Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On Tue, Feb 28, 2012 at 8:28 PM, Alejandro Imass a...@p2ee.org wrote: What you are saying seems impossible and makes no sense unless the router is assigning a public IP or is SIP aware and knows how to read the routing data contained inside the SIP packets, and none of the consumer routers are SIP aware AFAIK, especially not the WRT-54G. You can: a. Continue to tell us that what we are doing every day is impossible. or... b. Go set up a test in your lab with an Asterisk server, a cheap router, and a SIP client and see. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On Tue, Feb 28, 2012 at 8:58 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: [...] Yes, I have had no problems with Grandstream first gen ATAs, configured with server and credentials and shipped off, they just work. We use the HT-286, the server is on a public IP the nat setting on asterisk is set to yes and without port re-direction the ATAs have never connected from a private network, so I honestly find this SIP plug and play very hard to believe. But if it is true, then maybe you can actually help us figure out all the NAT issues we've had with SIP for the past 5 years. Perhaps, it is simply ignorance on our side and we have something fundamentally wrong in our set-up somewhere that may be have been causing these issues with NAT. Our set-up is fundamentally public and private Asterisk servers running on FreeBSD. Versions may vary from FBSD 7 thru 8.2 and Asterisk 1.4 and 1.6. We are planning to upgrade every server to FBSD 8.2 and Asterisk 1.8 but we are in that process right now. Some Asterisk run in jails so I can understand the NAT issues there may be caused by the server itself. I honestly *love* your OpenVPN idea but I have to find a cheap ATA that could run as an OpenVPN client. Taking the simplest example a simple Asterisk 1.6 server on a public IP running on the base system (not in a jail): We run an operation that spans several countries including Canada, the US and the Latin American Andean region. As examples, with Canadian ISPs such as Rogers and Bell we have always had to redirect the ports and use STUN server for the HT-286 to register to the Asterisk server. In the US we have the same problem with Comcast networks, so I don't understand how you say that you plug a Grandtream SIP ATA to a Comcast router and it just works. However, in a couple of NOLA countries the ISP's routers actually give public IPs, so if the SIP ATAs are connected directly to the ISP router, or in the DMZ then it just works as you say, BUT if the ATA is connected behind the firewall, or to a WiFi router, then we've _allways_ had to redirect ports. In every sigle customer we have had to send instructions on how to redirect ports, and of course to configure firewall if present. I just don't understand how you and other here say that a SIP ATA can just work. On the contrarty, with IAX2 using cheap AG-188N from Atcom they are just plug and play when shipped with a standard conf, and we have none of the quality issues you are referring to. We do have some call drops however, and some hangup problems but they don't affect our clients as much as having to deal with NAT issues. We may not run 15K extensions like you but I think we have a pretty good testing ground and have dealt with a fair share of NAT problems with SIP, that you and others here apparently don't have, so I am as amazed by your likeness of SIP as perhaps you are amazed as our likeness of IAX. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On 02/29/2012 08:22 AM, Alejandro Imass wrote: We use the HT-286, the server is on a public IP the nat setting on asterisk is set to yes and without port re-direction the ATAs have never connected from a private network, so I honestly find this SIP plug and play very hard to believe. But if it is true, then maybe you can actually help us figure out all the NAT issues we've had with SIP for the past 5 years. Perhaps, it is simply ignorance on our side and we have something fundamentally wrong in our set-up somewhere that may be have been causing these issues with NAT. The number of 'plain' SIP endpoints deployed behind consumer-grade NAT devices talking to Asterisk servers on public IP addresses is in the millions, if not the tens of millions. As has already been posted, Asterisk itself handles all the far-end NAT traversal duties necessary for this to work; neither the remote endpoint nor the NAT device need to do anything special, nor do they require any configuration. Rather than post a lengthy exposition on how widespread your network is and how technically astute your people are, you would probably accomplish much more to setup a simple test scenario as has been previously suggested, and if it does not work for you, post the details of the scenario and the failure here. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On Wed, Feb 29, 2012 at 9:44 AM, Kevin P. Fleming kpflem...@digium.com wrote: On 02/29/2012 08:22 AM, Alejandro Imass wrote: [...] The number of 'plain' SIP endpoints deployed behind consumer-grade NAT devices talking to Asterisk servers on public IP addresses is in the millions, if not the tens of millions. As has already been posted, Asterisk itself handles all the far-end NAT traversal duties necessary for this to work; neither the remote endpoint nor the NAT device need to do anything special, nor do they require any configuration. Rather than post a lengthy exposition on how widespread your network is and how technically astute your people are, you would probably accomplish much more to setup a simple test scenario as has been previously suggested, and if it does not work for you, post the details of the scenario and the failure here. We use SIP and IAX interchangeably, but had less hassle with IAX. The topic of the discussion on this thread was that SIP is so awesome and that IAX is a peice of crap. My point of view is that we've had many problems with SIP and NAT and that IAX just works great for us, and that in *our* experience IAX has worked better for us. Just to clear my head up a bit: are you supporting the argument that SIP is better for Asterisk than IAX? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On 02/29/2012 09:25 AM, Alejandro Imass wrote: On Wed, Feb 29, 2012 at 9:44 AM, Kevin P. Flemingkpflem...@digium.com wrote: On 02/29/2012 08:22 AM, Alejandro Imass wrote: [...] The number of 'plain' SIP endpoints deployed behind consumer-grade NAT devices talking to Asterisk servers on public IP addresses is in the millions, if not the tens of millions. As has already been posted, Asterisk itself handles all the far-end NAT traversal duties necessary for this to work; neither the remote endpoint nor the NAT device need to do anything special, nor do they require any configuration. Rather than post a lengthy exposition on how widespread your network is and how technically astute your people are, you would probably accomplish much more to setup a simple test scenario as has been previously suggested, and if it does not work for you, post the details of the scenario and the failure here. We use SIP and IAX interchangeably, but had less hassle with IAX. The topic of the discussion on this thread was that SIP is so awesome and that IAX is a peice of crap. My point of view is that we've had many problems with SIP and NAT and that IAX just works great for us, and that in *our* experience IAX has worked better for us. Just to clear my head up a bit: are you supporting the argument that SIP is better for Asterisk than IAX? I have no idea where you got that sort of conclusion. I was making a statement to counter your repeated arguments that using SIP behind a NAT without special configuration is 'impossible'. It's clearly not impossible, it's not even impractical. It is commonplace. Certainly there are plenty of examples of SIP endpoints working poorly behind NAT devices, and replacing that endpoint with an IAX2 endpoint curing the symptoms. Invariably, this is caused by the fact that the NAT device was attempting to 'help' the SIP endpoint, and failed miserably. In every case I can remember, turning off any SIP-specific functionality in that NAT device (which is not always possible) allowed the SIP endpoint to work as expected. There are certainly scenarios where deploying a SIP endpoint behind a NAT can be problematic; usually, these revolve around deploying a SIP *server* behind a NAT, but even this can be handled reasonably well by configuration options already present in Asterisk. Deploying SIP *clients* behind NATs, talking to a SIP server that is on a public IP, is generally trivial and takes no special effort at all. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On Wed, Feb 29, 2012 at 9:22 AM, Alejandro Imass a...@p2ee.org wrote: On Tue, Feb 28, 2012 at 8:58 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: [...] Yes, I have had no problems with Grandstream first gen ATAs, configured with server and credentials and shipped off, they just work. We use the HT-286, the server is on a public IP the nat setting on asterisk is set to yes and without port re-direction the ATAs have never connected from a private network, so I honestly find this SIP plug and play very hard to believe. But if it is true, then maybe you can actually help us figure out all the NAT issues we've had with SIP for the past 5 years. Perhaps, it is simply ignorance on our side and we have something fundamentally wrong in our set-up somewhere that may be have been causing these issues with NAT. Our set-up is fundamentally public and private Asterisk servers running on FreeBSD. Versions may vary from FBSD 7 thru 8.2 and Asterisk 1.4 and 1.6. We are planning to upgrade every server to FBSD 8.2 and Asterisk 1.8 but we are in that process right now. Some Asterisk run in jails so I can understand the NAT issues there may be caused by the server itself. I honestly *love* your OpenVPN idea but I have to find a cheap ATA that could run as an OpenVPN client. Taking the simplest example a simple Asterisk 1.6 server on a public IP running on the base system (not in a jail): We run an operation that spans several countries including Canada, the US and the Latin American Andean region. As examples, with Canadian ISPs such as Rogers and Bell we have always had to redirect the ports and use STUN server for the HT-286 to register to the Asterisk server. In the US we have the same problem with Comcast networks, so I don't understand how you say that you plug a Grandtream SIP ATA to a Comcast router and it just works. However, in a couple of NOLA countries the ISP's routers actually give public IPs, so if the SIP ATAs are connected directly to the ISP router, or in the DMZ then it just works as you say, BUT if the ATA is connected behind the firewall, or to a WiFi router, then we've _allways_ had to redirect ports. In every sigle customer we have had to send instructions on how to redirect ports, and of course to configure firewall if present. I just don't understand how you and other here say that a SIP ATA can just work. On the contrarty, with IAX2 using cheap AG-188N from Atcom they are just plug and play when shipped with a standard conf, and we have none of the quality issues you are referring to. We do have some call drops however, and some hangup problems but they don't affect our clients as much as having to deal with NAT issues. We may not run 15K extensions like you but I think we have a pretty good testing ground and have dealt with a fair share of NAT problems with SIP, that you and others here apparently don't have, so I am as amazed by your likeness of SIP as perhaps you are amazed as our likeness of IAX. If you can post some SIP debug info from an ATA trying to register without any redirection and also the relevant portions of your sip.conf, I am sure I can help. Do it from a new location with an el cheapo home router, Linksys WRTXXX. If I cannot help you in a few emails, we can take this offline. Actually paste your entire sip.conf in pastebin or something, as well as sip debug. Also the configs of your ATAs. I think you have over-engineered to the point of creating problems. This is very common. My philosophy is KISS Thanks, Steve T -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On Wed, Feb 29, 2012 at 8:34 AM, Kevin P. Fleming kpflem...@digium.com wrote: Certainly there are plenty of examples of SIP endpoints working poorly behind NAT devices, and replacing that endpoint with an IAX2 endpoint curing the symptoms. Invariably, this is caused by the fact that the NAT device was attempting to 'help' the SIP endpoint, and failed miserably. In every case I can remember, turning off any SIP-specific functionality in that NAT device (which is not always possible) allowed the SIP endpoint to work as expected. We have *never* found a single SIP helper to actually help anything. They always break everything. The only SIP-related setting I can think of that works is Enable consistent NAT found in Sonicwall routers (but turn off all other SIP helpers). The WRT54G and Apple Airport come to mind as stable and reliable home/small business routers that seem to just work. The WRTs seem to go bad every few years and need an occasional reboot, the Airports work forever in our experience without being touched. All stock out of the box. options already present in Asterisk. Deploying SIP *clients* behind NATs, talking to a SIP server that is on a public IP, is generally trivial and takes no special effort at all. This is today's reality. That's why I mentioned the possibility that he's using ancient devices and routers, or that Yugo. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On Wed, Feb 29, 2012 at 9:44 AM, Kevin P. Fleming kpflem...@digium.comwrote: On 02/29/2012 08:22 AM, Alejandro Imass wrote: We use the HT-286, the server is on a public IP the nat setting on asterisk is set to yes and without port re-direction the ATAs have never connected from a private network, so I honestly find this SIP plug and play very hard to believe. But if it is true, then maybe you can actually help us figure out all the NAT issues we've had with SIP for the past 5 years. Perhaps, it is simply ignorance on our side and we have something fundamentally wrong in our set-up somewhere that may be have been causing these issues with NAT. The number of 'plain' SIP endpoints deployed behind consumer-grade NAT devices talking to Asterisk servers on public IP addresses is in the millions, if not the tens of millions. As has already been posted, Asterisk itself handles all the far-end NAT traversal duties necessary for this to work; neither the remote endpoint nor the NAT device need to do anything special, nor do they require any configuration. Rather than post a lengthy exposition on how widespread your network is and how technically astute your people are, you would probably accomplish much more to setup a simple test scenario as has been previously suggested, and if it does not work for you, post the details of the scenario and the failure here. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org Agreed with one exception, the endpoint behind the NAT DOES need to be setup correctly to keep the router from seeing inbound traffic to the device as unsolicited and drop it. That is a function of the router but keep alives from Qualify on the Asterisk side, and setting the device to register every few minutes will keep that mapping open and alive, letting traffic pass as solicited. Thanks, Steve Totaro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On Wed, Feb 29, 2012 at 8:41 AM, Steve Totaro stot...@asteriskhelpdesk.com wrote: Agreed with one exception, the endpoint behind the NAT DOES need to be setup correctly to keep the router from seeing inbound traffic to the device as unsolicited and drop it. That is a function of the router but keep alives from Qualify on the Asterisk side, and setting the device to register every few minutes will keep that mapping open and alive, letting traffic pass as solicited. We use qualify=yes on Asterisk and a few months ago turned OFF the keep-alive feature on all SIP clients on our entire system. This is working fine, and we did it because of a strange bug/behavior with certain versions of Cisco SPA series firmware. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On Wed, Feb 29, 2012 at 10:43 AM, Carlos Alvarez car...@televolve.comwrote: On Wed, Feb 29, 2012 at 8:41 AM, Steve Totaro stot...@asteriskhelpdesk.com wrote: Agreed with one exception, the endpoint behind the NAT DOES need to be setup correctly to keep the router from seeing inbound traffic to the device as unsolicited and drop it. That is a function of the router but keep alives from Qualify on the Asterisk side, and setting the device to register every few minutes will keep that mapping open and alive, letting traffic pass as solicited. We use qualify=yes on Asterisk and a few months ago turned OFF the keep-alive feature on all SIP clients on our entire system. This is working fine, and we did it because of a strange bug/behavior with certain versions of Cisco SPA series firmware. -- Carlos Alvarez TelEvolve 602-889-3003 So you turned it off on the phones but use it on the Asterisk side? Do you set a value or just use qualify=yes? I had many problems with qualify over VSAT as ping times and jitter are crazy. 700ms ping times were considered Good from the IZ in Iraq to Equinix data center in VA, it took some tweaking to find the right value so a phone that was Reachable was not labeled Unreachable, I did want phones that were truly unreachable to be marked as such, more to spot patterns and act on them or with the vendor. Did you submit a bug report? If it is easy to reproduce and you feel like helping out, report it. I do not report issues if there is a simple way to do the same thing, but I know I should. What does the debug or strange behavior look like? Probably a variance in the RFC implementation. Thanks, Steve T -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On Wed, Feb 29, 2012 at 8:58 AM, Steve Totaro stot...@asteriskhelpdesk.com wrote: So you turned it off on the phones but use it on the Asterisk side? Do you set a value or just use qualify=yes? Yes, just as I said, just qualify=yes. Did you submit a bug report? If it is easy to reproduce and you feel like helping out, report it. I do not report issues if there is a simple way to do the same thing, but I know I should. Cisco makes it too difficult to submit bugs so I just don't care to help them. When we find a service-impacting bug we report it to our distributor, who tests it and presumably reports it, but I'm not sure. Also I wasn't sure if it was an Asterisk bug or Cisco bug, and didn't care enough to find out since a clean work-around was possible. What does the debug or strange behavior look like? Probably a variance in the RFC implementation. Soon as the keep-alive packet is sent, the phone is no longer reachable from Asterisk. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On 2012-02-28 21:22:44 +, Kevin P. Fleming said: On 02/28/2012 03:08 PM, Troy Telford wrote: [myprovider] type=friend username= secret= context=somecontext host=provider_server qualify=1000 disallow=all allow=g729 allow=ulaw auth=md5,rsa requirecalltoken=yes trunk=yes A serious bug with IAX2 trunking in recent versions of Asterisk (you did not mention what version you are using) was just resolved last week. You should test with 'trunk=no' to see if that is the cause of your problem; it seems very likely. trunk=yes was the source of the problem. So now I suppose I'll have trunk=no while I patiently wait for the fix to appear in Debian. - As an aside: I'm perfectly capable of compiling Asterisk; I prefer to use the packages for pretty much all of the reasons packages were invented. -- Troy Telford -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On 02/29/2012 11:35 AM, Troy Telford wrote: On 2012-02-28 21:22:44 +, Kevin P. Fleming said: On 02/28/2012 03:08 PM, Troy Telford wrote: [myprovider] type=friend username= secret= context=somecontext host=provider_server qualify=1000 disallow=all allow=g729 allow=ulaw auth=md5,rsa requirecalltoken=yes trunk=yes A serious bug with IAX2 trunking in recent versions of Asterisk (you did not mention what version you are using) was just resolved last week. You should test with 'trunk=no' to see if that is the cause of your problem; it seems very likely. trunk=yes was the source of the problem. Thanks for following up! The patch to resolve this problem was very small, but I understand your desire to wait for a package. It will be in the 1.8.11 release, although of course the Debian team could choose to backport the one-line fix into their existing release if they so choose. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On 2012-02-29 15:25:49 +, Alejandro Imass said: We use SIP and IAX interchangeably, but had less hassle with IAX. The topic of the discussion on this thread was that SIP is so awesome and that IAX is a peice of crap. The original question (mine) was that my sound quality when using IAX was bad; with SIP the sound quality was great. Critically, I mentioned that I wanted to use IAX; I even said I was willing to do some self torture to get IAX working properly. I only wanted some help in figuring out what was 'wrong' with my IAX configuration. After a few suggestions, Kevin Fleming noticed I was using trunk=yes, and it was likely that my Asterisk install was being affected by a just-fixed bug. Disabling trunking fixed the problem - the voice sounds great even in my worst-case scenerio (which was always almost unintelligible). The devolution into a flamewar is unfortunate, but such things are inevitable whenever a 'this' vs 'that' question is posed. For instance, is the Yugo really any worse than the competing Trabant? The only correct answer is to fling them both with a Trebuchet and see which one flies farther. -- Troy Telford -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On Wed, Feb 29, 2012 at 10:34 AM, Steve Totaro stot...@asteriskhelpdesk.com wrote: [...] If you can post some SIP debug info from an ATA trying to register without any redirection and also the relevant portions of your sip.conf, I am sure I can help. Do it from a new location with an el cheapo home router, Linksys WRTXXX. Yeah, I think it's time for me to shut up about SIP/NAT problems and, like you Carlos and Kevin pointed out, run a clean un-contaminated test lab to see if we can determine why our current set-up is so problematic with SIP and NAT. If I cannot help you in a few emails, we can take this offline. Thanks for offering to help. I will set-up a test lab but it's gonna take me some time to free a public server to do so. But it is obvious that the problem is on our side after reading all the responses. After all, VoIP is *not* by any means our core business we just use it as a tool, and up until now I thought that *everyone* using SIP ATAs and Asterisk had these NAT woes, so we just assumed it was so, and thought that mostly everyone had to perform particular configurations on the endpoints. It now seems obvious we are wrong. Anyway, my whole argumentative line in this thread is that in our particular case we found that IAX2 works great for _our_ set-ups and we don't share the view that IAX2 is a broken bat, and that in fact for us it just works great. Thanks, -- Alejandro Imass -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On Wed, Feb 29, 2012 at 1:05 PM, Troy Telford ttelford.gro...@gmail.com wrote: On 2012-02-29 15:25:49 +, Alejandro Imass said: We use SIP and IAX interchangeably, but had less hassle with IAX. The topic of the discussion on this thread was that SIP is so awesome and that IAX is a peice of crap. The original question (mine) was that my sound quality when using IAX was bad; with SIP the sound quality was great. Critically, I mentioned that I wanted to use IAX; I even said I was willing to do some self torture to get IAX working properly. Yeah, I wasn't referring particularly to the original post, just the way the thread turned against IAX like if it's not a viable solution and my point all along has been that for *us* IAX2 endpoints have worked better and easier to configure than SIP ones. Then it turned into a pissing contest, like you say, it happens in every list with the topic this or that. Again, as I pointed out to Steve above, and after reading all of your responses, our SIP/NAT woes seem obviously ignorance on our part, but that doesn't shadow the fact that IAX2 is working great for us with el-cheapo endpoints like Atcom's AG-188N and I would wish that many more manufacturers supported IAX2. We are happy with IAX and honestly never even had the need/curiosity to deal with the many SIP/NAT problems where sometimes it works great, and other times is a real pain in the ass that takes huge amounts of support to fix, and unhappy customers. On the other hand, IAX took some engineering efforts at first, but the support issues are practically non-existent. -- Alejandro Imass -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On Wed, Feb 29, 2012 at 1:26 PM, Alejandro Imass a...@p2ee.org wrote: On Wed, Feb 29, 2012 at 1:05 PM, Troy Telford ttelford.gro...@gmail.com wrote: On 2012-02-29 15:25:49 +, Alejandro Imass said: We use SIP and IAX interchangeably, but had less hassle with IAX. The topic of the discussion on this thread was that SIP is so awesome and that IAX is a peice of crap. The original question (mine) was that my sound quality when using IAX was bad; with SIP the sound quality was great. Critically, I mentioned that I wanted to use IAX; I even said I was willing to do some self torture to get IAX working properly. Yeah, I wasn't referring particularly to the original post, just the way the thread turned against IAX like if it's not a viable solution and my point all along has been that for *us* IAX2 endpoints have worked better and easier to configure than SIP ones. Then it turned into a pissing contest, like you say, it happens in every list with the topic this or that. Again, as I pointed out to Steve above, and after reading all of your responses, our SIP/NAT woes seem obviously ignorance on our part, but that doesn't shadow the fact that IAX2 is working great for us with el-cheapo endpoints like Atcom's AG-188N and I would wish that many more manufacturers supported IAX2. We are happy with IAX and honestly never even had the need/curiosity to deal with the many SIP/NAT problems where sometimes it works great, and other times is a real pain in the ass that takes huge amounts of support to fix, and unhappy customers. On the other hand, IAX took some engineering efforts at first, but the support issues are practically non-existent. -- Alejandro Imass I always posted that my view was based on experience. My nieces and I made a viable home phone system out of strings and paper cups It is a real pain when you grow so large and then have to switch over to SIP, might as well go with an Industry Standard then code that is and has always been broken since it's inception. You will find IAX2 trunking issues dating back to 2005 and all sorts of IAX2 related problems since I started way before Asterisk 1.0. They have never got it right, SIP either, but at least SIP is compliant enough to work just about all the time unless. Try IAX, the predecessor of IAX2. My alternator is currently not charging my battery enough for nightime driving unless I turn off the radio and A/C. It is fine without the extra variables. This is nothing new. Knowing that when the demand rises, my battery will die and the vehicle will falter and eventually stall means I am going to replace the alternator. Say I need my High Beams or to charge something via cig lighter, I will end up stranded and need to take emergency action. I could buy a used alternator, but I have no past experience with it and have no idea how it will perform. My choice of proper course of action is to put in something that is known by all to work, maybe a bad unit, but backed by an immediate exchange. I will replace the battery and inspect other potential problem areas and eliminate them as well. Now I will have averted any problems down the road by doing it the right way rather than hopping along on something that has been borken since day one. If you are going to do the job, do it right from the start so that you can grow or change with ease and use real recognized standards. If you are just playing around, do whatever. Actually do whatever, and learn the hard way, I don't care, just trying to help. Thanks, Steve Totaro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On my Asterisk system, I'm using a provider that provides both IAX2 and SIP connectivity. Personally, I'd prefer to use IAX2, and that's what my account is setup to use. However, I'm having a problem: With IAX2: - Incoming Voice from my Provider - Asterisk = Sounds great - Outgoing Voice from Asterisk - my Provider = Sounds terrible By terrible, I mean skips, stutters, and distortion. It can be difficult (sometimes impossible) to understand. It doesn't matter what codec I use (at least between G.729, GSM, or ulaw). On the other hand: With SIP: - Incoming Voice from my Provider - Asterisk = Sounds great - Outgoing Voice from Asterisk - my Provider = Sounds great The obvious conclusion is to simply use SIP; however as I've said, I'd prefer to use IAX2 - plus, I'm curious why SIP sounds great, while IAX2 only sounds good one-way (ie. incoming to my asterisk system). The server for my provider is identical in either case. So I figure it's one of a few things: - misconfiguration - My ISP (Comcast) is throttling or giving a low priority to IAX, but not SIP - If there's something I can do here, I'd like to know, but I doubt it. - a problem with my provider - In which I'll contact them. For the first case - misconfiguration, I'd appreciate some input. My iax.conf is fairly straightforward: [general] bandwidth=low jitterbuffer=yes forcejitterbuffer=no encryption = yes autokill=yes maxcallnumbers=12 maxcallnumbers_nonvalidated=4 [guest] type=user context=default callerid=Guest IAX User [myprovider] type=friend username= secret= context=somecontext host=provider_server qualify=1000 disallow=all allow=g729 allow=ulaw auth=md5,rsa requirecalltoken=yes trunk=yes Firewall: Asterisk is behind a connection-tracking firewall; in my case, I've noticed that my own connection to my provider has always been sufficient to allow connection tracking to just work - and incoming calls are accepted without problems, and voice travels in both directions (albeit not so well when outgoing). I have configured my firewall to forward incoming connections on port 4569 to my Asterisk box, and tested. This had no effect on call quality (which is no surprise given it's the /outgoing/ voice that's problematic). Outgoing connections are fairly typical for a NAT setup - anything can go out. Any other ideas before I give up on using IAX? Thanks -- Troy Telford -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
I'd try turning off the jitterbuffer and see if that makes things better. I just traced a similar call quality issue transferring calls incoming DAHDI on one * box to another * box, and turning off the jitterbuffer on the side that couldn't hear (in my case, the * box with the DAHDI lines, as the DAHDI callers couldn't hear the remote callers) fixed the call quality issue. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Troy Telford Sent: Tuesday, February 28, 2012 4:08 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great On my Asterisk system, I'm using a provider that provides both IAX2 and SIP connectivity. Personally, I'd prefer to use IAX2, and that's what my account is setup to use. However, I'm having a problem: With IAX2: - Incoming Voice from my Provider - Asterisk = Sounds great - Outgoing Voice from Asterisk - my Provider = Sounds terrible By terrible, I mean skips, stutters, and distortion. It can be difficult (sometimes impossible) to understand. It doesn't matter what codec I use (at least between G.729, GSM, or ulaw). On the other hand: With SIP: - Incoming Voice from my Provider - Asterisk = Sounds great - Outgoing Voice from Asterisk - my Provider = Sounds great The obvious conclusion is to simply use SIP; however as I've said, I'd prefer to use IAX2 - plus, I'm curious why SIP sounds great, while IAX2 only sounds good one-way (ie. incoming to my asterisk system). The server for my provider is identical in either case. So I figure it's one of a few things: - misconfiguration - My ISP (Comcast) is throttling or giving a low priority to IAX, but not SIP - If there's something I can do here, I'd like to know, but I doubt it. - a problem with my provider - In which I'll contact them. For the first case - misconfiguration, I'd appreciate some input. My iax.conf is fairly straightforward: [general] bandwidth=low jitterbuffer=yes forcejitterbuffer=no encryption = yes autokill=yes maxcallnumbers=12 maxcallnumbers_nonvalidated=4 [guest] type=user context=default callerid=Guest IAX User [myprovider] type=friend username= secret= context=somecontext host=provider_server qualify=1000 disallow=all allow=g729 allow=ulaw auth=md5,rsa requirecalltoken=yes trunk=yes Firewall: Asterisk is behind a connection-tracking firewall; in my case, I've noticed that my own connection to my provider has always been sufficient to allow connection tracking to just work - and incoming calls are accepted without problems, and voice travels in both directions (albeit not so well when outgoing). I have configured my firewall to forward incoming connections on port 4569 to my Asterisk box, and tested. This had no effect on call quality (which is no surprise given it's the /outgoing/ voice that's problematic). Outgoing connections are fairly typical for a NAT setup - anything can go out. Any other ideas before I give up on using IAX? Thanks -- Troy Telford -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The message does not contain any threats AVG for MS Exchange Server (2012.0.1913 - 2114/4837)-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
My first two guesses are that encryption is hosing you or that the single-channel nature of IAX2 may have something to do with it. IAX2 talks on 1 channel, SIP uses twisted pair connotation on two channels (as I understand it). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Troy Telford Sent: Tuesday, February 28, 2012 3:08 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great On my Asterisk system, I'm using a provider that provides both IAX2 and SIP connectivity. Personally, I'd prefer to use IAX2, and that's what my account is setup to use. However, I'm having a problem: With IAX2: - Incoming Voice from my Provider - Asterisk = Sounds great - Outgoing Voice from Asterisk - my Provider = Sounds terrible By terrible, I mean skips, stutters, and distortion. It can be difficult (sometimes impossible) to understand. It doesn't matter what codec I use (at least between G.729, GSM, or ulaw). On the other hand: With SIP: - Incoming Voice from my Provider - Asterisk = Sounds great - Outgoing Voice from Asterisk - my Provider = Sounds great The obvious conclusion is to simply use SIP; however as I've said, I'd prefer to use IAX2 - plus, I'm curious why SIP sounds great, while IAX2 only sounds good one-way (ie. incoming to my asterisk system). The server for my provider is identical in either case. So I figure it's one of a few things: - misconfiguration - My ISP (Comcast) is throttling or giving a low priority to IAX, but not SIP - If there's something I can do here, I'd like to know, but I doubt it. - a problem with my provider - In which I'll contact them. For the first case - misconfiguration, I'd appreciate some input. My iax.conf is fairly straightforward: [general] bandwidth=low jitterbuffer=yes forcejitterbuffer=no encryption = yes autokill=yes maxcallnumbers=12 maxcallnumbers_nonvalidated=4 [guest] type=user context=default callerid=Guest IAX User [myprovider] type=friend username= secret= context=somecontext host=provider_server qualify=1000 disallow=all allow=g729 allow=ulaw auth=md5,rsa requirecalltoken=yes trunk=yes Firewall: Asterisk is behind a connection-tracking firewall; in my case, I've noticed that my own connection to my provider has always been sufficient to allow connection tracking to just work - and incoming calls are accepted without problems, and voice travels in both directions (albeit not so well when outgoing). I have configured my firewall to forward incoming connections on port 4569 to my Asterisk box, and tested. This had no effect on call quality (which is no surprise given it's the /outgoing/ voice that's problematic). Outgoing connections are fairly typical for a NAT setup - anything can go out. Any other ideas before I give up on using IAX? Thanks -- Troy Telford -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On 02/28/2012 03:08 PM, Troy Telford wrote: [myprovider] type=friend username= secret= context=somecontext host=provider_server qualify=1000 disallow=all allow=g729 allow=ulaw auth=md5,rsa requirecalltoken=yes trunk=yes A serious bug with IAX2 trunking in recent versions of Asterisk (you did not mention what version you are using) was just resolved last week. You should test with 'trunk=no' to see if that is the cause of your problem; it seems very likely. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
I've tried turning jitterbuffer off - doesn't make a difference. (And why should it? The Jitterbuffer only applies to incoming calls, doesn't it?) On 2012-02-28 21:12:48 +, Noah Engelberth said: I'd try turning off the jitterbuffer and see if that makes things better. I just traced a similar call quality issue transferring calls incoming DAHDI on one * box to another * box, and turning off the jitterbuffer on the side that couldn't hear (in my case, the * box with the DAHDI lines, as the DAHDI callers couldn't hear the remote callers) fixed the call quality issue. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Troy Telford Sent: Tuesday, February 28, 2012 4:08 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great On my Asterisk system, I'm using a provider that provides both IAX2 and SIP connectivity. Personally, I'd prefer to use IAX2, and that's what my account is setup to use. However, I'm having a problem: With IAX2: - Incoming Voice from my Provider - Asterisk = Sounds great - Outgoing Voice from Asterisk - my Provider = Sounds terrible By terrible, I mean skips, stutters, and distortion. It can be difficult (sometimes impossible) to understand. It doesn't matter what codec I use (at least between G.729, GSM, or ulaw). On the other hand: With SIP: - Incoming Voice from my Provider - Asterisk = Sounds great - Outgoing Voice from Asterisk - my Provider = Sounds great The obvious conclusion is to simply use SIP; however as I've said, I'd prefer to use IAX2 - plus, I'm curious why SIP sounds great, while IAX2 only sounds good one-way (ie. incoming to my asterisk system). The server for my provider is identical in either case. So I figure it's one of a few things: - misconfiguration - My ISP (Comcast) is throttling or giving a low priority to IAX, but not SIP - If there's something I can do here, I'd like to know, but I doubt it. - a problem with my provider - In which I'll contact them. For the first case - misconfiguration, I'd appreciate some input. My iax.conf is fairly straightforward: [general] bandwidth=low jitterbuffer=yes forcejitterbuffer=no encryption = yes autokill=yes maxcallnumbers=12 maxcallnumbers_nonvalidated=4 [guest] type=user context=default callerid=Guest IAX User [myprovider] type=friend usernamesecretcontext=somecontext host=provider_server qualify=1000 disallow=all allow=g729 allow=ulaw auth=md5,rsa requirecalltoken=yes trunk=yes Firewall: Asterisk is behind a connection-tracking firewall; in my case, I've noticed that my own connection to my provider has always been sufficient to allow connection tracking to just work - and incoming calls are accepted without problems, and voice travels in both directions (albeit not so well when outgoing). I have configured my firewall to forward incoming connections on port 4569 to my Asterisk box, and tested. This had no effect on call quality (which is no surprise given it's the /outgoing/ voice that's problematic). Outgoing connections are fairly typical for a NAT setup - anything can go out. Any other ideas before I give up on using IAX? Thanks -- Troy Telford -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The message does not contain any threats AVG for MS Exchange Server (2012.0.1913 - 2114/4837) -- Troy Telford -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
encryption=yes is meaningless if the provider doesn't support it (mine doesn't). I put it there in the wild hope they eventually will - and no config change will be needed on my part. Still, when I changed it to encryption=no, and tested there wasn't any difference. So I've tried disabling the jitterbuffer, and encryption, and there's no effect on call quality - outgoing (from me - provider) sounds bad/distorted, while incoming sounds great. On 2012-02-28 21:14:55 +, Danny Nicholas said: My first two guesses are that encryption is hosing you or that the single-channel nature of IAX2 may have something to do with it. IAX2 talks on 1 channel, SIP uses twisted pair connotation on two channels (as I understand it). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Troy Telford Sent: Tuesday, February 28, 2012 3:08 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great On my Asterisk system, I'm using a provider that provides both IAX2 and SIP connectivity. Personally, I'd prefer to use IAX2, and that's what my account is setup to use. However, I'm having a problem: With IAX2: - Incoming Voice from my Provider - Asterisk = Sounds great - Outgoing Voice from Asterisk - my Provider = Sounds terrible By terrible, I mean skips, stutters, and distortion. It can be difficult (sometimes impossible) to understand. It doesn't matter what codec I use (at least between G.729, GSM, or ulaw). On the other hand: With SIP: - Incoming Voice from my Provider - Asterisk = Sounds great - Outgoing Voice from Asterisk - my Provider = Sounds great The obvious conclusion is to simply use SIP; however as I've said, I'd prefer to use IAX2 - plus, I'm curious why SIP sounds great, while IAX2 only sounds good one-way (ie. incoming to my asterisk system). The server for my provider is identical in either case. So I figure it's one of a few things: - misconfiguration - My ISP (Comcast) is throttling or giving a low priority to IAX, but not SIP - If there's something I can do here, I'd like to know, but I doubt it. - a problem with my provider - In which I'll contact them. For the first case - misconfiguration, I'd appreciate some input. My iax.conf is fairly straightforward: [general] bandwidth=low jitterbuffer=yes forcejitterbuffer=no encryption = yes autokill=yes maxcallnumbers=12 maxcallnumbers_nonvalidated=4 [guest] type=user context=default callerid=Guest IAX User [myprovider] type=friend username= secret= context=somecontext host=provider_server qualify=1000 disallow=all allow=g729 allow=ulaw auth=md5,rsa requirecalltoken=yes trunk=yes Firewall: Asterisk is behind a connection-tracking firewall; in my case, I've noticed that my own connection to my provider has always been sufficient to allow connection tracking to just work - and incoming calls are accepted without problems, and voice travels in both directions (albeit not so well when outgoing). I have configured my firewall to forward incoming connections on port 4569 to my Asterisk box, and tested. This had no effect on call quality (which is no surprise given it's the /outgoing/ voice that's problematic). Outgoing connections are fairly typical for a NAT setup - anything can go out. Any other ideas before I give up on using IAX? Thanks -- Troy Telford -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Troy Telford -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On 2012-02-28 22:17:37 +, Danny Nicholas said: Ok Steve, obviously you've outsmarted at least this poster. On the one hand, IAX2 has purchased things for you (won't go as far as saying it bought your Mercedes), but on the other hand it is being dropped by providers as we speak. So are you saying it can be a good thing if you have the time and skill level to pursue it, but beginners should leave it alone? I understood Steve to mean the following: - Secure locations like IAX. There's only one port to monitor or allow through a firewall, which is pretty compelling. - Aforementioned locations can't get IAX to work well. - So they hire Steve to get IAX to work properly, and he makes money. At least, that's my take. -- Troy Telford -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
People around here either hate me or love me. I post experience and am accused of bragging or whatever. As a reader, I want to know who is giving me advice and what it is based on. $40k/wk of long distance through VoicePulse. I have the invoices, that is high usage, others attack me for posting information like this, I think I know why but I don't care. You have to have thick skin on these lists, the more technical, the more you better have done your homework or get flamed. It is from years of experience, not outsmarting anyone. It took me months to figure out that it just doesn't work well and as you can see, all of the posts are very dated. Nobody outsmarted anyone, just pure experience and experience of MANY other people that use Asterisk. Many did not wish to make waves and emailed me directly that they either came to the same conclusion or that they switched due to my suggesting and had no more problems. Digium and Digium FanBoys will argue that IAX2 is the best thing since sliced bread. Digium will ALWAYS tow the party line. It was either Flemming or Lesher that actually posted that it was in an official release so it couldn't have bugs. That was the end of listening to Digium about IAX2. That statement was archived with my reply of how ridiculous the statement was. It is all on the mailing list. The compensation thing is very true, people drink the cool-aide about IAX2 and it sounds great. Then it turns out that they go to production, and audio sucks, customers are complaining. It becomes a huge problem obviously to an ITSP or any call center. As I said, my experience is dated, but I have been one of the most prolific people in the Asterisk community, I spoke at Astricon in 2007 on Large Call Center Track and was the #1 poster for the year, a year or two ago. I predate most of Digium Staff. I do this stuff in the real world, over VSAT or whatever connectivity you can think of, my experience is real, not a developer in the world of code. To answer your question, maybe you can spend time and get it to work correctly, I have no idea, but why? Why not just use SIP and be done with it. Also realize that the dated posts have replies that are ridiculous like VoicePulse is probably laying people off right now as we speak. If a challenge drives you and you have tons of time to possibly never figure it out and go to SIP, then by all means, do it. If you want it to just work, use OpenVPN to get your single port, don't believe the Digium party line and replies about using OpenSER or whatever it is called now. I get past the firewall and NAT issues with OpenVPN. My standard now is Vyatta with NTOP, Asterisk, Webmin installed. I only use SIP and use OpenVPN. I build Asterisk from source and menuconfig, I remove all that is not needed, including IAX2. I do download all the sound files in different languages and codecs. It just works. I like things that just work. Thanks, Steve Totaro On Tue, Feb 28, 2012 at 5:17 PM, Danny Nicholas da...@debsinc.com wrote: Ok Steve, obviously you’ve outsmarted at least this poster. On the one hand, IAX2 has purchased things for you (won’t go as far as saying it bought your Mercedes), but on the other hand it is being dropped by providers as we speak. So are you saying it can be a good thing if you have the time and skill level to pursue it, but beginners should leave it alone? ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Steve Totaro *Sent:* Tuesday, February 28, 2012 3:59 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great ** ** PSS ** ** http://bit.ly/ywiwzt On Tue, Feb 28, 2012 at 4:56 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: Google or click this link http://bit.ly/ywiwzteve Steve Totaro IAX and then stop wasting your time, go with SIP even if you need to create VPN tunnel(s). ** ** Forget IAX2 and save yourself time you will never get back. ** ** IAX2 has put tens of thousands of dollars in my pockets from the DoD, DoS, prime contractors to ITSPs around the world. ** ** Thanks for IAX2 Digium! ** ** Thanks, Steve Totaro ** ** On Tue, Feb 28, 2012 at 4:30 PM, Troy Telford ttelford.gro...@gmail.com wrote: I've tried turning jitterbuffer off - doesn't make a difference. (And why should it? The Jitterbuffer only applies to incoming calls, doesn't it?)** ** On 2012-02-28 21:12:48 +, Noah Engelberth said: I'd try turning off the jitterbuffer and see if that makes things better. I just traced a similar call quality issue transferring calls incoming DAHDI on one * box to another * box, and turning off the jitterbuffer on the side that couldn't hear (in my case, the * box with the DAHDI lines, as the DAHDI callers couldn't hear the remote
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
I have no interest in the penis-measurement competition firing up here, but I'll say that we have 100% abandoned IAX from all of our systems due to a myriad of issues. These days it offers no real advantages in our opinion. On Tue, Feb 28, 2012 at 4:03 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: People around here either hate me or love me. I post experience and am accused of bragging or whatever. As a reader, I want to know who is giving me advice and what it is based on. $40k/wk of long distance through VoicePulse. I have the invoices, that is high usage, others attack me for posting information like this, I think I know why but I don't care. You have to have thick skin on these lists, the more technical, the more you better have done your homework or get flamed. It is from years of experience, not outsmarting anyone. It took me months to figure out that it just doesn't work well and as you can see, all of the posts are very dated. Nobody outsmarted anyone, just pure experience and experience of MANY other people that use Asterisk. Many did not wish to make waves and emailed me directly that they either came to the same conclusion or that they switched due to my suggesting and had no more problems. Digium and Digium FanBoys will argue that IAX2 is the best thing since sliced bread. Digium will ALWAYS tow the party line. It was either Flemming or Lesher that actually posted that it was in an official release so it couldn't have bugs. That was the end of listening to Digium about IAX2. That statement was archived with my reply of how ridiculous the statement was. It is all on the mailing list. The compensation thing is very true, people drink the cool-aide about IAX2 and it sounds great. Then it turns out that they go to production, and audio sucks, customers are complaining. It becomes a huge problem obviously to an ITSP or any call center. As I said, my experience is dated, but I have been one of the most prolific people in the Asterisk community, I spoke at Astricon in 2007 on Large Call Center Track and was the #1 poster for the year, a year or two ago. I predate most of Digium Staff. I do this stuff in the real world, over VSAT or whatever connectivity you can think of, my experience is real, not a developer in the world of code. To answer your question, maybe you can spend time and get it to work correctly, I have no idea, but why? Why not just use SIP and be done with it. Also realize that the dated posts have replies that are ridiculous like VoicePulse is probably laying people off right now as we speak. If a challenge drives you and you have tons of time to possibly never figure it out and go to SIP, then by all means, do it. If you want it to just work, use OpenVPN to get your single port, don't believe the Digium party line and replies about using OpenSER or whatever it is called now. I get past the firewall and NAT issues with OpenVPN. My standard now is Vyatta with NTOP, Asterisk, Webmin installed. I only use SIP and use OpenVPN. I build Asterisk from source and menuconfig, I remove all that is not needed, including IAX2. I do download all the sound files in different languages and codecs. It just works. I like things that just work. Thanks, Steve Totaro On Tue, Feb 28, 2012 at 5:17 PM, Danny Nicholas da...@debsinc.com wrote: Ok Steve, obviously you’ve outsmarted at least this poster. On the one hand, IAX2 has purchased things for you (won’t go as far as saying it bought your Mercedes), but on the other hand it is being dropped by providers as we speak. So are you saying it can be a good thing if you have the time and skill level to pursue it, but beginners should leave it alone? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro Sent: Tuesday, February 28, 2012 3:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great PSS http://bit.ly/ywiwzt On Tue, Feb 28, 2012 at 4:56 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: Google or click this link http://bit.ly/ywiwzteve Steve Totaro IAX and then stop wasting your time, go with SIP even if you need to create VPN tunnel(s). Forget IAX2 and save yourself time you will never get back. IAX2 has put tens of thousands of dollars in my pockets from the DoD, DoS, prime contractors to ITSPs around the world. Thanks for IAX2 Digium! Thanks, Steve Totaro On Tue, Feb 28, 2012 at 4:30 PM, Troy Telford ttelford.gro...@gmail.com wrote: I've tried turning jitterbuffer off - doesn't make a difference. (And why should it? The Jitterbuffer only applies to incoming calls, doesn't it?) On 2012-02-28 21:12:48 +, Noah Engelberth said: I'd try turning off the jitterbuffer and see if that makes things better. I just traced a similar call
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
Roger That, I am an IC. I contract with the Government to little ten phone shops. From VA/MD/DC area, I have been contracted and flown in to many large call center locations that were CONUS and OCONUS. My facebook is Steve Totaro in Reston VA. LinkedIN is more accurate, but my resume speaks the truth. Thanks, Steve Totaro On Tue, Feb 28, 2012 at 5:44 PM, Troy Telford ttelford.gro...@gmail.comwrote: On 2012-02-28 22:17:37 +, Danny Nicholas said: Ok Steve, obviously you've outsmarted at least this poster. On the one hand, IAX2 has purchased things for you (won't go as far as saying it bought your Mercedes), but on the other hand it is being dropped by providers as we speak. So are you saying it can be a good thing if you have the time and skill level to pursue it, but beginners should leave it alone? I understood Steve to mean the following: - Secure locations like IAX. There's only one port to monitor or allow through a firewall, which is pretty compelling. - Aforementioned locations can't get IAX to work well. - So they hire Steve to get IAX to work properly, and he makes money. At least, that's my take. -- Troy Telford -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
And the dude arrives talking about penis.. On Tue, Feb 28, 2012 at 6:07 PM, Carlos Alvarez car...@televolve.comwrote: I have no interest in the penis-measurement competition firing up here, but I'll say that we have 100% abandoned IAX from all of our systems due to a myriad of issues. These days it offers no real advantages in our opinion. On Tue, Feb 28, 2012 at 4:03 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: People around here either hate me or love me. I post experience and am accused of bragging or whatever. As a reader, I want to know who is giving me advice and what it is based on. $40k/wk of long distance through VoicePulse. I have the invoices, that is high usage, others attack me for posting information like this, I think I know why but I don't care. You have to have thick skin on these lists, the more technical, the more you better have done your homework or get flamed. It is from years of experience, not outsmarting anyone. It took me months to figure out that it just doesn't work well and as you can see, all of the posts are very dated. Nobody outsmarted anyone, just pure experience and experience of MANY other people that use Asterisk. Many did not wish to make waves and emailed me directly that they either came to the same conclusion or that they switched due to my suggesting and had no more problems. Digium and Digium FanBoys will argue that IAX2 is the best thing since sliced bread. Digium will ALWAYS tow the party line. It was either Flemming or Lesher that actually posted that it was in an official release so it couldn't have bugs. That was the end of listening to Digium about IAX2. That statement was archived with my reply of how ridiculous the statement was. It is all on the mailing list. The compensation thing is very true, people drink the cool-aide about IAX2 and it sounds great. Then it turns out that they go to production, and audio sucks, customers are complaining. It becomes a huge problem obviously to an ITSP or any call center. As I said, my experience is dated, but I have been one of the most prolific people in the Asterisk community, I spoke at Astricon in 2007 on Large Call Center Track and was the #1 poster for the year, a year or two ago. I predate most of Digium Staff. I do this stuff in the real world, over VSAT or whatever connectivity you can think of, my experience is real, not a developer in the world of code. To answer your question, maybe you can spend time and get it to work correctly, I have no idea, but why? Why not just use SIP and be done with it. Also realize that the dated posts have replies that are ridiculous like VoicePulse is probably laying people off right now as we speak. If a challenge drives you and you have tons of time to possibly never figure it out and go to SIP, then by all means, do it. If you want it to just work, use OpenVPN to get your single port, don't believe the Digium party line and replies about using OpenSER or whatever it is called now. I get past the firewall and NAT issues with OpenVPN. My standard now is Vyatta with NTOP, Asterisk, Webmin installed. I only use SIP and use OpenVPN. I build Asterisk from source and menuconfig, I remove all that is not needed, including IAX2. I do download all the sound files in different languages and codecs. It just works. I like things that just work. Thanks, Steve Totaro On Tue, Feb 28, 2012 at 5:17 PM, Danny Nicholas da...@debsinc.com wrote: Ok Steve, obviously you’ve outsmarted at least this poster. On the one hand, IAX2 has purchased things for you (won’t go as far as saying it bought your Mercedes), but on the other hand it is being dropped by providers as we speak. So are you saying it can be a good thing if you have the time and skill level to pursue it, but beginners should leave it alone? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro Sent: Tuesday, February 28, 2012 3:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great PSS http://bit.ly/ywiwzt On Tue, Feb 28, 2012 at 4:56 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: Google or click this link http://bit.ly/ywiwzteve Steve Totaro IAX and then stop wasting your time, go with SIP even if you need to create VPN tunnel(s). Forget IAX2 and save yourself time you will never get back. IAX2 has put tens of thousands of dollars in my pockets from the DoD, DoS, prime contractors to ITSPs around the world. Thanks for IAX2 Digium! Thanks, Steve Totaro On Tue, Feb 28, 2012 at 4:30 PM, Troy Telford ttelford.gro...@gmail.com wrote: I've tried turning jitterbuffer
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On 2012-02-28 21:22:44 +, Kevin P. Fleming said: A serious bug with IAX2 trunking in recent versions of Asterisk (you did not mention what version you are using) was just resolved last week. You should test with 'trunk=no' to see if that is the cause of your problem; it seems very likely. For the record: 1.8.8.2~dfsg-1 (via Debian packages). I've tried trunk=no, and it might have made a difference (I'll have a better idea after some more testing.) -- Troy Telford -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
They said the same thing in 2005, 2008, now Every release. You never answered the question as to why you don't want to use SIP. Is there a reason, or do you just want to torture yourself? Thanks, Steve T On Tue, Feb 28, 2012 at 6:23 PM, Troy Telford ttelford.gro...@gmail.comwrote: On 2012-02-28 21:22:44 +, Kevin P. Fleming said: A serious bug with IAX2 trunking in recent versions of Asterisk (you did not mention what version you are using) was just resolved last week. You should test with 'trunk=no' to see if that is the cause of your problem; it seems very likely. For the record: 1.8.8.2~dfsg-1 (via Debian packages). I've tried trunk=no, and it might have made a difference (I'll have a better idea after some more testing.) -- Troy Telford -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
BTW, Trunking was the other selling point of IAX2 besides using 1 port which is easily a DDOS target and also probably still an implantation problem of using one thread and one proc for all calls. Trunking allowed for less overhead then SIP since all the overhead for the concurrent calls were combined into one stream. Without trunking, you only have the single port thing. It is quite easy to open the correct ports for SIP, some just have GUI with a SIP checkbox, IPTables is simple and there are tons of howtos. Thanks, Steve T On Tue, Feb 28, 2012 at 6:29 PM, Steve Totaro stot...@asteriskhelpdesk.comwrote: They said the same thing in 2005, 2008, now Every release. You never answered the question as to why you don't want to use SIP. Is there a reason, or do you just want to torture yourself? Thanks, Steve T On Tue, Feb 28, 2012 at 6:23 PM, Troy Telford ttelford.gro...@gmail.comwrote: On 2012-02-28 21:22:44 +, Kevin P. Fleming said: A serious bug with IAX2 trunking in recent versions of Asterisk (you did not mention what version you are using) was just resolved last week. You should test with 'trunk=no' to see if that is the cause of your problem; it seems very likely. For the record: 1.8.8.2~dfsg-1 (via Debian packages). I've tried trunk=no, and it might have made a difference (I'll have a better idea after some more testing.) -- Troy Telford -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
IAX is not supported or taken seriously outside the Asterisk ghetto, and that's good enough reason not to use it, IMHO. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
Hey Alex, Hope you are well. Just a piece of advice. Many or most people do not know the real definition of ghetto and take it as a negative, poor, racial, black, connotation. Your vocabulary and and ability to articulate correctly can get you in trouble sometimes. Anyone that thinks that the word Ghetto means anything above, or a racial slur should look up the true definition. It isn't even an insult to the Asterisk Community. By definition, the Asterisk Community is an online Ghetto. Just wanted to clear that up before someone tries to label you. Thanks, Steve T On Tue, Feb 28, 2012 at 6:37 PM, Alex Balashov abalas...@evaristesys.comwrote: IAX is not supported or taken seriously outside the Asterisk ghetto, and that's good enough reason not to use it, IMHO. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
Wow Wikipedia was the only place that had the original meaning and not the slur or slang meaning. A *ghetto* is a section of a city predominantly occupied by a group who live there, especially because of social, economic, or legal issues. The term was originally used in Venice http://en.wikipedia.org/wiki/Venice to describe the area where Jews http://en.wikipedia.org/wiki/Jews were compelled to live. On Tue, Feb 28, 2012 at 6:55 PM, Steve Totaro stot...@asteriskhelpdesk.comwrote: Hey Alex, Hope you are well. Just a piece of advice. Many or most people do not know the real definition of ghetto and take it as a negative, poor, racial, black, connotation. Your vocabulary and and ability to articulate correctly can get you in trouble sometimes. Anyone that thinks that the word Ghetto means anything above, or a racial slur should look up the true definition. It isn't even an insult to the Asterisk Community. By definition, the Asterisk Community is an online Ghetto. Just wanted to clear that up before someone tries to label you. Thanks, Steve T On Tue, Feb 28, 2012 at 6:37 PM, Alex Balashov abalas...@evaristesys.comwrote: IAX is not supported or taken seriously outside the Asterisk ghetto, and that's good enough reason not to use it, IMHO. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On Tue, Feb 28, 2012 at 6:36 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: [...] Without trunking, you only have the single port thing. It is quite easy to Nope. The main reason _we_ use IAX is because it's easier for NAT open the correct ports for SIP, some just have GUI with a SIP checkbox, It may be true for you but it's certainly not the truth. - SIP requires redirection of ports if behind a NAT which is about 99% of home users, whether behind a WiFi router or an ISP private network. - SIP requires far more set-up and support effort and it's not a valid choice for a simple to use home-phone. (a) ISP routers change IPs frequently, (b) the router may change the ATA's private IP rendering the port redirection broken. - A public SIP (w/o a VPN) requires careful control (e.g. contactpermit in Asterisk) to limit the IPs that can connect to the public box. Else you will get serous harm from things like SIPVicious attacks. ISP change their IPs frequently so maintaining your user/ip list is almost impossible. IAX2 was very vulnerable as well up to 2009 but many things in this regard have changed and are much better. Granted, these security issues are common for both SIP and IAX2 but IMHO it's easier to manage with IAX. - In a NAT scenario SIP requires a couple of redirected ports per extension, which is a no-go for SMB installations requiring several ATAs without going to the extent of installing a more powerful equipment than a simple ATA. - You may use OpenVPN with SIP as you said but requires a PC which is not an option for a simple VoIP business that delivers something like Vonage, just plug it and it works. AFAIK there is no port redirection or any special configuration to use Vonage and it works almost on any network set-up (I don't use Vonage but know people that do). So if something like Vonage is using SIP it's probably using a VPN software like you recommend. Anyway, the point is that SIP and IAX2 have both pros and cons and I don't consider IAX2 to be a broken bat like you state. On the contrary, I think it works pretty well, and we use both SIP and IAX2 targeted to simple Home, SOHO and SMBs that just want to plug it and work. We get that with IAX2 and not with SIP so from our experience is completely the opposite of what you say. -- Alejandro Imass IAX2 is supported on cheap ATAs by several chineese companies and they work quite well. IPTables is simple and there are tons of howtos. Thanks, Steve T On Tue, Feb 28, 2012 at 6:29 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: They said the same thing in 2005, 2008, now Every release. You never answered the question as to why you don't want to use SIP. Is there a reason, or do you just want to torture yourself? Thanks, Steve T On Tue, Feb 28, 2012 at 6:23 PM, Troy Telford ttelford.gro...@gmail.com wrote: On 2012-02-28 21:22:44 +, Kevin P. Fleming said: A serious bug with IAX2 trunking in recent versions of Asterisk (you did not mention what version you are using) was just resolved last week. You should test with 'trunk=no' to see if that is the cause of your problem; it seems very likely. For the record: 1.8.8.2~dfsg-1 (via Debian packages). I've tried trunk=no, and it might have made a difference (I'll have a better idea after some more testing.) -- Troy Telford -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
Oops, I meant da Asterisk 'hood. Thanks for the protip. On 02/28/2012 06:55 PM, Steve Totaro wrote: Hey Alex, Hope you are well. Just a piece of advice. Many or most people do not know the real definition of ghetto and take it as a negative, poor, racial, black, connotation. Your vocabulary and and ability to articulate correctly can get you in trouble sometimes. Anyone that thinks that the word Ghetto means anything above, or a racial slur should look up the true definition. It isn't even an insult to the Asterisk Community. By definition, the Asterisk Community is an online Ghetto. Just wanted to clear that up before someone tries to label you. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On 2012-02-28 23:29:53 +, Steve Totaro said: They said the same thing in 2005, 2008, now Every release. You never answered the question as to why you don't want to use SIP. Is there a reason, or do you just want to torture yourself? Probably self-torture, yes. I want to at least try to use IAX2 because I can - learn more about Asterisk for the experience, etc. Now that I've found problems, I want to know if it's a problem with my configuration, or if it was my ISP, or perhaps my provider. I have no problem at all with SIP. It seems to be the direction everybody is going - including Digium: From the Asterisk 10 Codecs and Audio Formats page: Note that the additional codecs discussed here are available for use in Asterisk's SIP channel driver, only. Asterisk 10 does not make them available for IAX2, MGCP, SSCP, H.323, UniSTIM, etc. Digium's fax driver doesn't work with IAX2... even in ulaw passthrough mode. So if I can find that yes, it's so much a problem with my configuration but a bug in the software, then I'll be satisfied switch to what works (ie. SIP). -- Troy Telford -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
Perhaps your users live in an internet ghetto where the routers are similar to Yugos with spinners. We haven't run into any routers that don't do NAT properly in a very very long time. On Tue, Feb 28, 2012 at 5:07 PM, Alejandro Imass a...@p2ee.org wrote: On Tue, Feb 28, 2012 at 6:36 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: [...] Without trunking, you only have the single port thing. It is quite easy to Nope. The main reason _we_ use IAX is because it's easier for NAT open the correct ports for SIP, some just have GUI with a SIP checkbox, It may be true for you but it's certainly not the truth. - SIP requires redirection of ports if behind a NAT which is about 99% of home users, whether behind a WiFi router or an ISP private network. - SIP requires far more set-up and support effort and it's not a valid choice for a simple to use home-phone. (a) ISP routers change IPs frequently, (b) the router may change the ATA's private IP rendering the port redirection broken. - A public SIP (w/o a VPN) requires careful control (e.g. contactpermit in Asterisk) to limit the IPs that can connect to the public box. Else you will get serous harm from things like SIPVicious attacks. ISP change their IPs frequently so maintaining your user/ip list is almost impossible. IAX2 was very vulnerable as well up to 2009 but many things in this regard have changed and are much better. Granted, these security issues are common for both SIP and IAX2 but IMHO it's easier to manage with IAX. - In a NAT scenario SIP requires a couple of redirected ports per extension, which is a no-go for SMB installations requiring several ATAs without going to the extent of installing a more powerful equipment than a simple ATA. - You may use OpenVPN with SIP as you said but requires a PC which is not an option for a simple VoIP business that delivers something like Vonage, just plug it and it works. AFAIK there is no port redirection or any special configuration to use Vonage and it works almost on any network set-up (I don't use Vonage but know people that do). So if something like Vonage is using SIP it's probably using a VPN software like you recommend. Anyway, the point is that SIP and IAX2 have both pros and cons and I don't consider IAX2 to be a broken bat like you state. On the contrary, I think it works pretty well, and we use both SIP and IAX2 targeted to simple Home, SOHO and SMBs that just want to plug it and work. We get that with IAX2 and not with SIP so from our experience is completely the opposite of what you say. -- Alejandro Imass IAX2 is supported on cheap ATAs by several chineese companies and they work quite well. IPTables is simple and there are tons of howtos. Thanks, Steve T On Tue, Feb 28, 2012 at 6:29 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: They said the same thing in 2005, 2008, now Every release. You never answered the question as to why you don't want to use SIP. Is there a reason, or do you just want to torture yourself? Thanks, Steve T On Tue, Feb 28, 2012 at 6:23 PM, Troy Telford ttelford.gro...@gmail.com wrote: On 2012-02-28 21:22:44 +, Kevin P. Fleming said: A serious bug with IAX2 trunking in recent versions of Asterisk (you did not mention what version you are using) was just resolved last week. You should test with 'trunk=no' to see if that is the cause of your problem; it seems very likely. For the record: 1.8.8.2~dfsg-1 (via Debian packages). I've tried trunk=no, and it might have made a difference (I'll have a better idea after some more testing.) -- Troy Telford -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On Tue, Feb 28, 2012 at 7:19 PM, Carlos Alvarez car...@televolve.com wrote: Perhaps your users live in an internet ghetto where the routers are similar to Yugos with spinners. We haven't run into any routers that don't do NAT properly in a very very long time. Perhaps you should read again and point out where I state that is a router/NAT problem. I said that the configuration of routers and redirecting ports is a pain in the ass for users and creates a lot of support problems that simply don; t exist with IAX2. With a IAX2 ATA you just plug it and works. This cannot be done with SIP and off the shelf cheap ATAs, period. Also, respect netiquette and don't top post and use derogatory remarks and keep your discussion technical. -- Alejandro Imass On Tue, Feb 28, 2012 at 5:07 PM, Alejandro Imass a...@p2ee.org wrote: On Tue, Feb 28, 2012 at 6:36 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: [...] Without trunking, you only have the single port thing. It is quite easy to Nope. The main reason _we_ use IAX is because it's easier for NAT -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On Tue, Feb 28, 2012 at 5:35 PM, Alejandro Imass a...@p2ee.org wrote: works. This cannot be done with SIP and off the shelf cheap ATAs, period. We do it, so cannot seems to be a strong word. It's not perfect, but our IAX problems outnumbered the SIP problems by at least double. Your mileage clearly varies. Also, respect netiquette and don't top post and use derogatory remarks and keep your discussion technical. Unfortunately, top-posting has become normal on this list. I'm just fitting into how others were quoting the conversation. If you find my remarks derogatory, I don't particularly care. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On Tue, Feb 28, 2012 at 7:41 PM, Carlos Alvarez car...@televolve.com wrote: On Tue, Feb 28, 2012 at 5:35 PM, Alejandro Imass a...@p2ee.org wrote: works. This cannot be done with SIP and off the shelf cheap ATAs, period. We do it, so cannot seems to be a strong word. It's not perfect, Please expand as to how you set-up a SIP ATA behind a common home router set-up, without port redirection and/or use of a SIP proxy and/or STUN server? Unless the ATA has some sort of magic (e.g. VPN support) it _cannot_ be done. but our IAX problems outnumbered the SIP problems by at least double. Your mileage clearly varies. [...] Unfortunately, top-posting has become normal on this list. I'm just fitting into how others were quoting the conversation. If you find my Top posting seems to be more popular due to use broken smart phone MUAs that can't reply in-line. But if you have the means it should be avoided for future reference and direct people to read the archives and find useful information. remarks derogatory, I don't particularly care. You should care. Words like ghetto, your users, and using name brands like Yugo in a pejorative way are all derogatory and may direct the discussions on a personal level which should always be avoided. I don't drive a Yugo but if I did I could easily be offended by the pejorative use of the brand. We are all here to share our knowledge and our valuable time so to make it worthwhile we should all care about conserving netiquette. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On Tue, Feb 28, 2012 at 6:14 PM, Alejandro Imass a...@p2ee.org wrote: Please expand as to how you set-up a SIP ATA behind a common home router set-up, without port redirection and/or use of a SIP proxy and/or STUN server? Unless the ATA has some sort of magic (e.g. VPN support) it _cannot_ be done. Go buy a WRT-54G or nearly any consumer-class router and just plug in a SIP device. Done. It works. We *never* work on customer routers and very rarely have to tell them to reconfigure their router at all. My own home configuration is an Airport Extreme with zero configuration. So either these are very old routers you're having a problem with, or buggy SIP devices, or something else. You should care. Hmm, let me check the reading... http://i1-win.softpedia-static.com/screenshots/Care-Meter_1.png don't drive a Yugo but if I did I could easily be offended by the pejorative use of the brand. It's a piece of junk and everyone knows it, including the owners, so who cares? -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On Tue, Feb 28, 2012 at 7:07 PM, Alejandro Imass a...@p2ee.org wrote: On Tue, Feb 28, 2012 at 6:36 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: [...] Without trunking, you only have the single port thing. It is quite easy to Nope. The main reason _we_ use IAX is because it's easier for NAT And it is easier for NAT because it uses one port as I stated, next open the correct ports for SIP, some just have GUI with a SIP checkbox, It may be true for you but it's certainly not the truth. - SIP requires redirection of ports if behind a NAT which is about 99% of home users, whether behind a WiFi router or an ISP private network. Um, not when the server is on a public IP and the phones are configured correctly. - SIP requires far more set-up and support effort and it's not a valid choice for a simple to use home-phone. (a) ISP routers change IPs frequently, (b) the router may change the ATA's private IP rendering the port redirection broken. What about Magic Jack or Vonage? The phone registers regularly with the server so that negates everything above. I don't do simple home setups, but they are simple home setups, your words, not mine. I have only had to redirect ports if the server is behind a NAT. Get a SNOM 370, flash with OpenVPN, run as a client and no problems, not that there would be anyway. I have placed 20 business phones behind NAT with no special configuration and no issues but a bad phone or two in two years I have hostage negotiators with OpenVPN and a softphone on their laptops, they travel the world and never have problems except maybe bandwidth. - A public SIP (w/o a VPN) requires careful control (e.g. contactpermit in Asterisk) to limit the IPs that can connect to the public box. Else you will get serous harm from things like SIPVicious attacks. This can easily be mitigated by running on nonstandard ports. Fail2Ban, and a ton of other products can help, but yes, you are correct. A competent Admin is required to check logs daily and configure things correctly. ISP change their IPs frequently so maintaining your user/ip list is almost impossible. I use IP=dynamic with no problems but people tying to guess a password that is the extension and MAC of the phone. Dictionary attack is nothing. With a Gig pipe and fail2ban, no problems. Also, I don't know where you live but I got Comcast@home when it first came out and my IP has never changed. ISPs in this area say dynamic but they are static, at least the big two, Verizon and Comcast for home use. IAX2 was very vulnerable as well up to 2009 but many things in this regard have changed and are much better. Granted, these security issues are common for both SIP and IAX2 but IMHO it's easier to manage with IAX. Security was never really the issue if you read the thread. It is about voice quality. - In a NAT scenario SIP requires a couple of redirected ports per extension, which is a no-go for SMB installations requiring several ATAs without going to the extent of installing a more powerful equipment than a simple ATA. Not in my experience, phone registers with server on public IP, no problems except some obscure setting on a firewall. Easy enough to google away. - You may use OpenVPN with SIP as you said but requires a PC which is not an option for a simple VoIP business that delivers something like Vonage, just plug it and it works. Wrong, the SNOM 370 works great with OpenVPN. You just contradicted yourself as far as plug and play. The SNOM 370 can also act as a bridge over the VPN tunnel using the LAN port so the whole office is behind either split tunnel or direct VPN. Any other SIP phone behind the SNOM with VPN bridging will also be on the VPN as well as workstations. AFAIK there is no port redirection or any special configuration to use Vonage and it works almost on any network set-up (I don't use Vonage but know people that do). So if something like Vonage is using SIP it's probably using a VPN software like you recommend. Magic Jack is pure SIP, no VPN Anyway, the point is that SIP and IAX2 have both pros and cons and I don't consider IAX2 to be a broken bat like you state. On the contrary, I think it works pretty well, and we use both SIP and IAX2 targeted to simple Home, SOHO and SMBs that just want to plug it and work. We get that with IAX2 and not with SIP so from our experience is completely the opposite of what you say. That is fine, I added disclaimers and small shops. I deal in the 15,000 calls a day minimum realm, so we live in different worlds. Two cups and and a string work too -- Alejandro Imass IAX2 is supported on cheap ATAs by several chineese companies and they work quite well. IPTables is simple and there are tons of howtos. Thanks, Steve T On Tue, Feb 28, 2012 at 6:29 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: They said the same thing in 2005, 2008, now
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On Tue, Feb 28, 2012 at 7:41 PM, Carlos Alvarez car...@televolve.comwrote: On Tue, Feb 28, 2012 at 5:35 PM, Alejandro Imass a...@p2ee.org wrote: works. This cannot be done with SIP and off the shelf cheap ATAs, period. We do it, so cannot seems to be a strong word. It's not perfect, but our IAX problems outnumbered the SIP problems by at least double. Your mileage clearly varies. Yes, I have had no problems with Grandstream first gen ATAs, configured with server and credentials and shipped off, they just work. Also, respect netiquette and don't top post and use derogatory remarks and keep your discussion technical. Unfortunately, top-posting has become normal on this list. I'm just fitting into how others were quoting the conversation. If you find my remarks derogatory, I don't particularly care. I follow the direction of the conversation. If people are top posting, then I follow suit, bottom, then I bottom post, inline as we are, then that is what I do. When in Rome -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
Roger That! On Tue, Feb 28, 2012 at 8:28 PM, Carlos Alvarez car...@televolve.comwrote: On Tue, Feb 28, 2012 at 6:14 PM, Alejandro Imass a...@p2ee.org wrote: Please expand as to how you set-up a SIP ATA behind a common home router set-up, without port redirection and/or use of a SIP proxy and/or STUN server? Unless the ATA has some sort of magic (e.g. VPN support) it _cannot_ be done. Go buy a WRT-54G or nearly any consumer-class router and just plug in a SIP device. Done. It works. We *never* work on customer routers and very rarely have to tell them to reconfigure their router at all. My own home configuration is an Airport Extreme with zero configuration. So either these are very old routers you're having a problem with, or buggy SIP devices, or something else. You should care. Hmm, let me check the reading... http://i1-win.softpedia-static.com/screenshots/Care-Meter_1.png don't drive a Yugo but if I did I could easily be offended by the pejorative use of the brand. It's a piece of junk and everyone knows it, including the owners, so who cares? -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
Just to stir the pot a bit, I am a member of a worldwide private network of Asterisk and AstLinux users. the network uses IAX exclusively, and we have no issues relating to audio quality with a large variety of providers, routers, host machines, and expertise in configuration of the specific nodes Many ( in the US and Canada ) use a PSTN connection as well as the private network using voip.ms with equally stellar quality using IAX IAX was chosen as the default network protocol because of the many issues with SIP, routers, and ( later ) the many attempts at break-ins. As an aside, didn't the manufacture of the Yugo die with the death of Yugoslavia? Most of the Yugo's shortly thereafter? If anyone has one now it may be close to the value of a Delorian No replies necessary or even desired. John Novack Carlos Alvarez wrote: On Tue, Feb 28, 2012 at 6:14 PM, Alejandro Imassa...@p2ee.org wrote: Please expand as to how you set-up a SIP ATA behind a common home router set-up, without port redirection and/or use of a SIP proxy and/or STUN server? Unless the ATA has some sort of magic (e.g. VPN support) it _cannot_ be done. Go buy a WRT-54G or nearly any consumer-class router and just plug in a SIP device. Done. It works. We *never* work on customer routers and very rarely have to tell them to reconfigure their router at all. My own home configuration is an Airport Extreme with zero configuration. So either these are very old routers you're having a problem with, or buggy SIP devices, or something else. You should care. Hmm, let me check the reading... http://i1-win.softpedia-static.com/screenshots/Care-Meter_1.png don't drive a Yugo but if I did I could easily be offended by the pejorative use of the brand. It's a piece of junk and everyone knows it, including the owners, so who cares? -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On Tue, Feb 28, 2012 at 8:28 PM, Carlos Alvarez car...@televolve.com wrote: On Tue, Feb 28, 2012 at 6:14 PM, Alejandro Imass a...@p2ee.org wrote: Please expand as to how you set-up a SIP ATA behind a common home router set-up, without port redirection and/or use of a SIP proxy and/or STUN server? Unless the ATA has some sort of magic (e.g. VPN support) it _cannot_ be done. Go buy a WRT-54G or nearly any consumer-class router and just plug in a SIP device. Done. It works. We *never* work on customer routers and very rarely have to tell them to reconfigure their router at all. What you are saying seems impossible and makes no sense unless the router is assigning a public IP or is SIP aware and knows how to read the routing data contained inside the SIP packets, and none of the consumer routers are SIP aware AFAIK, especially not the WRT-54G. The other option is that the SIP ATA has WAN and LAN ports and the SIP device is being assigned a public IP. SIP does not NAT by itself because it can't, because there is no routing info, it's simply impossible. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro Imass Sent: Tuesday, February 28, 2012 10:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great On Tue, Feb 28, 2012 at 8:28 PM, Carlos Alvarez car...@televolve.com wrote: On Tue, Feb 28, 2012 at 6:14 PM, Alejandro Imass a...@p2ee.org wrote: Please expand as to how you set-up a SIP ATA behind a common home router set-up, without port redirection and/or use of a SIP proxy and/or STUN server? Unless the ATA has some sort of magic (e.g. VPN support) it _cannot_ be done. Go buy a WRT-54G or nearly any consumer-class router and just plug in a SIP device. Done. It works. We *never* work on customer routers and very rarely have to tell them to reconfigure their router at all. What you are saying seems impossible and makes no sense unless the router is assigning a public IP or is SIP aware and knows how to read the routing data contained inside the SIP packets, and none of the consumer routers are SIP aware AFAIK, especially not the WRT-54G. The other option is that the SIP ATA has WAN and LAN ports and the SIP device is being assigned a public IP. SIP does not NAT by itself because it can't, because there is no routing info, it's simply impossible. _ You (or more correctly your Asterisk or SIP/RTP proxy) will handle all of that NAT stuff for you. The only time I've ever had issues with SIP and NAT is when the router tries to do SIP ALG while Asterisk is also trying to do NAT fixups. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
