[asterisk-users] Same provider - IAX sounds bad, SIP sounds great

Tue Feb 28 17:11:22 CST 2012

And the dude arrives talking about penis......

On Tue, Feb 28, 2012 at 6:07 PM, Carlos Alvarez <carlos at televolve.com>wrote:

> 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
> <stotaro at 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 <danny at 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-bounces at lists.digium.com
> >> [mailto:asterisk-users-bounces at 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
> >>
> >>
> >>
> >> OOOOPSS
> >>
> >>
> >>
> >> http://bit.ly/ywiwzt
> >>
> >> On Tue, Feb 28, 2012 at 4:56 PM, Steve Totaro
> >> <stotaro at 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.groups at 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 +0000, 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-bounces at lists.digium.com
> >> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Troy
> Telford
> >> Sent: Tuesday, February 28, 2012 4:08 PM
> >> To: asterisk-users at 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
> >>
> >>
> >>
> >> --
> >> _____________________________________________________________________
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> >> New to Asterisk? Join us for a live introductory webinar every Thurs:
> >>               http://www.asterisk.org/hello
> >>
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> >>
> >> 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
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> >>  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
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> >>   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
> > --
> > _____________________________________________________________________
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> > New to Asterisk? Join us for a live introductory webinar every Thurs:
> >               http://www.asterisk.org/hello
> >
> > asterisk-users mailing list
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> >   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> --
> 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
>
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