Hi,
I want to be able to extract QoS parameters such as jitter, delay and packet loss from VoIP calls made using my SER(both peer to peer for my public clients and also for clients where the voice will be routed with mediaproxy). I would like to develop something as opposed to purchasing a software package.
From searching the Internet the last few days I have seen that the
following options seem to be available:
1) RTCP RTCP stream will contain this information. The problem is that sniffers cannot clearly distinguish RTCP packets from other UDP packets unless something like an IP address or specific UDP port is supplied. I had considered using TCPDUMP but then where do I position this in the network when voice is going peer to peer between client and I'm not sure how this can work to monitor all calls when I might not be aware of the end user IP addresses.
2) SIP INFO extension This also seem to be an option but most off the shelf phones don't seem to support this and this would required modification of a SIP ua.
3) SNMP I thought maybe SNMP might be an option but the SER snmp module no longer exists...
Does anyone have any comments on the above? Are the statements that I made correct or can anyone think of other ways to monitor the voice QoS? I am trying to understand how commercial applications have accomplished this.
Many thanks in advance for advice. Aisling.
---- Original Message ---- From: clona@cyberhouse.no To: ashling.odriscoll@cit.ie Subject: Re: [Serusers] Status of SER SNMP module Date: Sun, 29 Jan 2006 20:30:02 +0100
Hi Aisling,
The SNMP module is dead as a killed turtle :(
-Atle
- Aisling O'Driscoll ashling.odriscoll@cit.ie [060129 20:26]:
Hello,
I'm just wondering what the current status of the SER SNMP module
is?
Is it currently supported and is there somewhere I can find documentation on it?
Many thanks, Aisling.
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