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

Tue Feb 28 15:58:51 CST 2012

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<asterisk-users-bounces at lists.digium.com>[mailto:
>>> asterisk-users-**bounces at lists.digium.com<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 <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<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<http://lists.digium.com/mailman/listinfo/asterisk-users>
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120228/0cfd8778/attachment.htm>