asterisk users - Feb 2012 - 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

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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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
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

On 02/28/2012 03:08 PM, Troy Telford wrote:
> [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
A serious bug with IAX2 trunking in recent versions of Asterisk (you did 
not mention what version you are using) was just resolved last week. You 
should test with 'trunk=no' to see if that is the cause of your problem;
it seems very likely.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

On Wed, Feb 29, 2012 at 10:43 AM, Carlos Alvarez <carlos at
televolve.com>wrote:
> On Wed, Feb 29, 2012 at 8:41 AM, Steve Totaro
> <stotaro at asteriskhelpdesk.com> wrote:
> > Agreed with one exception, the endpoint behind the NAT DOES need to be
> setup
> > correctly to keep the router from seeing inbound traffic to the device
as
> > unsolicited and drop it.  That is a function of the router but keep
> alives
> > from Qualify on the Asterisk side, and setting the device to register
> every
> > few minutes will keep that mapping open and alive, letting traffic
pass
> as
> > solicited.
>
> We use qualify=yes on Asterisk and a few months ago turned OFF the
> keep-alive feature on all SIP clients on our entire system.  This is
> working fine, and we did it because of a strange bug/behavior with
> certain versions of Cisco SPA series firmware.
>
>
> --
> Carlos Alvarez
> TelEvolve
> 602-889-3003
>
>
So you turned it off on the phones but use it on the Asterisk side?

Do you set a value or just use qualify=yes?

I had many problems with qualify over VSAT as ping times and jitter are
crazy.  700ms ping times were considered "Good" from the IZ in Iraq to
Equinix data center in VA, it took some tweaking to find the right value so
a phone that was "Reachable" was not labeled "Unreachable",
I did want
phones that were truly unreachable to be marked as such, more to spot
patterns and act on them or with the vendor.

Did you submit a bug report?  If it is easy to reproduce and you feel like
helping out, report it.  I do not report issues if there is a simple way to
do the same thing, but I know I should.

What does the debug or strange behavior look like?  Probably a variance in
the RFC implementation.

Thanks,
Steve T
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