SIP Trunking Basics
Getting Acquainted with SIP Trunking
Most medium and large enterprises today use a PBX telephone system to provide voice and other communications services to their employees. The telecommunication network connections that connect the PBX to a telephone service provider are known as trunks. Trunks connect the individual lines (desk phones) to a bigger line, which connects back to the network. Trunking allows an enterprise to pay for fewer phone “lines” than it has users with phones.
Today, modern systems use SIP trunks, which use VoIP and SIP. SIP trunking uses a data connection to carry voice signals as VoIP to a service provider who can handle that kind of voice signal. A carrier who can offer VoIP services is called an Internet Telephony Service Provider (ITSP).
SIP trunking relies on Internet protocols and Internet services instead of old-fashioned circuit-switched voice protocols and services. With SIP trunking, voice communications can be merged with the data services an enterprise uses. SIP is the key protocol that supports VoIP connections. SIP relied on clients and servers in the network to exchange information about who is connected to the network, where they are located, what resources are available, and who’s inviting people to begin a session.
A SIP trunk is a virtual connection between an Internet Protocol Private Branch Exchange (IP-PBX) and a telephone service provider providing SIP-based voice and UC services, connected over and enterprise’s data network connections.
Components of SIP Trunking
Some of the components needed to implement SIP trunking are hardware devices that you need installed on your network, while others are the services themselves, provided by a third party service provider.
The key element to SIP trunking is a phone system that can convert voice calls to VoIP calls for transmission across the SIP trunk. The most common and cost-effective mechanism for this is an Internet Protocol Private Branch Exchange, or IP PBX.
Session Border Controller (SBC)
The other key element of SIP trunking solutions is the Session Border Controller or SBC. The SBC is the device that sits on the border between an enterprise’s private network and the public network provided by the data and telephony service providers. Read more about SBCs here.
Session Management System
An optional, but useful addition to a SIP trunking implementation is a Session Management System. Without session management, many UC applications will have their own servers, their own management systems, their own policy enforcement, and their own policy database systems – each individually controlled and managed. Session management brings all these elements under a single system’s control, saving time and money. Session management provides intermediation and federation between different platforms, different networks, and different geographies.
ITSP/SIP Service Provider
Your SIP sessions need a public network provider to get to where they’re going if they’re not internal calls handled on your own enterprise network. That’s the role of a service provider. A service provider that provides transport and termination of SIP calls is the ITSP, or SIP service provider. They are like a traditional phone company, except their interface with your network is a data connection using SIP to control the flow and routing of sessions.
Benefits of SIP Trunking
SIP Trunking provides some concrete and measurable benefits to enterprises, by simplifying network elements, enabling new service, and reducing expenses.
Unify Your Access Network
SIP trunking lets you combine your voice and data network into a single, unified data connection. Today, most enterprises have separate network facilities to connect their Private Branch Exchange (PBX) to the network of their phone company for transporting inbound and outbound calls that are outside the company network. Most enterprises have Internet Protocol Private Branch Exchanges (IP-PBXs) that use SIP and VoIP for the portion of calls between the PBX and the actual phone, as well as using SIP and VoIP for intra-company calls. But, for calls that are coming from or going to third parties, these SIP sessions are using devices called Public Switched Telephone Network (PSTN) Gateways, which use old-fashioned circuit switched phone lines.
SIP trunking lets you get rid of the expensive PRI and Basic Rate (BRI) connections and the PSTN Gateways used for these calls. Moving to SIP trunking keeps your calls as SIP session instead of converting them to TDM, and delivers them to SIP service providers. With SIP trunking you can even use the data connection you already have in place for Internet and related services for your voice calls.
Getting rid of BRIs, PRIs, and PSTN gateways means less expense, fewer things to manage, and less “stuff” for an overworked IT staff to deal with.
Make Management Easier
When you eliminate PRIs and all their related equipment and standardize on data networks for voice and Unified Communications (UC), as you do with SIP trunking, you make the network easier to administer and manage. You have less equipment and network access connections to manage.
SIP trunking, with the addition of SBCs, allows you to centralize the management of your entire network – all your locations and IP PBXs. When you centralize control you can:
- Create centralized dialing plans that apply to every location in your enterprise
- Use a single carrier rather than contracting with multiple local and long distance carriers in each location
- Set your security and application policies one time in a centralized policy management database
- Centralize your billing and cost accounting, so your know exactly what you’re spending and where
SIP trunking makes it easier to add capacity to your VoIP and UC network when needed. In the old BRI/PRI days, if you needed more lines, you called the phone company and had them installed and integrated. With SIP trunking, because your voice and UC sessions are carried over your integrated IP data connection, you can simply allocate more or less bandwidth as needed. It is both easier and cheaper.
Deploy SIP Services Easily
Adopting SIP trunking and SBCs facilitates the adoption of new SIP based UC services and applications. SBCs have the ability to make issues of location, vendor interoperability, and legacy equipment obsolescence go away. The SBC is designed to intermediate between different and disparate network locations and equipment choices – translating protocols, codecs, and more when needed. The result is a network that no longer looks like ten different and disconnected networks, but rather one big network. Applications can be deployed one time, in one place, and made accessible to everyone, even remote users. Services that can be enabled through SIP trunking include:
- Audio and video conferencing
- Audio recording
- Instant messaging (IM)
- Online collaboration (white boarding)
Enable Cloud Based Services
SIP trunking makes it easier to connect to services that are hosted in the cloud. Because your voice and UC communications are entirely IP and SIP based, there’s no longer a need to have applications delivered from a server connected directly to your IP PBX. Instead, a server could be located somewhere in the data center of a cloud service provider. Cloud services typically cost less and are quicker to deploy.
Make Your Network More Reliable and Resilient
Because it isn’t tied to a fixed set of PRI or BRI connections to a traditional phone service provider, a SIP trunking deployment is more resilient and reliable than a traditional PBX trunking solution. A SIP trunking solution puts much of the heavy lifting of routing and completing calls and sessions into hardened, redundant ITSP data centers off your site, instead of distributing it to a number of local phone exchanges and on local circuit-switched BRI/PRI connections.
The result is that when something bad happens (power outages, natural disasters, etc) it’s easy to reroute sessions to other locations And SIP trunking also makes it easy for those businesses who require this level of redundancy to work with more than one SIP trunking ITSP so in the unlikely case that an entire ITSP goes down, it’s easy to automatically reroute traffice through the other ITSP.
Going Local and Eliminating 800
Because SIP trunking is based on the Internet, it essentially takes location out of the equation. With the help of your SIP trunking ITSP, you can establish Points of Presence (POPs) in local areas where your do business without having to hacve a physical office presence there. Your business “looks” more local for your customers when you have a local number.
SIP Trunking Considerations
A big part of any SIP trunking deployment is the company that actually provides the services – the Internet Telephony Service Provider – ITSP. Many companies offer SIP trunking services. You may find SIP trunking offered by many types of vendors, including:
- Your traditional Time Division Multiplexing (TDM) telephone service provider
- The data service provider who offers your Multiprotocol Label Switching (MPSL) Wide Area Networking (WAN) services
- A “pure play” SIP trunking provider focused solely on SIP trunking
Most enterprises choose SIP trunking because of its cost savings benefits. To work out your pricing, consider the following:
- On-net traffic – How does the ITSP handle your on-net traffic – the traffic that goes from one of your locations to another?
- Pricing Structure – For the off-net traffic, you do have to pay for, how do you pay for it?
- International calls – These typically are charged at different rates on a per-country basis, so look closely at your usage to see which countries you call most frequently
The price itself – After you look at how the pricing structure matches up with your usage, you can look at the actual prices charged and create your business case to estimate your SIP trunking savings.
Dealing with DID and local calls
Direct Inward Dialing (DID) is simply the series of numbers that customers and partners use to dial into your network and Private Branch Exchange (PBX). Keep in mind two things regarding DID when evaluating an ITSP:
Number portability – Can you keep your existing numbers?
Local number availability – Beyond keeping your existing numbers, you may wish to establish local numbers in other locations, in lieu of toll-free 800 number services.
Dispersing aggregation points
Aggregation points are the spots where multiple SIP trunks from your offices come together and into the ISP service ITSP’s network. They’re a potential weak spot in a SIP trunking deployment not because there’s anything inherently unreliable about them, but simply because they can be a single point of failure when the unexpected happens. You can deal with this potential in two ways:
Ensure that the aggregation points are properly hardened. Your ITSP should be able to tell you about its redundancy and its disaster recovery plans and be able to give you a good story about how it would deal with failures.
If your network is big enough or mission critical enough, have multiple, geographically separated SIP trunking aggregation points and configure your network to gracefully failover when the worst case scenario plays itself out. This could be through contracting with two different ITSP or through one ITSP who can offer you this option.
Quality of Service
Your ITSP should be able to assure you that the quality of service you get from SIP trunking services meets or exceeds the quality of service from your existing TDM trunking, a few things to consider:
Voice and UC networks are built around peak traffic considerations – a reasonable calculation of how many simultaneous sessions need to be supported at your busiest hour of the day. They aren’t built around an assumption that everyone in your organization suddenly tries to use every IS application that they can all at the same time. It’s possible that in some extreme cases you’ll have more sessions than your network can handle – this is called blocking. Your ITSP should specify in its contract with you what an acceptable amount of blocking will be and be contractually obligated to maintain this.
Codecs are the software algorithms that digitize and compress voice and video signals for transmission across an IP network. There are multiple codecs. The choice of codec impacts two things: The voice quality of the call, and the amount of bandwidth used. Ask your ITSP which codecs it supports and uses.
Look closely at the Service Level Agreement (SLA) for your SIP trunking services. Make sure there are penalties for SLA metrics that aren’t met. Your SLA will cover things like uptime, latency, and jitter.