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

[Troy Telford]

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