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. [EMAIL PROTECTED] _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
