We are about to replace Windstream PRI connection to SIP for saving $$ and flexibility. I have a Access Point Inc quote that show $272 a month instead of $950 a month we are currently paying. Have anyone of you experience with them or recommend other SIP provider out there? We are using Shoretel system. Thanks
Never heard of them, but the pricing seems about right for a single-PRI replacement. I believe NexVortex is probably the most popular trunk provider in the group aside from whatever ISP you have (talk to @CGreenTX).
NexVortex is definitely our preferred vendor, and we can help you with that if you would like. There are others lingering here that are also vendors if you have a preferred.
That said, the key thing with SIP providers is to understand the different ways they are sold.
A large portion of providers sell lines in the traditional phone line model where they have replaced the concept of a line with a channel. This could be unlimited use or metered, however you get exactly the number of channels you pay for. Honestly, this model is difficult to save money with unless you are absolutely certain that you don’t use your phones much at all.
Carriers like NexVortex don’t charge you for channels at all, charging instead purely based on usage. This model saves most people a lot of money over a traditional PRI, particularly churches who don’t use their phones to the level of a call center. Plans are based on a quantity of inbound/outbound minutes and as the size goes up you get a larger bundle of DID numbers included.
The less common option is to go completely usage based. Most churches like consistent bills so this isn’t popular, but there are many carriers out there that cater to the idea of paying per minute for actual usage. If you’re willing to go fully a la carte and per minute you may be able to save a significant amount of money, but you’re not going to get the same kind of support from these types of carriers.
Thanks for all. we just talk to Nextiva and it seems that they are offer the same thing. Charge by usage or metered. It seems that bit confusing at first but I got the idea. Other concern is how reliable is this? we are using Veriozn Fios 1G connection as our Internet line. Some vendor offer dedicated line per campus. I may to call NexVortex as well for information Thank you.
Great question… Any call crossing the open Internet is at the mercy of the prevailing winds that affect all traffic. Any packet loss, jitter, latency, etc. has the opportunity to cause problems with calls. Your chances of this happening increase as you go down the stack from dedicated fiber Internet all the way to consumer class Internet. Verizon FIOS, while fiber based and well above average, is still in the class of service comparable to most cable providers. We have customers running their VOIP over Comcast Coax connections and they’ve never had an issue. We have customers running VOIP over FIOS without issue. Just be aware that there is nothing guaranteeing quality so most issues with packet loss/jitter/etc. will need to be accepted as the cost of saving money.
Another factor is going to be your edge gateway and how it impacts SIP traffic. You will need to work with your PBX provider to properly configure your phone system and firewall to pass the traffic through without breaking the SIP headers or RTP streams. Some SIP providers will send you their own gateway device to put outside your firewall to completely avoid this issue.
In my experience, it’s pretty common for somewhere in your chain there to exist an adaptive jitter buffer, which smooths over a lot of the latency/jitter issues at the expense of causing delay - but those can’t be present in any modem-based system (like FAX). A consumer-class ISP won’t provide the consistency of service necessary for any SIP-based FAXing to work, so go ahead and write that off. From there, connection quality depends a lot on if they provider does direct media or not and how that affects the routing of the call traffic. For example, nexVortex proxies media and has direct connectivity with many different ISPs so they can ensure things like peering disputes don’t affect your call quality.
Another common issue with using consumer Internet for VoIP is they commonly have a SIP-ALG on your gateway that will require special steps to defeat. Honestly, my experience has been that you can switch from PRI phones to VoIP + Enterprise Fiber and stay pretty close to the same money if call quality is a primary concern.
We use Bandwidth.com for SIPs, and have had great success with them. They also offer a charity rate to MBS referrals ($15/trunk/month metered; $21.50 for unmetered). Let me know if we can help!
Regardless of the supplier you use or the cost, you need to ensure they will guarantee following technical specifications:
1/ low latency & SIP/VoIP traffic prioritisation
Voice is very sensitive to congestion and momentary delays (latency) as it is a real-time protocol. This means that SIP & VoIP traffic must be prioritised in your own LAN network (e.g. physically separate LAN & dedicated WiFi), high-prioritisation on your external router and your ISP/SIP provider must provide latency & VoIP prioritisation over their own network,
You can get away without these guarantees for perhaps a couple of hosted handsets, but as you increase the number simultaneous calls, the potential for congestion and latency degradation increases, which affect voice quality and can drop calls.
If you have an internal SIP platform on your LAN, you can control the problems much better and will have a much lower SIP/VoIP demand on your WAN connection. If you have hosted VoIP, then you will have a very much higher demand on your WAN link as all you internal calls will be ‘trombones’ down your WAN connection.
2/ minimum capacity of 120kbit/s per concurrent call
Whilst your PRI connection has dedicated capacity, it can be idle for much of the time as voice is always operated on a contended basis. Thus in traditional fixed circuit telecoms each dedicated channel has a fixed capacity of 64kbit/s per simultaneous call that is reserved even when no calls are present, so channel congestion issue.
However, when you move to VoIP (SIP is actually the signalling protocol for VoIP), then you have to take that 64kbit/s voice information and encapsulate it in an ethernet packed along with a lot of other information (SIP details for example) as well as the encoding information. Thus at least 50% of the packet content in VoIP is overhead data that is needed to correctly encode, decode, identify, route, prioritise and integrity check the voice data.
If you are experiencing congestion problems with VoIP, you can change some of the encoding options on the handset and the VoIP switch to reduce the size of the voice bit, but you lose quality in doing so, but you can get away with about 90kbit/s before you start to notice significant voice quality issues.
Thus when planning your SIP/VoIP implementation or considering changes, use 120kbit/s per VoIP terminal or per network link (WAN or LAN) and you will not go far wrong.
Bear in mind that if have users that are using FaceTime Audio, SkypeCall, WhatsApp Audio, FaceBook Audio and similar, these are all SIP/VoIP variants and will have SIP headers and any general traffic prioritisation plan should take account of these. Similarly ANY form of video chat using the above or similar caller to caller video connection (and some video streaming system) will also use SIP protocols and have the same latency, congestion, voice &/or video quality sensitivities. Video connections will consume from 5x to 15x the bandwidth per connection. Nothing kills a WAN connection quicker than lots of people trying to FaceTime each other!
Thank you so much! this is really good information.
I would like to follow up on this.
This may help others. I’ve been gathering the information from various provider such as NexVortex, Nextiva, Acces Point, Cox Communication and Verizon. NexVortex was close to my choice but I selected Cox Communication, thanks to Russ. Cox will provide separated (dedicated) line from Internet connection. They are bit more expensive than NexVortex but we still saves $$ from current PRI provider.
Now our Shoretel Partner , IPC Technologies, give us extra cost per “requirement” to moving PRI to SIP. In the quote, ST1D switch, Ingate SIParatorw/25 transersal license and SIP Trunk Software Lincenses, and Partner support with installation and Configuration. It is little less than 8K! I digging around a bit about Shoretel SIP and found out that we may not need Ingate SIParator becuase Cox is “Native” SIP provider. IPC haven’t give me the answer yet about Ingate, but I am start to think whether I need to trust this company. I am still investiating whether we need to replace our SGT1 switch to ST1D. We also have SG120 and 2 x SG90 voiceswitch. IPC told us that SG120 and SG90 are not compatible for SIP Trunk. I need to find out. Please let me know if you know anything about this.
Thank you all!
ShoreTel should be able to talk SIP with just about any provider natively. Your provider is bringing that to the party so they can handle the hand-off to the PBX and the NAT Traversal in a way they can monitor/manage.
If you’re curious about some alternatives, drop me a message. I’d prefer you not have to drop this kind of money, or replace equipment unnecessarily, in the name of saving a few hundred bucks on telco services.
Office365 Cloud PBX might be a good option. The PSTN lines provided are pretty reliable and clear.
Update: Shoretel works with Cox however, there are some report that there is some issue such as conferencing and it is not smooth as much as Ingate. Anyone out there who uses Cox SIP service on Shoretel Phone system?
IPC Tech, Shoretel partner telling me that Cox is not supported; however the Shoretel site showing the Cox is supported. Any input will be greatly appreciated. Thanks
One of the things to check on any IP system hardware is the reset time. E.g. if you have an internal PBX (e.g. an old analogue or digital hybrid), if you reset the system, it is usually back up and operating in less than 20 seconds. This means you have minimal interruption to telephone users who expect the phones to ALWAYS work ALL of the time.
I have found that many IP telephony systems (including some Cloud-based platforms) can have reset times of the order of several minutes. Check on the availability of any Cloud based service that their uptime is better than 99.9999%, including any router/dsl/sip connection to your premises. Check how sensitive your organisation may be to losing its phone system connectivity before accepting lower system availability to what you may already have.