[Asterisk-Users] Sound issues on SIP-SIP calls

[Martin Joseph]

On Mar 22, 2006, at 5:31 AM, Bjorn O wrote:

> Hello all!
>  
> For several months now we’ve been experiencing a really strange 
> problem with sound which best can be explained as choppy/stuttery, and 
> with a touch of echo on top. Basically, parts of a conversation might 
> be choppy, but often combined with some echo as well. The sound 
> problem can only be heard by us, not by the other party. But this is 
> the strange part:
>  
> The problem only occurs periodically, and only towards one single SIP 
> provider. All our equipment is SIP based (we use Cisco 79XX equipment 
> with firmware 7.5 and 8.x), and we have no PSTN lines. Since this has 
> been lasting for months, I don’t know the exact time the problem 
> occurred at first. I recall everything as fine when I ran Asterisk 
> 1.0.7, but through 1.0.9, 1.2.2 and 1.2.5 (HEAD) this problem 
> persists. However, I have a hard time believing this is caused by a 
> bug. Why? Well, as said, we have no problems with other providers. In 
> addition, we have Cisco phones that communicate with Asterisk at the 
> office (traffic is switched between two local subnets) and also over 
> the internet, and all calls made between these phones are completed 
> without any trouble.
>  
> So I tried doing a clean install on a fresh CentOS last week, but with 
> no luck – the problem is still there. I also checked if there were 
> load or IRQ issues, which there is not.The box runs APF firewall and 
> has ports 5060-5070, as well as 8000-10000 (for rtp) open tcp/udp. 
> From the phone information menu I can see that frame size is most 
> likely set to 20 ms, while the provider uses 20 or 40 ms (not sure).
>  
> I should add that this provider is among the larger ones in Norway 
> (IP24). Other users have reported Asterisk to work just fine with it 
> in the Provider’s local forum, but that is certainly not my 
> experience. So my question is, is there anything I might have 
> overlooked, or is there a particular way I can debug this issue?
>  

Sounds like a jitter issue, based on your description.  When this 
happens have you looked at a traceroute to the SIP call terminator?  
Perhaps there is a flakey router in your path?

Marty

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