What Is A SIP Trunk?
SIP = Session Initiated Protocol, the standard developed to set up and tear down VoIP calls and create dial tone, the ability for a VoIP phone to ring and so forth.
Basically a SIP trunk is a virtual phone line. SIP connects your IP ready PBX to the PSTN (Public Switched Telephone Network) where non-VoIP business and people are still making and receiving phone calls.
To connect via SIP we’ll be using an Internet Telephone Service Provider (ITSP) because somehow we have to get the VoIP from our phone system to the PSTN and the other people we’re calling (local termination)
It is possible to do SIP to SIP, where we bypass the PSTN entirely, but this doesn’t occur much at this point in business applications.
So a SIP trunk is a virtual phone line delivered over a dedicated internet pipe.
SIP Trunk Overview
One of the basic advantages of SIP trunking is we can CONVERGE our voice and internet data traffic, and fire the local phone company with all their expensive phone lines. Of course, this is the basic cost savings premise of all VoIP deployments.
However, this becomes especially important if we have multiple locations, lots of lines or PRI, T1s or analog lines. Another advantage is we get to enjoy dynamic bandwidth allocation. This means if we’re not on the phone, our internet bandwidth is freed up to use for other purposes like e-mail and so forth.
Sometimes we can have the local service provider we pick implement VoIP for us, so we can use our old PBX and phone equipment.
So SIP gives the multi-branch businesses tremendous money savings because we can not only start to migrate all voice services from existing lines at each location but also manage all this on a single IP network. No more “phone network and wires” and then a separate “computer network and wires”. It’s all being CONVERGED on one set of wires, and one network – the computer network if you will.
Additionally, we will pay less for a typical SIP trunk and can buy one at a time, compared with larger business needs where older type PRIs, required us to buy 23 or 24 lines minimum on a PRI T1.
Lastly, we get Direct Inward Dial (DID) numbers like PRI, along with unlimited INBOUND calling on most SIP offerings. We should also enjoy lower long Distance rates in most plans, sometimes even on an unlimited basis.
To recap, a SIP trunk is a virtual phone line, used for premise-based phone systems, not hosted PBX, must use either an IP ready PBX or have an integrated access device (IAD) installed by the service provider if you want to keep your old non-IP phone system.
Besides generally being less expensive per line, SIP trunking allows us to converge voice and data on the same pipe and does so dynamically, so we’re not paying for a bunch of lines that we use only occasionally (like when we had to plan for busy, peak calling periods)
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What’s Needed To
Use SIP TrunkING?
We’re going to need four basic things :
– A SIP provider / ITSP to sell us the trunks we need and then terminate our calls on the PSTN.
– A good internet connection that is designed for VoIP (sometimes this might come with the trunks – a good thing)
– IP PBX ready to do VoIP and accept our trunks, or a service provider implementation with an IAD so we can use our old PBX
– Some sort of “edge” device to put on our internet connection to assure the SIP is coming and going well, we don’t sound like Darth Vader or a Christmas Elf. This will at the same time protect us from the nasties on the internet (SIP border controller/gateway/SIP-enabled firewall)
So for those of you that would like more detail.
If you’re contemplating buying an IP PBX, then using SIP will avoid having to use a PSTN gateway like the IAD described above, because you’ll be using the ITSP to perform this function at their end.
There’s also a whole bunch of cool stuff we can do with SIP that will never happen with an IAD or PSTN gateway, like Unified Communications, video conferencing and Instant Messaging, etc.
It’s important to have Service Level Agreements (SLAs) on your Internet and SIP services, to assure your voice calls are clear, as well as of high quality. Big bandwidth doesn’t always equate to quality VoIP calls.
Lastly, although it’s great to converge services and so forth, remember that in the “old” TDM / PRI world, we had a dedicated 82/k channel for each voice call. Now with VoIP, we may well need more than this bandwidth on our internet pipe for each concurrent call if we’re using a CODEC such as G.711, that could be carving out over 120/k per call, depending on the implementation.
Add in UC, video and file transfers, and all of a sudden our employees are doing a great job of communicating and making it happen, but we’re chewing up bandwidth like no tomorrow!
And lastly (really this is the last “lastly”) <;-) make sure you have a good edge device and firewall. Not all firewalls deal with VoIP and NAT traversal issues well. You simply must have VoIP security, make sure the SIP trunks themselves (the provider) are using encryption since your phone calls will probably be going out over the public internet.
SIP Trunk Advantages
– Simplified converged IP network, especially in multi-site deployments where we have lots of PRIs, phone lines and internet needs. Let’s put it all on one set of wires, or network and manage it there.
– SIP trunk supports voice, video and Unified Communications (UC) since it’s basically computer language, all this stuff can use SIP and then used to keep all communications converged, connected and managed.
– Save money, SIP trunks will almost always be less than “old phone lines” and more so in multi-site deployments, allowing us to leverage over subscription schemes across the entire enterprise, instead of just by location as we do now with PRI, T1 or analog lines.
– Converged service – lets us use our nice, high-quality internet pipe or MPLS network for all services, easier management – one network and cheaper than maintaining separate facilities (wires/network access) for each voice and then data too.
– Single SIP trunk may replace many TDM trunks (at multiple locations) based on oversubscription policies as above.
– Dynamic bandwidth, allows us to plan for peak loads without having to buy a bunch of PRIs and have them sitting around waiting for busy times of the day at each location.
SIP Trunk Disadvantages
-Well, you need to buy an IP PBX to use SIP trunking. As I’ve said before, you can many times still do a service provider implementation of VoIP, using an IAD and keep your old phone system. Maybe a hosted system is better for you?
-Can I get DID in my rate center? This refers to making sure the SIP provider / ITSP actually has phone numbers in your rate center. Most do, but I have run into some areas where the local phone provider for whatever reason is not releasing DID/phone numbers for ITSPs to use for VoIP.
The more remote your business, the more chance there may be a DID availability issue. Make sure people calling you locally won’t have to pay long-distance charges.
-Single locations may not benefit from SIP Trunking. Customers are always surprised when I share with them “if it ain’t broke – don’t fix it” Small single locations may better be served by a couple of POTS lines with unlimited long distance or maybe a small hosted PBX offering.
-High bandwidth needs may require separate facilities for VoIP and data. Remember how I always say one of the main reasons for switching to VoIP is to be able to converge all voice and data on one wire/internet pipe?
I have a customer that has huge data transport needs. Since the files they are sending are so BIG and sent intermittently, the VoIP edge devices were having a hard time making the VoIP phone calls sound good. So we put in a simple MPLS circuit for the VoIP and SIP trunks, then left their big pipe for the file transfers.
-Security issues – because we’re putting our phone calls on the Internet with a SIP trunk (with old “phone lines” at least we had complete privacy unless the “phone guy” was listening in <;-)) we need to have bulletproof security. In your quest for lower-cost phone service, make sure your IP PBX and SIP trunk provider has this security.
-Traffic shaping may be required if we have either low-cost internet access (poor quality to begin with) or a very large number of concurrent calls occurring with large data transfers over our internet pipe (think FTP and e-mail servers on-site, other large files transfers)
In the end, if traffic shaping doesn’t cut it, we may have to break down and get decent dedicated internet (costs more than a cable modem folks…) or provide separate VoIP and data internet facilities as discussed above.
-Quality of service provider, umm, you might have read elsewhere here that you get what you pay for? No difference here. Remember, when we leave good old “Ma Bell” and go out on our own doing cool VoIP phone service type stuff, we need to assure we’re dealing with reputable companies.
One suggestion here, do the “one throat to choke” SIP trunk model. I.e.: get your SIP trunks and internet access from the same provider, then when something happens, less finger-pointing, it’s either your phone guy handling the IP PBX or it’s the SIP / Internet provider (ITSP)