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 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
[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'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
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
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
[asterisk-users] dahdi_dummy required?
I'm running Asterisk 1.8.4.4; I'm new to asterisk, and have been reading the Asterisk Definitive Guide, and it mentions that dahdi_dummy should be used to provide an interface for Asterisk to get kernel timing. - espescially if using timing-dependant modules. I have a minor question: is dahdi_dummy necessary or useful anymore - espescially for users who don't have DAHDI hardware? I ask because I just checked out dahdi 2.5 from svn built (against the Linux kernel 3.0) I noticed that dahdi_dummy didn't seem to be built; when I poked around in the changelog, I saw: * README: README: Remove references to dahdi_dummy. Since dahdi_dummy is no longer required remove the references from README. (issue #17959) Reported by: glen201 Origin: http://svnview.digium.com/svn/dahdi?view=revrev=9308 So am I correct in assuming dahdi_dummy isn't needed/useful anymore? -- 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
