[asterisk-users] Same provider - IAX sounds bad, SIP sounds great
Steve Totaro
stotaro at asteriskhelpdesk.com
Tue Feb 28 15:56:00 CST 2012
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/20120228/87cc74a9/attachment.htm>
More information about the asterisk-users
mailing list