Hi All,
Got a strange (at least IMHO) issue that doesn't make much sense to me.
Basic configuration is two sites with a site-to-site (aka router-to-router)
VPN. Both sites have Cisco 7961G phones [with SIP firmware] on users' desks,
and the only VoIP is internal - all of our outward telecom is on POTS or
Centrex-enabled POTS lines.
Site 1 has a Dell PowerEdge 1950 with Asterisk built from source and an AEX804E
to connect to the outside world. Site 2 has an Asterisk Appliance with the 4
FXO / 4 FXS configuration, with the FXS ports currently unused.
The PBXes at each site are configured to be essentially independent but with a
unified dial plan so that calls can be placed or transferred across the VPN
with a SIP trunk connecting the two PBXes, and canreinvite=no is set
everywhere. The only other "heavy" consumer of bandwidth across the VPN is a
real-time file replication suite that we use for file synchronization. While
this is the ultimate issue, I don't understand the phenomena I'm seeing:
If a user dials in to one of Site 2 FXO lines then dials across the VPN to a
user at Site 1 while the file replication job is running audio quality (to the
caller on the POTS line only) is abysmal-- audibly it sounds like about 50% of
the packets are dropped ("He__ Th__ __u __r, T__s is ___ln")
On the other hand if a user at Site 2 picks up one of the Cisco phones [with
the replication job still running] and dials across the VPN to a user at Site 1
audio quality is fantastic, ditto if a user at Site 1 calls to a user at Site 2.
Any ideas why the audio quality would be so markedly different when the only
thing that seems to be different is where the call is originating from (POTS
line vs. SIP phone)?
Replacing the border gear with equipment that's QOS aware and can handle
prioritization is already on the list (and may be in the process of being
ordered at this point)
Thanks,
Lincoln
--
Lincoln King-Cliby, CTS
Applications Engineer
ControlWorks Consulting, LLC
Crestron Authorized Independent Programmer
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