On Sat, Mar 19, 2005 at 04:36:33PM +0100, Esben Stien wrote: > Eric Dantan Rzewnicki <rzewnickie@xxxxxxx> writes: > > what free software alternatives are available. > What I really need is a client daemon hooked up to jack that I can > interface to with a non-interactive interface or an interactive > ncurses app (and maybe a graphical gtk+ frontend for some eyecandy). > It would also be nice to have a sip softphone on my web page. I took a > look at mozphone, but have not played with it yet. > Proprietary software is not an option. Thank you to everyone who participated in this thread. I was able to get agreement in principle that free and open standards are better for the long term. But, since skype has the greatest market penetration at the moment it will be used in the short term as the quick and easy solution. Not my preferred outcome, but at least I was able to present a reasonably well informed argument with the help of this list. I think it was enough to plant a seed for the future. For what it's worth I'm attaching notes from last week's DCLUG meeting about asterisk and VOIP. For some reason the archives of that list are subscriber only ... Thanks, -- Eric Dantan Rzewnicki | Systems Administrator Technical Operations Division | Radio Free Asia 2025 M Street, NW | Washington, DC 20036 | 202-530-4900 CONFIDENTIAL COMMUNICATION This e-mail message is intended only for the use of the addressee and may contain information that is privileged and confidential. Any unauthorized dissemination, distribution, or copying is strictly prohibited. If you receive this transmission in error, please contact network@xxxxxxxx -------------- next part -------------- [dclug] Notes from DC LUG Meeting on March 16, 2004 = Asterisk/VoIP/telephony *Przemek Klosowski* przemek at jazz.ncnr.nist.gov <mailto:dclug%40tux.org?Subject=%5Bdclug%5D%20Notes%20from%20DC%20LUG%20Meeting%20on%20March%2016%2C%0A%092004%20%3D%20Asterisk/VoIP/telephony&In-Reply-To=> /Thu Mar 17 12:30:04 EST 2005/ * Previous message: [dclug] Linux Memory Addressing <006515.html> * Next message: [dclug] Notes from DC LUG Meeting on March 16, 2004 = Asterisk/VoIP/telephony <006514.html> * *Messages sorted by:* [ date ] <date.html#6511> [ thread ] <thread.html#6511> [ subject ] <subject.html#6511> [ author ] <author.html#6511> ------------------------------------------------------------------------ That was an excellent talk!!! Thanks, Justin! Here are the notes of Robert Burgoyne and Eric Smith, with my edits. Please add your own edits and send them to me, preferably in a diff or 'prose' (like in 'add this paragraph after the section xyz') Presenter: Justin B. Newman; he does cryptography and VoIP consulting, owns a VoIP company and an ISP (binhost & binfone): http://www.binfone.com/ http://www.binhost.com/ * Uptime requirements for PBX vendors are orders of magnitude more stringent than in the PC world (99.999% availability allows only 5 minutes/year downtime) * Business grade IP phones cost just as much as non-VoIP phones and have the same compatibility issues. * We as technical people are more comfortable with IP data infrastructure as compared with telco. There is no easy way for inquisitive people to learn more about telco protocols, infrastructure, etc. as books are not readily available you must somehow obtain direct industry experience as an employee. One recommended book: Understanding Telephone Electronics, by Fike, John L.; Friend, Georg E.; ISBN # 0-672-27018-8 (1983) Old book: may not cover newer digital stuff. * If you're doing all of your VoIP infrastructure with low cost hardware keep lots of spare hardware in reserve. Use UPS or two. * structure and terminology: - POTS (Plain Old Telephone Service, an official industry term): regular phones using analog signaling, one pair cable and RJ-11 modular connections - PSTN (Public Switched Telephone Network): regular phone network owned by the phone companies - FCC: tries to figure it out and regulate telco. Currently keeps mitts off new technologies like VOIP - BRI/PRI (Base and Primary Rate Interface) ISDN (digital): links that aggregate 24 encoded POTS phone channels; they can be ganged up for virtually unlimited number of DIDs (direct inward dial numbers); the number of DIDs you have is unrelated to your number of phone lines, which is generally 23 B channels and 1 D channel per PRI. Nationwide, carriers like Verizon (VZ) is oversubscribed 8-12x for trunk lines. The regulatory authorities are OK with this, because it is sufficient for normal usage patterns - FXO: ports that accept POTS into a digital PBX (gateway); doing FXS signaling. - ATA (analog telephone adapter): POTS plugs in, Internet/VoIP comes out. - softphone: plugging headphone/mike to a sound card. * Protocols for call setup on the VoIP network, ordered by market share: * SIP is the most dominant of the protocols. Has trouble with NAT. * H.323 is used mainly by carriers; is the least compatible between implementations. * IAX is the preferred native protocol by Asterisk; well designed but not implemented other than by Asterisk. * MGCP is another protocol. * Voice codecs determine the type of compression being used. Most codecs have some loss; G.711 (uLaw) is lossless, requires 64Kbps of media, including approximately 9Kbps of IP overhead. Considered to be a bandwidth hog, so other codecs are available which are less bandwidth intensive. G.729 is the predominant low-bandwidth codec, ~9Kbps per channel, but it is patented and you must pay the license fees, about $10 per concurrent channel. The licensing mechanism for G.729 on Asterisk is based on MAC addresses. Another codec in use is GSM (~ 30 Kbps), as in cell phones; yet few GSM carriers use the GSM codec for cell phones! Codecs should negotiate (need one between the phone and PBX/gateway; another PBX<->PBX, and the final one PBX->other phone. In real life that is iffy---often negotiation breaks down and drops the calls. * Possible sampling issues with sound files used by Asterisk for voice messages, etc.: when talking to low bandwidth phones like cell phones can cause breakup or static. * SNOM phones run Linux; good quality (Made in Germany) * Cisco ATA186 is the standard ATA to connect POTS phones to Ethernet. * Sipura 841 is the phone he recommends, and he recommends their Sipura 3000 ATA ($99) also. The ATA enables you to create a small dial plan for issues like 911. * Facilities based CLEC (Competitive Local Exchange Carriers) providing services to businesses are doing lots of interesting service offerings. Mentions Laurel Telecommunications & Doug. Alternative is ILEC (incumbents, usually ossified), and reseller CLECs (at the mercy of RBOC/ILEC, neither service nor skills) * Lines are totally unrelated to phone numbers except as per the regulatory framework. Numbers may or may not cross LATA (local area as seen by FCC) boundaries---CLECS are more creative. * Automatic Number Identification (ANI) similar to caller id, represents the source of the call. Most billing relies on ANI, however some entities do not assert ANI. * e911 is the great FUD Factor for VoIP. Justin does not believe that the location determination requirement can really be solved as you decouple the telephone from a physical location. * DC has a shadow phone system for 911 connectivity. Refers to the shadow phone system as the 911 tandem. * NEBS certified PCs are the highest grade of reliability for telco infrastructure. With that comment he took a jab at IDE disks. * VZ says that if we bring fiber to your home than we dont have to share any of the infrastructure with CLECs. * Asterisk can be configured to be an upstream PRI interface for your existing PBX; its just another interface for your exist PBX in addition to whatever PRI it already has from VZ, etc. Check out http://asteriskathome.sourceforge.net/: easy to deploy asterisk and home automation distribution * Call Detail Records (CDRs) are comma delimited files used for telco billing. This is referred to as call rating. He suggests that there is existing software which can be purchased to help you to do call rating---which is non-trivial because of lots of corner cases. ------------------------------------------------------------------------ * Previous message: [dclug] Linux Memory Addressing <006515.html> * Next message: [dclug] Notes from DC LUG Meeting on March 16, 2004 = Asterisk/VoIP/telephony <006514.html> * *Messages sorted by:* [ date ] <date.html#6511> [ thread ] <thread.html#6511> [ subject ] <subject.html#6511> [ author ] <author.html#6511> ------------------------------------------------------------------------ More information about the dclug mailing list <http://www.tux.org/mailman/listinfo/dclug>
