[asterisk-users] Diagnosing poor call quality

[Michael Welter]

The advertised datarate (8mb/448k) are the speeds at which the circuit 
between the customer and the central office is clocked and has no 
relationship with *effective* throughput.  At the central office are 
*shared* facilities than connects each DSL connection with the network, 
and over subscription to these shared facilities cause congestion. 
Also, there is no QoS on the Internet, and congestion anywhere between 
the end points will cause poor call quality.

Disclaimer: The following information is several months old--I've since 
moved my customers away from Qwest DSL.

Here in Denver we have Qwest DSL "service" from a central office where 
the effective throughput drops to dialup speeds during the day.  Regular 
web/email users don't usually notice packet loss because dropped packets 
are recovered by the TCP protocol.  For VoIP on UDP, however, the call 
quality suffers to the point of being unusable (clicking, popping, and 
dropouts).

Furthermore, Qwest doesn't have Denver peering with the "rest" of the 
Internet.  To leave the Qwest network, connections typically go to DAL, 
LAX, or SFO on congested circuits.

So beware of VoIP over DSL.  Your users need to be aware of the 
tradeoffs between the cost of DSL vs. T1 and the effect on call quality.

Chris, if your customers are in the western US then please contact me 
about dedicated circuits.

Chris Bagnall wrote:
> Greetings list,
> 
> We have an issue with call quality at 2 sites where the users (4 Elmeg
> IP290s at one site, 2 SPA942s at the other) do not have an asterisk box
> on-site. Each site has an 8mb down/448k up ADSL connection and the phones
> connect via SIP to an asterisk box in a datacentre using g729.
> 
> The asterisk box in the datacentre connects to our other asterisk boxes
> providing pstn connectivity via IAX2. Latency between these boxes is between
> 1 and 2ms. The ADSL connections to the client sites are all consistently
> delivering latencies of sub-25ms to the datacentre and there is traffic
> shaping on that connection to give priority to any traffic from the phones'
> IPs.
> 
> Comments from the users at these sites are as follows:
> "call sounded like a dalek and I couldn't make out anything the caller was
> saying"
> "the phone on my desk is breaking up so badly it's virtually unusable"
> "calls sound like they're breaking up with metallic background noises"
> 
> We have quite a few customers with asterisk boxes on-site (with phones
> connected to them via the LAN) using ADSL connections from the same
> supplier, and are not having these issues with them.
> 
> canreinvite=no and nat=yes are set on all these devices, since they are
> behind NAT. Each device re-registers with asterisk every 5 minutes to
> prevent any possible NAT state timeouts.
> 
> Any pointers/places to look for potential problems would be much
> appreciated.
> 
> Regards,
> 
> Chris