[asterisk-users] Graphing Jitter Packet loss and Latency for SIP Calls

[Vikas]

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
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