deployed for added security, especially when sending and receiving data beyond the corporate network. However, when an SBC is placed in the data path of Office Communications Server, it can interfere with end-to-end data integrity and prevent communications from being established.

If your SIP connection will use a dedicated line, you should not deploy session border controllers. They do not enhance security, because the line is not publicly accessible, and the security features of the session border controller can interfere with end-to-end data integrity and can prevent communications from being established.

If your SIP connection will use the Internet, you can install session border controllers for their VPN and tunneling capabilities, but you should not enable their security features for the reasons discussed previously.

Is a specific VPN server required to connect Office Communications Server 2007 R2 to a SIP trunk over the Internet?

No. You can use whichever VPN server you prefer.

Which codecs are supported for SIP trunking with Office Communications Server 2007 R2?

The G.711 codec (a narrow band, 8-bit codec) is supported.

How much bandwidth do I need for a SIP trunk?

The bandwidth required for a SIP trunk depends primarily on the number of concurrent calls that it has to handle. The basic bandwidth calculation is:

SIP Trunk Peak Bandwidth = Maximum Simultaneous Calls × 80kbps

Performing a usage survey is the best way to determine required bandwidth.

For details, see SIP Trunking Drilldown: Bandwidth Considerations at http://go.microsoft.com/fwlink/?LinkId=180963 .

Is SIP trunking supported on Office Communications Server 2007?

No. SIP trunking support was introduced in Office Communications Server 2007 R2.

What features are supported with SIP trunking on Office Communications Server 2007 R2?

Office Communications Server 2007 R2 supports local and long distance telephone service, caller ID, and call hold over SIP trunks that are provided by ITSPs approved by the Microsoft Unified Communications Open Interoperability Program at http://go.microsoft.com/fwlink/?LinkId=178633 . For details about supported SIP trunking scenarios with Office Communications Server 2007 R2, see SIP Trunking Drilldown: Supported Scenarios at http://go.microsoft.com/fwlink/?LinkId=180965 .











Note:
If you have questions about the Microsoft Unified Communications Open Interoperability Program, you can contact the program administrators at msucoip@microsoft.com .

If I deploy SIP trunking with Office Communications Server 2007 R2, will that deployment be compatible with future releases of Office Communications Server?

Microsoft plans to make SIP trunking in Office Communications Server 2070 R2 compatible with SIP trunking in future releases.


Additional Resources and References

Help Topics at the Microsoft TechNet Library

Microsoft Unified Communications Partner Information


  • SIP Trunking Services Qualified for Microsoft Office Communications Server 2007 R2 at http://go.microsoft.com/fwlink/?LinkId=178632 .
    This section of the Microsoft Unified Communications Open Interoperability Page lists the SIP Trunking Services that have been independently certified to meet the requirements of the Unified Communications Open Interoperability Program.

  • Vendor Process for Microsoft Unified Communications Open Interoperability Program at http://go.microsoft.com/fwlink/?LinkId=178633 .
    This page gives the technical and marketing requirements that vendors must meet to qualify their SIP trunking offering for use with Microsoft Office Communications Server 2007 R2.

Internet Engineering Task Force Resources

SIP Forum Resources

Office Communications Server Resources


  • Visit the Office Communications Server main pag

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