08-20-2012 07:42 PM - edited 03-16-2019 12:47 PM
Hi Guy’s,
We have a branch of approx 10 6921 phones (connected to subscriber via DSL WAN) that are experiencing poor voice quality at a particular time in the afternoon like clock work….
Below is a stream capture of a problem call the I managed to re-create where the caller could hear me fine, however the quality of my audio became un-bearable….
I know that the MOS is quite low but what else should we be looking at?
Remote Address | = | 10.195.0.109/16434 |
Local Address | = | 10.210.42.60/23512 |
Start Time | = | 16:23:07 |
Stream Status | = | Not Ready |
Host Name | = | SEPC46413FE147E |
Sender Packets | = | 7300 |
Sender Octets | = | 145980 |
Sender Codec | = | G.729A |
Sender Reports Sent | = | 3 |
Sender Report Time Sent | = | 16:19:27 |
Rcvr Lost Packets | = | 6 |
Avg Jitter | = | 13 |
Rcvr Codec | = | G.729A |
Rcvr Reports Sent | = | 3 |
Rcvr Report Time Sent | = | 16:19:33 |
Rcvr Packets | = | 7290 |
Rcvr Octets | = | 145800 |
Cumulative Conceal Ratio | = | 0.2984 |
Interval Conceal Ratio | = | 0.0099 |
Max Conceal Ratio | = | 0.794 |
Conceal Secs | = | 129 |
Severely Conceal Secs | = | 115 |
Latency | = | 0 |
Max Jitter | = | 297 |
Sender Size | = | 20 |
Sender Reports Received | = | 9 |
Sender Report Time Received | = | 16:19:29 |
Rcvr Size | = | 20 |
Rcvr Discarded | = | 0 |
Rcvr Reports Received | = | 0 |
Rcvr Report Time Received | = | 0:00:00 |
MOS LQK | = | 2.8057 |
Avg MOS LQK | = | 2.0769 |
Min MOS LQK | = | 2 |
Max MOS LQK | = | 2.9192 |
MOS LQK Version | = | 0.95 |
Any advise much appreciated!
Cheers!
08-22-2012 03:28 AM
Jitter is a problem here (as well as the packet loss) That maximum jitter of 297 is huge. You should be aiming for jitter to be a fraction of that.
Jitter is caused when packets are delayed.
Check that QoS is correctly configured on your DSL link. If this is happening like clock-work, it could be that someone/thing is sending/receiving large IP packets, and these are delaying the RTP packets. You need to make sure that your DSL interface is configured to correctly fragment these large IP packets so as not to delay the RTP packets.
GTG
08-22-2012 02:43 PM
Thanks GR!
But isn't the jitter value in milliseconds? therefore 297 ms should be ok if the 6921 has a jitter buffer between 10 milliseconds (ms) to 1000 ms.
I am off the track here?
Cheers!
08-23-2012 12:09 AM
The point of a jitter buffer is to smooth out the variation in transmition time/delay of packets. Sure, you can have a 1000ms jitter buffer. But that would mean that the phone has to store 1000ms of voice data before playing it back to the person on the phone. If both ends of the VoIP call have 1000ms jitter buffers, that means it would take over 2 seconds for the person at the other end to respond to a simple "Hello".
If memory serves, you want jitter to be less than 40ms. So your max jitter of just under 300ms is way too high.
Once packets start arriving outside the jitter buffer window, the phone treats the packet as lost, and so either has to guess what the contents might have been, or just give silence. This is when quality starts falling off a cliff.
GTG
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