I am very new to VoIP but we have a disaster relief going on in New Orleans and we need VoIP implemented at an access site down there. It is going to be 25 phones communicating between two 2811 routers. The site is currently a 512k (256k CIR) Frame Relay that is at about 40% utilization (we have an OC3 head end). I was curious if any recommendations were known as to how much bandwidth the voice network would require. I was going to submit a request to upgrade the T1 but I wanted to see if anyone had any recommendations so we can run VoIP without impacting the existing data network. Any input would be greatly appreciated. Thanks!
You definitely need to look into implementing QoS on this link, i would recommend the following:LLQ to prioritize the voice packets, bandwidth depending on number of concurrent calls, perhaps for start assign 50 percent (128 kbps since this calculation is based on your bandwidth parameter) this will give you approximetely 5 G.729 simultaneous calls. For link slower than 768kbps implement LFI. Also, because your head end is OC-3 you might want to implement Frame-relay traffic shaping. If you need more than 5 simultaneous calls, I would recommand getting more CIR, if your provider does not drop DE packets you can try faking the bandwidth parameter to full line speed, note however that it will affect your routing protocol metric calulation. Another tool is also payload compression, however with G.729 you want save that much badwidth. Go over the SRND docs to get the configs.
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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 ...