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:
https://supportforums.cisco.com/discussion/10954591/snr-and-rssi-values
Also worth pointing out as per his post that the 7925g handsets can actually be used to perform site surveys! Another handy trick!
http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/7925g/7_0/english/administration/guide/7925wrnt.html#wp1117827
For more information
Collaboration Edge Deployment Guide
https://ciscocollab.wordpress.com/2014/01/29/deploying-collaboration-edge/
Default login to tandberg is admin password is TANDBERG
Default login to tandberg is admin password is TANDBERG
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
http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube_fund/configuration/xe-3s/asr1000/cube_fund-xe-3s-asr1000-book/voi-inbnd-dp-match-uri.html
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 ipv4:70.30.1.1
host ipv4:70.40.1.1
dial-peer voice 2 voip
corlist incoming INCOMINGFROMATTTOCUCM
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 70.30.1.1 and 70.40.1.1
You can now match incoming calls based on the SIP Address sending to you
http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube_fund/configuration/xe-3s/asr1000/cube_fund-xe-3s-asr1000-book/voi-inbnd-dp-match-uri.html
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 ipv4:70.30.1.1
host ipv4:70.40.1.1
dial-peer voice 2 voip
corlist incoming INCOMINGFROMATTTOCUCM
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 70.30.1.1 and 70.40.1.1
Subscribe to:
Posts (Atom)
Popular old posts.
-
Hi Guys Having spent a lot of time with customers working on vPC deployments, I have found quite a few of the gotcha's for vPC that I w...
-
Hi Guys! This blog post is attempting to be the DEFINITIVE guide on Jumbo MTU, It's a topic that DOES MY HEAD IN! There are SO many ...
-
So some of the readers of this blog might already know this little trick, and what's more some of you might be surprised I didn't kn...