SIP Trunking Checklist | Flowroute Blog

SIP Trunking Checklist: Everything you need to get started

Posted on November 26, 2014 by Antha Mack

The global market for SIP trunking grew 50% in 2013. That’s like the economy of a small country growing by half between Christmases. What that proves to me, is that SIP trunking is an important business communications tool you can’t ignore. If all this hype is inspiring you to get in on the action, there are some details you need to know before getting started. So here is the beginner’s guide to everything you need to use SIP trunking for your primary business phone connection.

SIP-trunking-checklist

Hardware needs

IP-PBX

While you can use an adapter to connect your traditional PBX via SIP trunking, IP-PBXes are more user friendly than traditional boxes, and give you the power to easily connect a branch office or remote workers via the Internet.

Based on the experience of our customers, the IP-PBXes we see the most success with are from FreeSWITCH, CudaTel, Asterisk, and 3CX. They all present different pros and cons depending on your structure and needs, but each one could be the subject of whole other post. What is important for now is that your IP-PBX gives you the features and functionality necessary to make the most of your investment.

Must have IP-PBX features:

  • Dial Plans and Call Routing: You need to be able to customize where and how calls are routed.
  • Hold Music: Your IP-PBX keeps calls going when they’re on hold and is responsible for what callers hear while on hold. If you’re picky about hold music, look into the options that come standard and what your customization options are.
  • Voicemail: Most times, your IP-PBX also hosts your voicemail. So think about capacity in terms of how many inboxes you need and how much storage you expect to fill.
Phones

When it comes to IP telephony, phones don’t necessarily sit on your desk. Softphone applications can be installed on your desktop, laptop, and even mobile device. It’s important to remember that your endpoints need to be SIP compatible, and ideally SIP RFC 3261 compliant. Our customers have good experiences with phones from Snom, Grandstream, Yealink, and Obi-hai. For a reliable softphone check out Bria or 3CX.

Firewall

Web thieves are everywhere. You need a firewall to protect your connections. A firewall can come in many forms, including a router or modem, and is essentially a bouncer with a guest list detailing which types of data are allowed in and from where. Your options are broad, and I strongly suggest you do your research. Firewalls can be a major sore spot for SIP communications. The first thing you should do when putting your new bouncer in place is to disable SIP ALG if it’s there. More often than not, having customers disable the SIP ALG on their firewall solves any voice quality issues they’re having.

When configuring your firewall, tell it to only allow SIP packets from your trusted SIP trunking providers, and accept RTP from any IP address within your system’s media port range to make sure you open enough ports to handle your maximum level of concurrent calls, otherwise you’ll end up with bounced calls.

Connection needs

Without sufficient bandwidth, you’ll back up traffic behind a bottleneck. When that happens, packets are delayed and calls are hit with jitter and latency – people on the other end will say things like, “You’re cutting out.” and “Why didn’t you get enough bandwidth to handle your calls?”

Conservatively, you need 80kbps per SIP based call. Multiply that by your expected concurrent calls and add it to your current peak Internet consumption to calculate your needs. While you’re at it, make sure your wireless routers are able to handle the additional load from laptop and mobile based softphones.

Some will tell you that you need to have a dedicated IP connection for your voice. In reality, we don’t see a lot of businesses go to that extreme. But it is a very good idea to prioritize voice traffic on your network and with your ISP by making sure SIP and RTP packets get through first.

Security

The security of your voice connection depends largely on the security of your network. It’s a good idea to review these tips for securing your network to keep your SIP trunking connection out of the hands of remorseless Internet pirates.

It’s also a very good idea to initiate the security/toll-fraud controls provided by your SIP Trunking carrier. One way you can protect your organization from fraud is to only allow outbound calls from whitelisted static IP addresses. If your IP addresses are dynamic, you can use services to create static hosts that route to dynamic IPs.

Functionality needs

Phone numbers

There are processes you’ll need to follow to get your existing numbers routing with your new SIP trunking carrier, and they can take time – losing service providers sometimes require up to six weeks to release phone numbers. To port your numbers you’ll need a fresh (no more than 30 days old) copy of your phone bill that lists the phone numbers you’re porting, the service address, and the billing phone number – it’s information your old carrier will require for identification and verification.

If you want new numbers, remember, it’s really hard to get customized numbers that aren’t toll-free. It’s almost luck of the draw. If there are specific rate centers you want numbers in, make sure your new SIP carrier can supply numbers in that region.

Toll-free numbers

The global supply of 800, 888, 877, and 866 toll-free numbers is exhausted. That doesn’t mean they’re all in use, it just means that someone owns them and might be willing to part with them for a price if there’s one you simply must have. Outside of those, there are 855’s, and 844’s available.

Additional Services

Because SIP trunking isn’t connected to a physical address the way your traditional phone lines are, 911 service is an add-on feature you’ll need to configure with the proper address.

If for some reason, your power is cut or your Internet connection is lost, you’ll need another way to receive calls that activate immediately. Failover routing is a feature that sends calls to an alternate destination (mobile phone, or another unaffected location) in the event your main line is unavailable.

The Payoff

SIP trunking helps your business be more productive and can dramatically cut costs. You’ll be able to connect Unified Communications and integrate with WebRTC applications, plus better handle BYOD bill reconciliation. And now that you know what you need, you’re ready to start shopping. To help you get started, here’s a collection of advice from buyers who have been through the process, and a list of expert tips for choosing the right SIP trunking provider.

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