[Asterisk-Users] SPA-841 "Decode Latency"?

[alan]

We're investigating audio quality issues in our system; maybe someone
can help. We're using Asterisk as a basic PBX, with a single PRI on one
side and SIP phones on the other: Sipura SPA-841's.

We're experiencing several audio effects which seem to commonly
correspond to network failures (packet loss, high jitter, etc manifested
as "robot voice", dropouts, periodic buzzing, etc). However, all the
network monitoring we're doing on our fully switched, underutilized
100baseT-FD network shows that we have sub-1ms ping times and no jitter
to speak of.

Looking at the SPA-841's main Status page, I see call status:
Line State:	Connected	Tone:	None
Encoder:	G711u		Decoder:	G711u
Type:		Inbound		Remote Hold:	No
Callback:			Peer Name:	xxxxxxxxxx
Peer Phone:	xxxxxxxxxx	Duration:	00:09:53
Packets Sent:	29545		Packets Recv:	29666
Bytes Sent:	4727360		Bytes Recv:	4746560
Decode Latency:	50 ms		Jitter:	0 ms
Round Trip Delay:	0 ms	Packets Lost:	0
Packet Error:	0		Mapped RTP Port:	16396 >> 0

I have not yet seen Jitter above 2ms, or significant packet loss; round
trip delay is always 0. But periodically, "Decode Latency" will spike up
to the 150-300ms range. This seems to correspond to audio effects such
as a periodic "bzzt" sound in the handset.

Does anyone have any familiarity with "decode latency," specifically
with Sipura devices? Why would it take 200+ms to decode a 20ms RTP
packet? G.711u has existed for over 30 years, how hard could it be?

Thanks,

Alan Ferrency
pair Networks, Inc.
alan at pair.com