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VOIP calls Quality

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penroy

MIS
Feb 12, 2006
116
JM
I have 3 pbx connected by frame and wireless link. I Ihave a call centre where all calls terminated and then routed to their destination.

How can I do the following.

1.To be able to allocate bandwidth to voice traffic.
2.prioritize voip traffic over data accross these links.

Does any one have any sugestion on what can done to make the voice quality good going over the link.

Nb(cisco routers,cisco switch are in the network)

 
I have been using the i2004 phones, which have the ability to share the network connection between the phone and the pc. So far I have 3 branches fully deployed this way with no issues. I originally doubted the quality of this design, so I added IP phones this way in my IT department. Tech guys make the best test subjects, as they are less prone to make a gigantic issue out of nothing.
I also though have pri's between all sites, most of the pri's at smalller branches are split 6 channels voice, 17 data.
 
You need to set up whats called QoS (Quality of Service) on all the routers and Switches in the network where the VoIP calls are handled (LAN and then MAN/WAN if these are involved) You Cisso doc's should help you on this. It's quite an involved area to set up correctly, proberbly beyound the scope of TekTips.
As a rough giude, the more "hops" your going through, the more likely your going to encounter an issue.


Only the truly stupid believe they know everything.
Stu.. 2004
 
I'd also suggest using VLANS to "seperate" the voice and data

Take Care

Matt
If at first you don't succeed, skydiving is not for you.
 
VLANs, 802.1p or Diffserv are common QoS implementations.
For Diffserv, your server have to marked Voice packets.
Additionnaly, if you're expecting to have PC behind Ip set, 802.1q have to be enable for trunking Voice packets.

QoS is not 1 solution. It's strongly depends of how you network is design. I suggest to find Whitepapers on the Net speaking about it.
 
Wireless can easlily be an issue. There is no QoS on most wireless devices as the delay can create issues. The other responses are correct when it comes to VLans and QoS. There are several good areas you may want to search for "QoS" and "VLAN". You may want to look in the Cisco forum and even the Avaya Definity or IP Office forum for QoS help. Remember that Cisco has some propietary protocols on their switches that coudl actually hurt voip calls on non Cisco phone systems. They are pretty easy to disable but once again you may want to search for "CDP" (Cisco Device Protocol.Hope this helps.
 
CDP interferes with (non cisco) VOIP? Thats one to file away for future reference!

Are there any docs, or is this experience based?

Take Care

Matt
If at first you don't succeed, skydiving is not for you.
 
It is in every IP Telephony Implementation guide.

One of the "best practices" notes for each Avaya implementation is CDP causing potential issues.


Here is an example of the text

"Note: Since the Avaya products do not work with the Cisco CDP protocol; the command set port cdp disable should be used to disable the CDP protocol on the switch ports that are connected to the Avaya devices. "
--------------------------------------------------------------------------------
 
First thing I would do is remove any wireless links that you have any voice traffic on. There is no QoS for Data going over WIFI...

Second make sure all your routes over you Cisco WAN netowrk have some type of QoS programmed in them. My preferred set up on Cisco (for any type of PBX vendor)is the following:

Fair wieghted Queuing Policy Map / Access List
Set your TCP and UDP ports that are used for your voice
traffic in you Access List so they wil have priority.
Set how much bandwidth will be allocated to that Policy Map. Most installs get between 512k and 768k out of a 1.5m T-1.

Since you are running on a frame relay network, there is another very important factor to consider !!! What is your CIR from your office to telco?? You might have a full T-1 from your office to telco but your only paying for a 256k committed rate. Basically what happens is that after you access more than 256k on your fame the carrier has the right to start dropping packets. In the data enviorment you might not notice this change because data can regenerate the packets again if lost and resend. In the voice enviorment if your packet doesn't get there the first time and with as little latency as possible (under 44ms is ideal), voice quality just went out the door !!!

Also chech and replace the simple things at Layer 1. Cat.5 cables not being seated correctly or a lose connection with cause latency in your voice traffic. Specifically the cable going from your Smart Jack at the Dmarc (where telco extends the curcit) to your equipment.

From my 8 years of doing voice via networks the above idems are the biggest reasons why a customer has good or bad voice quality.

Just remember there is no such thing as VoiP(Voice over IP). All that has happened is Data got bigger !!!
 
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