[asterisk-users] Asterisk dropping audio

Edoardo Serra edoardo.serra at webrainstorm.it
Fri Jan 26 12:26:53 MST 2007


Hi all,

I have a problem with Asterisk dropping audio.
While in call, audio gets dropped for a while (from 5 to 60 secs, and 
obviously users often hangup, this means that I'm not sure the audio is 
always coming back after 60 secs), in the meantime the call remains up 
and no SIP signalation is generated.

It happens randomly so it's very difficult to debug.
I cannot see common circumstances when it happens (load average is 
always between 0.10 and 0.95, concurrent calls from 1 to 60 on a 2xXeon 
3GHz with 2GB RAM).

Calls are terminated to PSTN via other Asterisks with E1 (IAX2) or via 
SIP to other VoIP carriers.
That problem happens with every different termination randomly, it also 
happens with calls between our users.
(Well... I cannot exclude it's a termination problem, but I cannot find 
a common way to reproduce it)

I'm using Asterisk 1.2.13 with res_perl (used to do lcr and to post 
customized cdr to mysql)
I also tried 1.2.14 without solving that issue
Kernel is a 2.6.18 vanilla on a linux gentoo

I have g729 codec from digium installed and licensed, there are enough 
licenses available (I was tihinking of an issue of codec but I'm not 
sure it happens only with g729 calls)
I now installed free g729 to see if it helps but I don't have any 
feedback yet

I have an OpenSER acting as a load balancer for 2 asterisks but I don't 
think it could be responsible for that (I'm not using any kind of RTP 
proxy, rtp stream goes directly from user to asterisks)

Every kind of help is really appreciated

Regards

Edoardo Serra
WeBRainstorm S.r.l.



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