Hi everybody, I use asterisk as a PSTN gateway and found that the audio quality would significantly decrease when asterisk has been busy for a while. Using tcpdump I can see approx. 25% fewer rtp packtes coming from the asterisk box than from my sip client. The lost packets seem to follow an even pattern. About every 4th packet seems to be missing. The audio sounds like gargling sounds are added to it. Audio to the PSTN however is not affected.
This behaviour can be observed on two different UP boxes - one with two E400p and another one with two TE410p in it. Both boxes experienced heavy traffic before showing this behaviour (close two 240 concurrent channels for about 24 hours). Trying to narrow down the problem I found the following: All calls routed to or from the PSTN showed this behaviour. The problem would show no matter whether IAX2 or SIP channels were used on the VOIP side. The direction of a call would not affect the problem. However, calling an IAX2 target from a SIP UA through the same asterisk box whould showed crystal clear audio both ways. The problem also exists independent of the sip ua (Grandstream, Zultys and X-ten were affected). Interesting enough restarting asterisk would reliably solve this issue. Has anybody experienced something like this before? Or even better: knows how to fix it? Thilo _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
