[Asterisk-Users] Audio pausing over IAX trunk

[Steve Kann]

Florian Overkamp wrote:

>Hi Steve, 
>
>  
>
>>-----Original Message-----
>>    
>>
>>>I am having a problem with periodic breaks in audio over an 
>>>      
>>>
>>IAX trunk. 
>>    
>>
>>>The interruption only happens in one direction, and (I think) only 
>>>with clients built on the open source libiax.
>>>
>>>Codec is irrelevant, and jitterbuffer on/off seems to make no 
>>>difference either. The pause happens every few seconds, and 
>>>      
>>>
>>is regular.
>>    
>>
>
>  
>
>>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