[asterisk-users] Asterisk sends packets on 8004/udp
For the life of me can't figure out why the Asterisk server generates an enormous quantity of outgoing packets on port 8004/udp. They seem to have no effect whether they are blocked by the firewall or not. We're running SIP. Everything appears to be OK (except a large number of ChanSpy write buffer overflow messages, which I also don't understand). How can I discover whether these packets are desirable or not? What should I do with them? I have found no documentation on port 8004. Thanks in advance, Yitzhak Bar Geva ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk sends packets on 8004/udp
For the life of me can't figure out why the Asterisk server generates an enormous quantity of outgoing packets on port 8004/udp. They seem to have no effect whether they are blocked by the firewall or not. We're running SIP. Everything appears to be OK (except a large number of ChanSpy write buffer overflow messages, which I also don't understand). How can I discover whether these packets are desirable or not? What should I do with them? I have found no documentation on port 8004. Thanks in advance, Yitzhak Bar Geva ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] NAT traversal packet loss measurement
How can one measure the effect of NAT traversal packet loss? We currently have no solution for NAT traversal for our SIP clients. There is no doubt that packets are getting lost. What is not clear is how much damage this does. On the face of it, everything seems fine. Could this be so? Perhaps we're suffering a degradation in quality or our call setup times could be improved. How can we measure this? What's the simplest method of preventing packet loss due to NAT traversal in a SIP environment? Thanks, Yitzhak ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
