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

Tue Feb 28 16:17:37 CST 2012

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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-- 
Troy Telford

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