QOS

So I began yet another QOS venture tonight. Watched the nuggets video(s). Here are just some random notes. They will probably be of no use to anybody but me, but here they are:

TOP NETWORK CULPRITS (in terms of good service over the wire!)

1. Lack of bandwidth
2. Packet loss
3. Delay
4. Jitter (variation of delay...i.e. on packet has 20ms delay, next 5, next 55...

QOS MODELS

1. Best Effort (it gets there when it gets there!! like a bitchy old mail lady who is disgruntled with her job)
2. Integrated Services Model (reserved bandwidth from point A ----> point B using RSVP) intserv
3. Differentiated Services Model (this is the best...puts up and breaks down as packets traverse interface)

IMPLEMENTING QOS

1. Do a network audit and identify the critical applications
2. Identify the application requirements (i.e. what protocols...bandwidth...etc)
3. Divide that traffic into groups (i.e. classify it into approx. 4-11 groups)
4. Create and apply policy to those groups

-Qos is basically just classifying and marking traffic

CONGESTION AVOIDANCE MECHANIMS

1. FIFO and Tail-drop
2. Random early detection (RED)
3. Weighted random early detection (WRED)

-Policing traffic drops all packets after a certain threshold is reached (i.e. you say no more after 200kbps)
-Shaping traffic merely queues packets when a pre-defined limit is reached

METHODS OF IMPLEMENTING QOS

1. CLI (hardest method)
2. MQC (modular QOS CLI)
3. AutoQOS (cisco proprietary)
4. SDM
5. QPM (plug-in of CiscoWorks)
   

POTS setup on FXS ports

Here is just a basic setup where the two routers have FXS ports with regular analog telephones sitting on them. The two routers are connected via some kind of ethernet link, and share IP addresses on the same subnet. Here is the visio with embedded configs:

Also just some basic "show" commands to post here:
show voice port summary
show dial-peer voice summary
debug voip dial-peer

....noted

Back for the ONT!

Sorry dudes...been out for the holidays. Back and cranking on the ONT. Passed the ISCW a while back with a decent score, but really want to kill this one to complete the CCNP. Here we go...

Default Cisco Voice sample size is 20ms
***Large sample sizes cause more coding delay, and result in greater, more noticeable ruined audio if the packet is lost or dropped.

http://www.voip-calculator.com/calculator -- a good voip calculator


Total packet size/payload size = total bandwidth requirement/total codec requirement
so to get total bandwidth requirement take:
total packet size * codec requirement / payload size = total bandwidth required

so for G.729 on a frame relay take this:
(6 (for frame header)+40 (l3 and l4)+ 20 (payload size for g.729) * 8kbps (codec req.)
_________________________________________________________________________________
20 (payload size for g.729)

= 26.4 kbps(multiply by total number of calls you expect at 1 time)

VAD- Voice activity detection is a mechanism that stops sending traffic during periods of silence

Thats a start. Also watched the VOIP design considerations video. Contained tidbits about call managers, phones, voip design models, and call admission controls.