[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.
More information about the asterisk-users
mailing list