Re: [Asterisk-Users] InfoWeek Article on VOIP
Michael Graves wrote: Here's t link: http://www.informationweek.com/story/showArticle.jhtml;jsessionid=JUEFVG ENEA01YQSNDBCCKH0CJUMEKJVN?articleID=165702588 The bottom line is that they compare retail VOIP providers like Comcast Cable, Time-Warner Cable, ATT, Vonage, Packet8 et al. Their methodology seems sound. Their conclusion is that retail VOIP services don't yet match the PSTN for reliability call quality. It is interesting that all of these retail providers use ATA type devices. I wonder how some of the stronger true ITSPs like Level3 or even Nufone, VOIPJet, etc would fare, especially with an all digital scheme...ie hard IP phones. My own sense is that my IP base calls are cleaner than my SBC lines. I accept that they're less reliable, but much of that I attribute to the fact that I'm no Linux guru and I use a retail DSL line as my IP access. Michael Graves How do you see an ATA as different from and IP hardphone? As far as I can tell having the phone and ATA integrated isn't all THAT desirable, but that's me, I like to be able to choose the features on my phone and be able to connect it to the net... But that's just me. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] InfoWeek Article on VOIP
I agree with you but not 100% with them. An IP to Ip call on an ATA flat out is better . Now don't even get me started about cellular. My Service dosen't drop calls in the middle of conversations. VoIP is a notch better than Cellular. Michael Graves wrote: Here's t link: http://www.informationweek.com/story/showArticle.jhtml;jsessionid=JUEFVG ENEA01YQSNDBCCKH0CJUMEKJVN?articleID=165702588 The bottom line is that they compare retail VOIP providers like Comcast Cable, Time-Warner Cable, ATT, Vonage, Packet8 et al. Their methodology seems sound. Their conclusion is that retail VOIP services don't yet match the PSTN for reliability call quality. It is interesting that all of these retail providers use ATA type devices. I wonder how some of the stronger true ITSPs like Level3 or even Nufone, VOIPJet, etc would fare, especially with an all digital scheme...ie hard IP phones. My own sense is that my IP base calls are cleaner than my SBC lines. I accept that they're less reliable, but much of that I attribute to the fact that I'm no Linux guru and I use a retail DSL line as my IP access. Michael Graves -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] InfoWeek Article on VOIP
On Sat, 2005-07-16 at 10:12 -0700, Michael D Schelin wrote: I agree with you but not 100% with them. An IP to Ip call on an ATA flat out is better . Now don't even get me started about cellular. My Service dosen't drop calls in the middle of conversations. VoIP is a notch better than Cellular. What a lot of people dont consider with VoIP is the qualiuty of their ISP and how well connected their ISP is to everything else. My ISP for example (only game in town that isnt dialup) has 1 feed from sprint, I am guessing a T3 (I live in a rural area) and no QoS of any kind. So in general they suck for VoIP because of the latency they add to the link. Many people I have talked to think internet access is internet access and the contention rate is never thought of. This greatly affects any review of VoIP. Of course a private IP network (again a lot of people think VoIP as voice over the internet not thinking about private networks) is usually better because it can be tweaked for voice apps specifically. Even if you dont have a private network adjusting packet size and jitter buffers for that link specifically can increase performance. It ends up being more than just tossing a box on the net with asterisk or whatever on it. Now that I think about it I havent looked anywhere for network performance tuning for voice apps, does voip-info have a wiki page? If not perhaps it should with general properties based on link types and all and possibly specifics for certain operating systems and/or network equipment. Since updating wikis is against my religion I am unable to do this (strict religion, forbids me from contributing to any GPL project - forced to release my code BSD style if free, or updating wikis). But there are enough people that do not follow the same religion as me. That may help with performance all around, and increase the user experience. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] InfoWeek Article on VOIP
On Sat, 16 Jul 2005 10:10:29 -0700, Bruce Ferrell wrote: Michael Graves wrote: Here's t link: http://www.informationweek.com/story/showArticle.jhtml;jsessionid=JUEFVG ENEA01YQSNDBCCKH0CJUMEKJVN?articleID=165702588 The bottom line is that they compare retail VOIP providers like Comcast Cable, Time-Warner Cable, ATT, Vonage, Packet8 et al. Their methodology seems sound. Their conclusion is that retail VOIP services don't yet match the PSTN for reliability call quality. It is interesting that all of these retail providers use ATA type devices. I wonder how some of the stronger true ITSPs like Level3 or even Nufone, VOIPJet, etc would fare, especially with an all digital scheme...ie hard IP phones. My own sense is that my IP base calls are cleaner than my SBC lines. I accept that they're less reliable, but much of that I attribute to the fact that I'm no Linux guru and I use a retail DSL line as my IP access. Michael Graves How do you see an ATA as different from and IP hardphone? As far as I can tell having the phone and ATA integrated isn't all THAT desirable, but that's me, I like to be able to choose the features on my phone and be able to connect it to the net... But that's just me. I have personally used Cisco ATAs, Sipura-2000s and 3000s. When I begin investigating switching to IP phones I tried Pingtel, Grandstream, Zultys, Snom and Polycom. To be fair I used each one for a couple of months, often as my primary desk phone if it looked like the device would cut it. I settled on Polycom 600s and 500s for my home office. I only have 5 phones. As someone who works from a home office professionally I feel that the call quality, multi-line capability and availability of serious business features are important. For a while , before I had a production * server, I had a pair of Sipura units connected to a 4 line Panasonic KSU system. The Polycom's simply sounded best, feel best in the hand, and have the on-board tools that I use daily. ATAs just don't go that far for me. I don't see it as having the phone and the ATA integrated. It's a SIP phone. Asterisk sees it as something slightly different than an ATA. I may have multiple registrations, of which several may be in use at once. It supports simple SMS stlye messaging. Heck the IP600 even has a micro-browser built into it, although I've not used this yet myself. I agree with others who have chimed in that IP-to-IP calls can sound better than PSTN calls. I have a co-worker who has a SipGate account in the UK. Calls to him via SipGate go out through my FreeWorldDialup account. They sound great. So good that in silent moments we often think that we've been severed, even with no silence suppression on the line. It really would be great to have a truly wideband codec available within Asterisk. I recall reading that the wideband version of iLBC is not released under GPL. Anyone know more about this? Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] InfoWeek Article on VOIP
On Sat, 2005-07-16 at 17:05 -0500, Michael Graves wrote: I agree with others who have chimed in that IP-to-IP calls can sound better than PSTN calls. I have a co-worker who has a SipGate account in the UK. Calls to him via SipGate go out through my FreeWorldDialup account. They sound great. So good that in silent moments we often think that we've been severed, even with no silence suppression on the line. One thing that many PSTN providers are doing for calls when they went digital is to insert small quantities of noise into the line. That way people do not think they are disconnected. There is a bunch of documentation on this, and even some that applies to VoIP providers/equipment doing the same (its basically a faint bit of white noise so you hear *something*). -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] InfoWeek Article on VOIP
trixter http://www.0xdecafbad.com wrote: On Sat, 2005-07-16 at 17:05 -0500, Michael Graves wrote: I agree with others who have chimed in that IP-to-IP calls can sound better than PSTN calls. I have a co-worker who has a SipGate account in the UK. Calls to him via SipGate go out through my FreeWorldDialup account. They sound great. So good that in silent moments we often think that we've been severed, even with no silence suppression on the line. One thing that many PSTN providers are doing for calls when they went digital is to insert small quantities of noise into the line. That way people do not think they are disconnected. There is a bunch of documentation on this, and even some that applies to VoIP providers/equipment doing the same (its basically a faint bit of white noise so you hear *something*). It's sometimes called comfort noise... As far as I'm aware, it's only done in VoIP. I spent 15 years working with digital switches/T1 channel banks. I guess it might have been built in and I just didn't know about it, but we were very concerned about excess noise and quantization noise as it was. We used to inject a 1004 test tone and then use a notch filter to measure the amount of quantization noise at the reciever. Just as a by the by, G.711ulaw is the codec used in channel banks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] InfoWeek Article on VOIP
On Sat, 2005-07-16 at 16:12 -0700, Bruce Ferrell wrote: It's sometimes called comfort noise... As far as I'm aware, it's only done in VoIP. I spent 15 years working with digital switches/T1 channel banks. I guess it might have been built in and I just didn't know about it, but we were very concerned about excess noise and quantization noise as it was. We used to inject a 1004 test tone and then use a notch filter to measure the amount of quantization noise at the reciever. I have seen it done at the switches that telcos use, not at the end user such as in a channel bank. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
