First post here. We are about to look at rolling out a Cisco IP Telephony solution and are currently looking for partners to work with us..
Before we do that, I wanted to understand some basics if that's OK?
I understand that the call setup is achieved via RTP which is seperate to the the voice packets?
I've read that RTP is not often sent in the EF class and is normally sent in the AF class - is this the case, if so, what kind of bandwidth do we need to allocate for RTP?
Looking at the models - Centralised, Distributed, Deployment and Geographical Diversity (I think that covers it). Our model looks to be distributed with one large Campus site and lots of smaller branches - if someone in a branch calls someone else in the same branch, does the actual call go via the HQ site or is it just the RTP (i.e. call setup) that is sent to the HQ - i.e. are the actual voice packets are sent across the same LAN or do they have to be sent via the HQ?
Hi, thanks for the response. So, if a user calls another user on the same LAN, would the voice packets go direct between the 2 users on the LAN? I'm assuming just the call setup is sent to the HQ but the actual packets are sent direct between users, is this correct?
Per-call bandwidth depends one what codec you use (e.g. G.711 vs. G.729) and what network transport it crosses. You'll need to determine the codec choice for intra- and inter-location calls. Additionally, http://www.bandcalc.com/ provides a handy tool for multiplying this out.
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The short answer is that you don't.... That isn't entirely true while at
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Topology: IP Phone > Switches > Microsoft NPS setup to forward 802.1x
proxy to > ISE 2.1 patch 3 Authentication: EAP-TLS using Cisco MIC SANs
Phone Models 802.1X support? 802.1x flavor Addtl Comment EAP-MD5 EAP-TLS
Cisco 3905 Y Y N Cisco 6911 Y Y N Cisco ...