Cost savings on trunks is not exclusive to ITSP's and SIP trunks. The providers of ISDN did not just roll over and abandon their networks, they reduced their prices and are rigorously competing with the ITSPs. In many cases, PRI can be much less than SIP trunks, and provide for that ultimate, high quality telephone experience. And there are companies that have bundles packages of T1 or Ethernet over Copper that competes well.
To sell someone low cost SIP trunks, that are not designed reliably, whether on sipXecs or Cisco, Avaya or any other brand name, makes little or no sense. Value is not getting what you pay for, but something that performs as well as products at higher costs. From: [email protected] [mailto:[email protected]] On Behalf Of Tim Ingalls Sent: Friday, February 17, 2012 7:25 PM To: Discussion list for users of sipXecs software Subject: Re: [sipx-users] sipXecs Commercial Feasibility Thanks for the great comments. I agree that QOS is difficult and PSTN connections typically just work. But I'm trying to leverage the cost savings of using SIP trunking with an ITSP in order to help customers justify switching their phone system. In this tough economy, saving a customer 50% of their monthly phone bill can make the difference between selling a system and not selling one. I have a few questions, below: Thanks, Tim Ingalls Shared Communications, Inc. 801-618-2102 Office On 02/16/2012 09:19 PM, Becker, Jesse wrote: I would have to agree with Tim and state that most of your issues are most likely ISTP related and/or network related. I am not one of the developers, so I am not defensive, however, I will say that I have using and testing SipX on and off since 2004 (going back to the Pingtel days). The one thing I can say is that all of my deployments and labs have utilized physical PSTN connections (PRI and or Copper). Keep in mind there is no QOS on the internet. With proper engineering you can control level of QOS on your network and to QoS going to your firewall, however, everything after that is hope and pray. That's a good point. What PSTN gateway(s) are you using? Are there any tricks to configuring it to work with sipXecs? I have a carrier I plan to use (Airespring) that uses MPLS to deploy their SIP trunking. With that system, QOS across to the ITSP should not be a problem. I am currently using SipX in my IT department and so far it performs ways better then the 400K Cisco phone system that the college implemented five years ago. We currently are using SIP trunking between the two systems allow interop of calling between Cisco and SipX users. We are currently planning on replacing the Cisco system and going with a 3-server redundant system campus wide. A couple of suggestions/points I can offer 1. Make sure your ITSP is sending your inbound calls to 5080 and not 5060. This is required to use the building SBC of SipX 2. Let your SipX server take care of all your NAT transversal. Turn off any built in VoIP transversal setting that may be build into your firewall Does that suggestion include not using any port forwarding rules (5080 and 5060)? I briefly switched off my port forwarding rules for 5060 and 5080 and everything kept working. I put it back just to be safe. 3. Configure your firewall so that only your ITSP's gateways can communicate with you. Leaving yourself exposed allow you to be DOS attacked by others on the internet who use robotos to exploit you for toll fraud. It is possible that their denial of service attempts could cause issues to your proxy and registar service. I assume doing that on my Tomato Firmware router would mean either using custom firewall rules or implementing port forwarding while identifying the source IP address. My issue with that is that I wonder if the same IP address will always be initiating the session from the ITSP. If they have several servers, wouldn't it be possible that the packets will come from multiple IP addresses? 4. If you have the capability, give your SIP traffic and RTP traffic priority or guarantee on firewall. Heavy regular internet use could cause delay or loss of your call traffic if you do not have QOS on your network and firewall. I used to use QOS settings, but I discontinued that practice after I switched to a 20Mbps connection from Comcast. Inbound QOS doesn't really work that well anyway when prioritizing from the WAN side. If you have a specific solution for that, I'm all ears. Switch to a PRI or copper lines if you don't want to deal with QoS etc. I have installed and used various telecom systems over the last 14 years. I will not stand on a soup box and say that SipX is 100% bug free, however, I will take SipX over any other open-source or commercial product any day. I will also say that I have given and received better support on this list that I every received from a 25,000 dollar Cisco smartnet contract. I know of a local network of community college campuses that have been trying to implement Cisco call manager for the past 3 years without success. The guy who runs the IT shop is certified in Cisco, so that's what he wants to use. Just my 2 cents. I truly appreciate your insight, Jes. Jes On Thu, Feb 16, 2012 at 9:58 PM, Todd Hodgen <[email protected]> wrote: Tim, This type of email is disappointing for me. They highlight negative experiences that are experienced by a user, and do nothing to help this project. I'm sure you are being open and honest, but all that it shows is that you are struggling with the system, in your deployment, and it gives a black eye to a very well design platform that a lot of dedicated individuals have poured their heart and soul into. I can tell you that I have many commercial customers that use sipXecs daily, and I have ZERO issues to deal with on a daily basis. But, how did I get to this point - many years of experience with telephony, understanding the components that are required for engineering a great solution, and time and patience in a lab environment to understand what is commercially viable for my customers. This is a great solution, but it is not for everyone. I also have two other commercial solutions available for my customers, which I use when sipXecs is not a perfect fit. However, that is rare. In general, the majority of the issues I have ever had are related to ITSP reliability, QOS related to the Network, occasional software issues with Telephones, and the occasion bug in sipXecs, like any other software platform. The sipXecs issues have never been a show stopper for any of my customers. I can't be any more honest or direct about my experience. There are resources available for a fee that can help your learning curve, or the eZuce platform may be a better fit for you. I can say with certainty, this is a great product, that I have no fear in offering to a commercial customer, and know they will be 100% satisfied when the project is done. Most of my business is referrals from very happy customers. It really is that good! My advice is understand the product, invest in excellent equipment, and chose the network design and the ITSP or Telco carefully. Mindful that this is an open source platform, it would be very rare for me to deploy a new software release until it has been vetted in the field for at least a few months or more. I am right now updating 4.2.1 customers to 4.4, although I started deploying 4.4 in the August timeframe for new customers. Not everyone is a good fit for offering open source products as a commercial option, that is why there are so many options out there, and I would sincerely consider eZuce as one of them, with the right design, the right components, and the right technical expertise to offer it to a commercial customer. Quite frankly, the trace process for sipXecs, along with features of Wireshark, and other applications commercially available on the market seem to work well for isolationing issues. Is it perfect, not by a long shot. But I have seen less in much more expensive products on the market today. From: [email protected] [mailto:[email protected]] On Behalf Of Tim Ingalls Sent: Thursday, February 16, 2012 6:03 PM To: sipx-users list Subject: [sipx-users] sipXecs Commercial Feasibility Hi Everyone. I promise I am not trying to be a troll. I have some serious questions that I hope I can ask honestly and get some honest feedback about using the free version of sipXecs as a commercial product. I implemented sipXecs about a year ago. My hope was to find something more reliable than Asterisk/Trixbox/FreePBX and easier to deploy. My purpose was to start selling phone systems and SIP trunking to businesses as a VAR. So far, after testing the system day in and day out as my home/home-office phone system, I haven't found it stable enough to feel comfortable selling it to customers. I have had a host of issues with sipXecs, and every time I think I've got the platform stable, something else fails and I get one of those barely-descriptive error messages in my email inbox. I've followed the instructions from the book, the wiki, and this forum, but still have issues every month. Some of the issues are as follows: * Routing inbound calls to an auto-attendant worked great for a long time and then just stopped working one day. After connecting the call, visitors were greeted with no sound at all. I decided after hours of trying everything to just skip the auto-attendant and deactivate it. * With both Vitelity and Voip.ms, I have problems where periodically an authentication request is rejected. Instead of re-trying immediately, sipXecs waits a full 10 minutes to try to connect again. * Nuances about how certain ITSPs (e.g., Vitelity and Voip.ms) work, and how you can and cannot connect to them without getting strange behavior like inbound audio not working, rejected authentication requests, etc., take days and weeks to isolate sometimes. These are not very well tested nor documented. I think that a serious effort at interop testing and certification should be undertaken with detailed results --warts and all-- posted so that someone can make an educated decision when selecting an ITSP to use with sipXecs. * Just a few days ago, calls that were transfered to voicemail resulted in the call failing and the ITSP routing the call to my failover phone number (my cell phone) -- this is after the call initially rang correctly. Rebooting the system fixed it for some reason. Why? * Periodically, (perhaps due to a sipviscious attack) certain services just stop working. Sometimes it is the proxy service. Sometimes it is the registrar service. Sometimes it is the NAT traversal feature as a result of temporarily not being able to reach the STUN server assigned (since there is no back-up STUN server setting). Why should these services just fail and require human intervention to restart them? Can't they just time out for a certain short period and then fix themselves? * CID doesn't work reliably. I change all of the settings as I'm told in the wiki, but it still doesn't get transmitted correctly (or at all). For some of my users, it works flawlessly, and for others it doesn't work at all. * Doing a SIP trace to isolate an issue is a pain in the neck. In Asterisk, all you have to do is type "asterisk -rvv" and you can see a dialog stream which you can read quickly. With sipXecs, you have to run a series of research tasks to find the call in question, convert the time to UTC, grep for the time stamp in a big list of calls, then create a merged XML file, then load it into SIPViewer, and then find what you are looking for. The process takes at least 5 minutes if you are an expert. Those are just a few examples. I'm always wondering what is going to go wrong next. It drives me (and my wife and kids) crazy. I never had this many problems with Trixbox. I'm not saying that sipXecs doesn't have its good points. I'm just saying that over the last year+ since I started using 4.2 and then 4.4, it has been anything but reliable. Reliability is the number one need for commercial clients. Yes, I'll admit that it could all be my fault. It probably is. But there are so many options, so many opinions, so many sources of information, (there are even so many places to set port numbers for various things) that it seems you have to do only sipXecs development for a living to be able to deploy it correctly. It is far from simple. And that complexity is part of the problem. I know that some of you have deployed many of these systems in a commercial setting, so I have to ask you, how do you do it? I'm too afraid that if I deploy sipXecs in an actual customer's location that they'll hate me within a few months and ask for their money back. How do you set everything up (selection of ITSP, etc.) so that the system is rock-solid reliable? Can we collect some rock-solid fool-proof (as much as possible) recipes that are known to work reliably every time? This seems to be something that should be placed on the wiki. I know that there are 100+ ways to configure the system (SIP trunking gateway configs, various hardware, ITSP settings, dial rules, etc.). I'm looking for just the recipes that make the system reliable. I also know that there are various conflicting opinions on this forum about what works and what doesn't. I'm looking for PROVEN opinions. This is my final shot before I give up on the platform. I'd even be willing to partner with someone who has a near-flawless system implemented and pay you to do the technical part if you can prove your solution is stable. Until I find the answer to this problem, I can't use sipXecs as the cornerstone of my business plan and will have to move on. If I can solve this issue, I'd be willing to pay for further development out of my profits. I know someone will suggest that I should just sell Ezuce's commercial products. Based on what I've experienced so far, I don't think I'd feel confident in relying on Ezuce to be the partner in question. If the open-source version has these problems, what's to say that the commercial version is any better? Does anyone else experience the same reliability issues? Also, is anyone willing to have a phone conversation about this and impart some wisdom or have a partnership conversation? -- Thanks, Tim Ingalls Shared Communications, Inc. 801-618-2102 Office _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- <https://www.google.com/a/cpanel/sunyulster.edu/images/logo.gif?service=mail > Jesse Becker | Technical Manager Network+ | Linux+ Certified Professional DATATEL+SGHE @ SUNY Ulster 491 Cottekill Road, Stone Ridge, NY 12484 Tel 845-687-5064 | Fax 845-687-5105 <mailto:[email protected]> [email protected] | <http://www.sunyulster.edu/> www.sunyulster.edu Check out our knowledge base: <http://kb.sunyulster.edu/> http://kb.sunyulster.edu _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
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