asterisk users - Mar 2006 - Problem with chan_iax.c implimentation causes bad audio?

I received an e-mail from a vendor who says:

"We have recently become aware of an issue in the chan_iax2
implementation of IAX2. This issue leads to degraded audio quality.
Due to this we are urging everyone to move to SIP."

I don't want to discount what this person is talling me, but I'm
curious to know why I would only be having issues connecting to his
servers, and also what exactly the issue is (if anyone knows).   I was
always under the impression that IAX2 was a better way to connect
servers and was more advanced (jitterbuffer/etc) then sip was.

Can anyone comment on this?

On Mar 20, 2006, at 3:47 PM, Matt wrote:
> I received an e-mail from a vendor who says:
>
> "We have recently become aware of an issue in the chan_iax2
> implementation of IAX2. This issue leads to degraded audio quality.
> Due to this we are urging everyone to move to SIP."
>
> I don't want to discount what this person is talling me, but I'm
> curious to know why I would only be having issues connecting to his
> servers, and also what exactly the issue is (if anyone knows).   I was
> always under the impression that IAX2 was a better way to connect
> servers and was more advanced (jitterbuffer/etc) then sip was.
>
> Can anyone comment on this?
Which vendor?  I use IAX2 for connections to both Nufone and Teliax, 
and they both are working great.  I don't know what the vendor is 
referring to,  but I suspect without more info that it's FUD.

On Monday 20 March 2006 18:47, Matt wrote:
> "We have recently become aware of an issue in the chan_iax2
> implementation of IAX2. This issue leads to degraded audio quality.
> Due to this we are urging everyone to move to SIP."
That's from Asterlink; The way that they handle their IAX2 registrations
seems
a little different from everyone else, and as a precaution they are 
recommending everyone use SIP to terminate through them.

These are a smart bunch of guys, but my personal opinion is that it's their 
implementation that is causing issues, not IAX2 itself since, as others have 
noted, several other VOIP termination providers are not experiencing these 
issues.

-A.

Matt wrote:
> I received an e-mail from a vendor who says:
> 
> "We have recently become aware of an issue in the chan_iax2
> implementation of IAX2. This issue leads to degraded audio quality.
> Due to this we are urging everyone to move to SIP."
> 
> I don't want to discount what this person is talling me, but I'm
> curious to know why I would only be having issues connecting to his
> servers, and also what exactly the issue is (if anyone knows).   I was
> always under the impression that IAX2 was a better way to connect
> servers and was more advanced (jitterbuffer/etc) then sip was.
> 
> Can anyone comment on this?
There have been a number of interoperability issues with iax over the 
last year or so. It seems the majority are related to bugs associated 
with counter rollovers, jitterbuffer changes, frames sent with identical 
counters/timestamps, dtmf encoding, issues with certain codecs, etc. I'd 
hate to have the job of creating a matrix of which * versions function 
with other versions knowing full well that multiple changes occurred 
between versions. If you search the bug tracker for open & closed iax 
issues, you'll see a number of them. (Note: not all iax changes came 
through the bug tracker either.)

Add to that the fact that iax is actually a proprietary protocol 
implementation (eg, not based on any current published/approved 
standards), and the fact that only folks that run asterisk actually use 
the protocol, you now have a fairly major support issue from the itsp's 
perspective. Couple all of the above with how many newbies try to 
implement an * system with almost zero knowledge of how to implement or 
support their own system, and its not difficult to understand why the 
itsp's have a support issue with iax.

Given the majority of itsp's have had to modify source code to address 
their own operational/business objectives, its not at all easy for them 
to keep up to date with asterisk releases & patches.

Compare that to the stability of the underlying sip/rtp protocols and I 
think you'll reach a conclusion that is similar to the itsp that told 
you that.

FWIW, I'll continue to use iax with my itsp's. ;)