Requirement: Monitor the QOS for the SIP phones connecting to the voip server.
Ideal solution: Browder based reporting software that I can install on the asterisk server (I use freepbx) and when I connect to this reporting engine it gives me the Jitter loss, packet loss and latency for each of the calls that the extensions connecting to this asterisk server make and receive. Network design: A. The sip endpoints: 6 polycom 650 phones in India connecting to an VOIP server. B. Network between the SIP endpoints and VOIP server: The Indian office has 5 different ISPs giving the internet connection. Each ISP has a different packet loss latnecy and Jitter and these change over time. So I want a way to be able to select the best ISP on a given day. C. VOIP server: hosted at he.net datacenter and acts as the gateway between the sip endpoints and the PSTN gateway. D. PSTN gateway: Broadvoice for outgoing calls and exgn.net for incoming calls on the 800 number Things I have looked at: 1. Wireshark -> I did not find a good reporting engine which I can automate to collect data and then graph it. 2. Endian 2.2 3. IPCop I would really appreciate any insights on how to monitor the QOS. Thanks for your time, Sysadmin http://www.debtconsolidationcare.com Internets First get out of debt community _______________________________________________ -- 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