Florian Overkamp wrote:
Hi Steve,
-----Original Message-----
I am having a problem with periodic breaks in audio over anIAX trunk.
The interruption only happens in one direction, and (I think) only with clients built on the open source libiax.is regular.
Codec is irrelevant, and jitterbuffer on/off seems to make no difference either. The pause happens every few seconds, and
Not unless you can describe the problem more clearly.
Which direction does this happen in, what exactly are these clients you're talking about, and what is does the network look like between the endpoints.
Okay, in my scenario it's like this:
SIP or MGCP phone (mixed env.) -> Asterisk box -> IAX -> Asterisk box -> PSTN or other Asterisk box
We notice users complaining of the fact that the remote end (PSTN) complained about audio drops, while the local user keeps hearing everything. I am not entirely sure if it is just that direction, because I hear noticeable crackles during the call from my (user) end too.
This appears to happen especially when the asterisk boxes involved have a few calls happening, when its nice and quiet on the box, things seem ok. This kind of thing is not or hardly noticable when calling yourself, which makes diagnosis difficult.
I've discussed this with other people on the list, and we notice the
following: IP links are _not_ congested and latency is very stable, so we
are not looking at a network issue. Others have observed that changing the
protocol from IAX2 to SIP is a good workaround. I have not yet been able to
confirm this because we are tied to Asterisk-stable which does not yet have
a very useable SIP dialling format. It's very hard to get a good handle on
this issue, because it pretty much requires a multihomed production box to
work with :-(
I'm not sure exactly what your problem is, but I think that the new JB may help; at the very least, you could run iax2 show netstats, and get an idea of what the right-most asterisk box is seeing.
Also my latest patchset would keep the JB out of the loop on the left-most asterisk box when it's bridging, and on the right-most box, it would use it if you were bridging to the PSTN (i.e. via zap, I guess), and would not use it when you were bridging to another asterisk box via a VoIP protocol..
See http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0002532
-SteveK
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