I found this link on http://www.techexams.net/forums. I'd heard of BlindHog before, but didn't know he had a guide for emulating a CCIE Voice lab! Apparently he piggybacked (pun definitely intended) off of someone else's work, but he added on top of it with the guides and videos on making it all work.

Wow, this is awesome. Great work, guys.

http://www.blindhog.net/how-to-build-a-voice-lab-with-gns3-and-vmware/

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So I'm back in school and finding time to work 40 hours, go to class, study, train my dog (duck hunting), and mix in a bit of sleeping/eating has proven difficult! This blog drew the short end of the stick, so it will be put on hold until I can get a random bit of time to keep up my VOIP studying.

/sadface

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I decided to take some time off of CBT's and reading to get in the thick of things with some labs.

I completed the scenario-based lab on setting up the embedded wireless access point and a wireless IP phone. The devices I used were:

Cisco UC520

Cisco 7920 Wireless IP Phone

The setup was actually quite easy using the Cisco Configuration Assistant. I already had the community built for my lab, so all that was required to complete this lab was to:

1.) Set up the wireless access point for voice usage

2.) Ensure the voice AP uses the voice vlan.

3.) Set up the network profile in the wireless IP phone.

4.) Connect and test voice calls using the voice AP.

After completing these steps, I had another person test calls with me. This was tested in an environment with a large amount of wireless saturation. It worked great and I could get about 50-60 yards before the signal would begin degrading. Another 10-20 yards before the call would drop.

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Now watching:

Gateways and Trunks: Connecting CME to Other Voice Systems

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Real-time Transport Protocol (RTP) - transports audio between devices

Uses random, even UDP ports between 16,384 and 32,766

Real-time Transport Control Protocol (RTCP) - transports call statistics between devices

Uses random, odd UDP ports between 16,385 and 32,767.

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Digital Signal Processors (DSPs) offload media processing functions from voice processing equipment.

They provide:

Coding - applying the codec and compressing the audio

Transcoding - if codec changes are needed, such as for a phone that can only receive G.729, the DSP will provide it

Media Termination Point (MTP) - does the work behind the hold feature

Conferencing - provides conferencing by combining audio from various sources into one audio stream

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Voice Activity Detection (VAD): Why waste bandwidth sending silence? VAD removes this need and saves an average of 35% bandwidth.

Compressed RTP: Compresses network and transport layer headers from 40 bytes to 2-4 bytes. Bandwidth savings are codec dependent because each with have a different overall total size, but it's around 40% with G.729, the most common Cisco codec.

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