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

Tue Feb 28 17:07:13 CST 2012

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
>>
>>
>>
>> --
>> _____________________________________________________________________
>> -- 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
>>
>>
>>
>>
>>
>>
>> --
>> _____________________________________________________________________
>> -- 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