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Where It All Began: Analog Connections
The Evolution: Digital Connections Understanding the PSTN The New Yet Not-So-New Frontier: VoIP
Note: The material in this lecture is covered in the CCNA Voice 640-461 Official Cert Guide. Since this textbook is recommended for this course, but not required, only the information from these slides will be tested in exams, and no other information from this textbook. Some of the information in this chapter is also covered in the required textbook, and these locations will be pointed out during the lecture.
Voice Over IP: Traditional Voice Versus Unified Voice Josh Lowe Winter 2012
Etc.
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A solution that allows multiple calls to be sent over a single wire is needed.
Digital connections are that solution
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This solves the problem of sending the signal long distances, but what about the huge number of cables required to send the voice?
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Each of these channels can support a single voice call, so theoretically a T1 can support up to 24 calls at a time
Outside of Canada, the US, and Japan corporations use E1 circuits which allows you to use up to 30 DS0s for voice calls
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Digital signals dont have this ability, they only send 1s and 0s
To solve this, two primary styles of signaling were created for digital circuits:
Channel associated signaling (CAS) Common channel signaling (CCS)
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T1 connections using CCS only have 23 usable DS0s for voice (as opposed to all 24 with CAS)
A signaling protocol used on the common channel sends the necessary information for all voice channels The most popular signaling protocol for CCS is Q.931 which is used for ISDN circuits
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When using T1 lines, the 24th time slot is always the signaling channel. With E1 lines the 17th time slot is always the signaling channel
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Photograph taken by Andrew Filer (afiler.com). Used under creative commons license
Voice Over IP: Traditional Voice Versus Unified Voice Josh Lowe Winter 2012
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Photograph taken by Andrew Filer (afiler.com). Used under creative commons license Josh Lowe Winter 2012
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http://cityinfrastructure.com/Voice/Voice.html
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SS7 is an out-of-band signaling method used to communicate call setup, routing, billing, and informational signaling through the PSTN to the destination. This is primarily a telephony service provider technology that you typically do not use from a customer standpoint
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VoIP is taking the 1s and 0s of digitized voice and placing them into a data packet with IP addressing information in the headers.
You can then take that VoIP packet and send it across the data network instead of a traditional telephony network So whats so great about sending voice over a data network instead of a telephony network?
Voice Over IP: Traditional Voice Versus Unified Voice Josh Lowe Winter 2012
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For example, say the word cow out loud to yourself, which takes about a second to say
There are really only three sounds, the k, the ahhhhh, and the wuh
Voice Over IP: Traditional Voice Versus Unified Voice Josh Lowe Winter 2012
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Using this process the G.729 codec is able to reduce the bandwidth from 64 kbps down to just 8 kbps for each call!
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G.711 is a baseline codec that pretty much every vendor should support on all of their devices so that phones from competing vendors can still communicate
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Packet loss
Jitter (delay variations)
Although this information is useful, it is not nearly as critical as the actual RTP audio streams Keep this in mind when you configure QoS settings RTCP creates a separate UDP session using an oddnumbered port (one number higher than the RTP port)
Voice Over IP: Traditional Voice Versus Unified Voice Josh Lowe Winter 2012
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