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

Troy Telford ttelford.groups at gmail.com
Tue Feb 28 15:55:08 CST 2012


encryption=yes is meaningless if the provider doesn't support it (mine 
doesn't). I put it there in the wild hope they eventually will - and no 
config change will be needed on my part.

Still, when I changed it to encryption=no, and tested there wasn't any 
difference.

So I've tried disabling the jitterbuffer, and encryption, and there's 
no effect on call quality - outgoing (from me -> provider) sounds 
bad/distorted, while incoming sounds great.

On 2012-02-28 21:14:55 +0000, Danny Nicholas said:

> My first two guesses are that encryption is hosing you or that the
> "single-channel" nature of IAX2 may have something to do with it.  IAX2
> "talks" on 1 channel, SIP uses "twisted pair" connotation on two channels
> (as I understand it).
> 
> -----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 3:08 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
> username=
> secret=
> context=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


-- 
Troy Telford





More information about the asterisk-users mailing list