[Asterisk-Users] ADSI phone vs. IP phone (and proper implementation thereof)

Ray Burkholder ray at oneunified.net
Mon Jan 19 17:38:07 MST 2004


Quoting daryl at introspect.net:

> > Why wouldn't you just use your existing Ethernet 
> > infrastructure putting 
> > the  IP phones inline between the wall jack and the PC? There are a 
> > number of IP phones that have builtin switch/hub that allows 
> > the PC to 
> > daisy chain off the IP phone.
> 
> Probably because it's well known that these setups are prone to failure
> of either the PC's connection, the phone's connection, or degredation of
> one/both.  It also breaks switch envirenments where spanning-tree
> portfast is enabled (not as big of a deal if the deployment is in
> concert with the infrastructure group, as it should be).
> 
> Vendors should NEVER have implemented this functionality into phones
> unless it was working under all conditions.  Personal experience shows
> that it is most definitely not on Cisco and 3Com products.  Others have
> told me their stories with other manufacturer's equipment.  None of it
> was good.
> 
> It's not a production-stable way to deploy phones.  Period.

I'm wondering if what you say is actually true.  According to recent media 
releases, Cisco has shipped over 2 million of their IP phones.  They must be 
doing something right.  Their phones are _designed_ to function and cooperate 
with the switch.  Obviously, the installer has to be totally familiar with all 
phone, switch, router and network settings in order to have a successful 
installation.

The switch needs to be configured with specific port, vlan, and class of 
service settings.  Accepted practice is to provide a voice vlan and a data vlan.

On the phone side, the phone knows to send voice on the specific vlan told to 
it by the switch , and to pass through data from the pc through the vlan told 
to it by the switch.  The phone knows to prioritize voice traffic over data 
traffic.  So does the switch.  And so on through the connection of switches and 
routers.  This ensures voice quality and precedence through out the network.

Voip quality is not necessarily about bandwidth (because it works on T1 data 
lines as well as GB ports), but about instantaneous bottlenecks in the 
network.  These instantaneous and random bottlenecks can occur in the cad 
environment mentioned.  But with appropriate COS (layer 2) and TOS (layer 3) 
settings in the phones, switches, and routers, these bottlenecks become non-
issues.

In addition, what many people forget, or learn by experience, is that you 
absolutely _must_ have everything running full-duplex, and to physically check 
errors and statistics on each port of the switch in order to verify that you 
have error free links.  You won't believe how many networks out there are 
broken because noone checks and fixes these issues.  A voip network _must_ have 
managed switches so you can verify these things.

There was mention of a heavy cad environment.  Say your computer is connected 
to the 100mbps port of the phone.  A g.711 call comes through.  The call takes 
around 80 kbps.  If I've done the math properly, the voice call takes only 
0.08% of the bandwidth, hardly something that will interfere with 'heavy cad 
users'.  More likely the opposite, the heavy cad users will interfere with the 
call, _but_ _only_ if the switch and phone are not configured properly for 
vlan, cos, tos, speed, and duplex settings.

So having said this, you mentioned that you have had personal experience where 
this functionality is built into, or does not work in Cisco's case.

Could you provide some additional info on what did not work for you?  Because 
my experience is opposite:  things do work when configured according to 
manufacturer specifications and using the correct equipment.

Ray.

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