Sunday, February 15, 2015

Understanding SNR and RSSI values

Hi Guys

I saw this post on a forum while investigating other issues and had to note it down for you and for my own reference! Finally a definitive answer easily understood to a question I have had for ages regarding determining wireless performance

SNR (Signal-to-Noise Ratio) is a ratio based value that evaluates your signal based on the noise being seen. So let's look at the components of the SNR and they see how to determine it.  SNR is comprised of 2 values and is measured as a positive value between 0db and 120db and the closer it is to 120db the better: Signal Value and Noise Value typically these are expressed in decibels (db).
     So we will look at the Signal (Also known as RSSI) first this value is measured in decibels from 0 (zero) to -120 (minus 120) now when looking at this value the closer to 0 (zero) the stronger the signal is which means it's better, typically voice networks require a -65db or better signal level while a data network needs -80db or better.  Normal range in a network would be -45db to -87db depending on power levels and design; since the Signal is affected by the APs transmit power & antenna aswell as the clients antenna.

Great stuff, found the post here:

 Also worth pointing out as per his post that the 7925g handsets can actually be used to perform site surveys! Another handy trick!

 For more information

Tuesday, February 3, 2015

Match Incoming calls based on SIP URI!!! (Or SIP Host or SIP Ip address, super useful!) match incoming dial-peer on sip address

Hi Guys!

You can now match incoming calls based on the SIP Address sending to you

This could be incredibly useful in a situation where you don't know what numbers the provider might be sending but you need a way to distingush a provider call from another call.

See below some example configuration:

voice class uri  ACMESIPTRUNK sip
 host ipv4:
 host ipv4:

dial-peer voice 2 voip
 description ### Incoming calls from AT&T SIP Trunk ###
 session protocol sipv2
 incoming uri via ACMESIPTRUNK
 voice-class codec 1 
 dtmf-relay cisco-rtp sip-kpml sip-notify

This will make this dial-peer voice 2 be the incoming dial-peer for any calls from host and