[Asterisk-Users] Ast 1.0.7, IP-500's with unmanaged switch...remote end missing bits of audio

[Marty Mastera]

I thought I would throw this out there and see if anyone has any
ideas...I have the same problem at 2 locations.
 
The complaint from the users is that calls "cut out", "kinda like when
you have spotty cell coverage". Doesn't seem to matter whether the call
is incoming or outgoing, although it might be true that my users hear
the remote party cut out, while the remote party doesn't notice the same
from my users...
 
Location 1:
 
- SDSL 1.5 Mpbs with static IP, Netopia 4652 SDSL router (enabled
"Prioritize Delay Sensitive Data" to recognize tos=lowdelay per Netopia
support)
- Dell PowerEdge SC420 with TDM04B (currently only using one port.  the
single analog line is call forward on busy to my IAX provider)
- Asterisk CVS-v1-0-02/22/05 using IAX to connect to my provider over
the public internet - I have run pings for an extended period of time
against my provider's server and get no packet loss.
- In IAX.conf: tos=lowdelay, jitterbuffer=yes, also enabled "Prioritize
Delay Sensitive Data" on the Netopia to support tos=lowdelay per Netopia
support
- Average ping time to my provider: 160 ms with no packet loss
- 8 Polycom IP-500's running SIP 1.4.1.0040 and bootrom 2.6.1 using ulaw
only
- Small Business Server 2003 set up as DC for the network
- Two network laser printers
- 24 port unmanaged switch (all phones are home run back to a patch
panel, patched from there into a switch port.  The DSL modem, printers
and server are patched into the switch in the same way)
- 8 pc's running XP Pro, all plugged into the switch port on the back of
the IP-500's
 
Location 2:
 
- Full rate data T1
- Dell PowerEdge SC1420 with no TDM hardware at all (this location
connects SIP directly to the T1 providers BroadSoft switch and does not
go over the public internet)
- Asterisk 1.0.7 - using SIP to connect with my provider (not across
public internet, not natted since the Cisco IAD does the SIP mangling
for us)
- Average ping time to the broadsoft switch: 42 ms
- 8 Polycom IP-500's running SIP 1.4.1.0040 and bootrom 2.6.1 using ulaw
only
- Small Business Server 2003 set up as DC for the network
- One network printer
- 24 port unmanaged switch (all phones are home run back to a patch
panel, patched from there into a switch port.  The DSL modem, printers
and server are patched into the switch in the same way)
- 8 pc's running XP Pro, all plugged into the switch port on the back of
the IP-500's
 
 
As you can see, the only commonalities are Dell hardware (but not
models), Asterisk (but not versions), IP-500's (including sip and
bootrom version), SBS 2003, 24 port unmanaged switch, the fact that all
the pc's are plugged into the switch ports on the phones.
 
Same symptoms at both locations.  I cannot determine any specific causes
(ie it doesn't seem to be inbound vs. outbound, etc).  I have checked
all the pc's for viruses and worms, changed switch ports, etc...the only
theory I have right now is that since the Polycoms give priority to
outbound phone traffic vs a connected PC, that outbound voice is getting
the QOS it needs.  Versus inbound voice which gets no priority treatment
once it hits either LAN since the switch can't do any QOS.  Am I on the
right track with this theory?  Do I need to try a managed switch, giving
priority to voice to make sure that both incoming and outgoing voice
packets are preferred?  On a side note, at what point (size - number of
clients) is a managed switch recommended?  required?
 
If I'm off base with the QOS theory, what else should I be looking at?  
 
 
Marty
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