Posts from ‘small office phone systems’
VoIP Network Monitor ‘SIPQOS’ Launches Beta
Many of us have struggled with VoIP Network Monitoring, keeping tabs on our network without having to manually review the health is always a hassle and concern. For every network my team erected we needed to erect a proper monitor. For smaller networks and even VoIP phone systems the traditional Network Monitors were far to expensive to implement and required port mirroring which meant servers had to be deployed in the VoIP network that required monitoring.
So, we created SIPQOS… SIPQOS is a service that allows VoIP network administrators to attach virtual SIP endpoints to their network which send calls to-and-fro and monitors those calls for interruption. It’s a simplistic approach to a complex problem, if the network drops a registration or if a call fails it’s likely (from personal experience at least) that the issue applies to the entire network and other endpoints are experiencing the same problem. SIPQOS won’t take the place of more expensive in-network solutions but it does a great job of providing redundant VoIP network monitoring and SIP-based VoIP phone system monitoring as well.
An excerpt from the announcement we made on the 10th…
VANCOUVER, February. 10 – SIPQOS (pronounced SIP-KWOSS), a new entrant in the VoIP network monitoring market has launched a beta of its remote VoIP network monitoring service today. SIPQOS released the first product to bring the power of remote VoIP network monitoring by combining embedded SIP (Session Initiation Protocol) User Agents, web services and some secret sauce. SIPQOS monitors VoIP networks remotely and alerts network administrators when a problem has been detected.
SIPQOS is doing a great job for us and provides redundant VoIP network monitoring on a production network we run today. It also fills a void where others solutions fell flat, SMS alerts are critical and SIPQOS delivers in spades on that front. Those interested should give it a whirl, it’s free to sign up and the plans after the 30 day trial are cheap by anyone’s standards.
Written by Erik Lagerway - Visit Website
SIP trunks are simply another way of saying VoIP Provider for your phone system. A SIP trunk is a connection from a PBX (phone system) using SIP (Session Initiation Protocol) to an ITSP (Internet Telephony Service Provider).
It might sound complicated but it’s really quite simple, SIP trunks take the place of your legacy telephone company. Most phone systems out there today are more than a couple of years old and are likely based on circuit switched technology. Newer IP-PBXs use packet switching technology, which means they leverage the Internet to deliver the same features you have now, and then some. The difference could be minor or major depending on what your PBX is capable of and what your ITSP can deliver in terms of features and functionality.
Since the PSTN (public switch telephone network) is tied to aging circuit switched technology it has limitations in terms of what media it can support. Essentially, it can deliver low quality voice, that’s it.
SIP Trunks replace older PRI and POTS interfaces that we are used to and bring to the table a wide variety of communications options. Depending on your IP-PBX and your ITSP you could potentially look forward to HD (High Defenition) Voice and potentially HD Video.
HD voice (and video) for small business in Canada will happen, it’s only a matter of time. As broadband providers increase upstream bandwidth and dual WAN link-failover devices become common place, SIP trunking will accellerate in growth and on-net (calls made on the ITSP network) HD Voice will become common place.
Unfortunately, HD communication off-net (eg. PSTN) is not going anywhere at any great speed. Jeff Pulver is back as he reboots the communications industry with his new HD Communication Summit. I welcome Jeff back with open arms, if anyone can convince operators to increase speed towards wide-band/HD adoption it would Jeff Pulver.
Today we can see SIP trunking providers and hosted pbx providers supporting wideband codecs and devices on their networks. This will allow user to communicate in high definition with other users that have devices that support it, in brief you could have better calls between you and your colleagues in the office and remote office workers connected to the same PBX, and that is a step in the right direction.
Written by Erik Lagerway - Visit Website
G.711 is the default audio CODEC for most Response Point phones and requires approximately 90Kbps bandwidth upstream (your voice going out) and 90Kbps bandwidth downstream (your caller’s voice coming in).
To calculate peak usage take the peak concurrent callers x 90Kbps. For example: 5 concurrent calls x 90Kbps = 450Kbps is the required bandwidth for each direction. Keep in mind, this does not account for VPN usage for remote users or voice mail to email etc.
As an example, if you have a 1Mbps ADSL connection from your service provider, you might have an upstream bandwidth of approximately 700 Kbps. A conservative approach is to estimate just over half of the upstream bandwidth is available, ISPs generally over-sell their bandwidth. In this case, you could safely support 4 simultaneous G.711 calls if you were not doing anything else (e.g. downloading email, listening to online radio, downloading large files, etc.) on that connection.
The SMB Digital Voice network also supports G.729, which uses approximately 20Kbps bandwidth upstream (your voice going out) and 20Kbps bandwidth downstream (your caller’s voice coming in) for each call. G.729 provides very good call quality while minimizing bandwidth usage. The only noticeable difference would likely arise during on-net calls (calling other users on the SMB Phone network). G.711 offers a higher quality on-net call because G.711 does not compress audio, but as soon as the the call is handed off to the PSTN the call quality between G.711 and G.729 is hardly noticeable.
G.729 offers some real benefits, the most obvious is the 400% decrease in bandwidth capacity requirements. G.729 also handles Jitter more efficiency during times where low bandwidth / high congestion would likely render a similar call using G.711 unintelligible.
You can force your phone to use G.729 on Response Point handsets but some are harder to configure than others. For example, on Aastra 675x phones the global SIP settings are grayed out out via Javascript on page load making it tough to set the codec.
As a general rule of thumb, we like to recommend an independent broadband connection that you can use for Response Point. You may want to acquire a router that has dual WAN link failover, VPN Server (for remote sites) and some QOS traffic shaping functionality.
Written by Erik Lagerway - Visit WebsiteI wrote an article over at the SMB Phone blog on Response Point VPNs and remote workers. If you are having some issues with VPNs and Response Point this might help.
Written by Erik Lagerway - Visit Website
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Back in 2004 I wrote a post relating to the VON Canada Panel I sat on with Niklas Zennstrom. It was an interesting debate on open standards (SIP in this case) and closed networks, specifically Skype. I was quite vocal about how silly I thought Skype was not to include SIP, a few of you picked up on that 
It looks like something good came of the eBay purchase as we now see a Skype pushing towards open standards, good stuff!
On a similar note, I heard a rumour that it’s likely Jason Fischl the current CTO at Counterpath (Xten) will be going over to work with Jonathan Christensen (General Manager – Media Platform) at Skype. Jason was an early advocate of SIP in the IETF and works with some of the best minds on the subject: Cullen Jennings, Robert Sparks, Alan Duric come to mind.
This could get interesting.
I will do some testing with SkypeforSIP & Response Point and post the results along with my comments on what this new offer from Skype might mean for Response Point.
Written by Erik Lagerway - Visit Website
VS 
I will admit, this is a bit of a silly comparison but the truth is that I have had a few customers (and some analysts) asking for some clarification on the new Google Voice offer and how it may compete with Response Point when coupled with an ITSP. The fact is they really do not compete in any measurable way and they could easily compliment each other.
Major Differences
The obvious major difference is that Response Point is a small business phone system, Google Voice is really a service offering targeted at individuals.
When we combine Response Point with an ITSP (Internet Telephony Service Provider) we start seeing some similarities in the services between the two offers but they are really meant for 2 distinctly different purposes.
Response Point offers an actual premise-based system with a base unit, handsets and features like; auto-receptionist, DID integration, hunt groups, voice mail to email integration etc. All of the things one would expect when purchasing a small business phone system.
Google Voice service is an overlay service on whatever you have today, so if your existing phone system is simply not cutting it, it’s unlikely that Google Voice is going to be able to transform it into the system of your dreams. It’s true that Google Voice will allow you to take advantage of certain features but don’t expect to find a Park, Hold or Transfer or anything fancy like speech recognition.
Google Voice is an inbound-centric service. Most features can only be used with an inbound call, that includes call recording and call joining.
How they play nice together
One could use the Google Voice – simulring feature to call your Response Point phone number and at the same time it could call your mobile.
Google Voice – call recording is a handy feature that is currently not a feature offered in the Response Point system.
Google Voice – voice mail transcriptions is a handy way to receive visual voice mails via email and SMS.
Google Voice – call widgets allow users to put callback widgets on a website. This will allow the visitor to put in their phone number and the system will call them and then it will call your Google Voice number.
Google Voice – SMS is a cool way to compose, accept and manage text messages while maintaining control over the devices associated with that service.
Potential ‘Gotchas’
The Google Voice service is only available in the US. Even US subscribers can only forward/simring their Google Voice numbers to other US numbers but that is likely to change to include international countries in the near future.
In theory, the Google Voice call should go wherever the media is sent. Call Routing results may vary depending on the Response Point ITSP you choose.
When calling out, your existing phone number (Caller ID) will be presented to the callee unless you use the dial-out feature, which is (IMHO) a bit of a hassle. This causes some problems as most of us are used to calling people back on the number we last saw from them. Fortunately, many ITSPs (unlike the conventional phone companies) will allow you to change your Caller ID number to match your Google Voice number.
Google Voice does not address LNP (Local Number Portability) at all right now. Which means you can not bring your existing numbers to Google Voice, you have to choose a new number.
Written by Erik Lagerway - Visit WebsiteSubscribe
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