Richard:
Thank you for your question.
Out of Order Packets can cause voice quality problems in the audio path destined to the ADTRAN unit. The quality problems can be short cut-outs in audio or more severe such as no inbound audio at all.
Below is a sample output from “show voice quality-stats”
Start Out of Discard Delay
ID Time From To Length Codec Order Pkts* Avg Max
--------------------------------------------------------------------------------------------
5989 12:12 PM 5082 3650 4:56 G711u 7 0 50 50
6003 12:14 PM 5080 3123 1:11 G711u 2 0 50 100
6004 12:14 PM 1002 3665 0:38 G711u 0 0 60 90
6005 12:15 PM 3665 8205 0:20 G711u 0 0 56 80
6006 12:15 PM 3665 1002 0:22 G711u 0 0 70 92
6007 12:15 PM 8203 5080 0:59 G711u 0 0 50 50
6008 12:15 PM 3665 5081 1:35 G711u 2 0 51 60
6009 12:15 PM 3669 5562 0:22 G711u 2 0 65 85
6010 12:15 PM 8998 1002 0:25 G711u 0 0 50 50
ID
The local call-id used internally by the switchboard to differentiate call legs.
Start Time
The starting time of the call based on the system clock.
From
The calling party number.
To
The called party number.
Length
The length of time the call is connected.
Codec
The voice codec selected for the call.
Out of Order Packets
An out-of-order packet is counted if the sequence number received is greater than the sequence number expected; such a packet will count as out-of-order AND will also be counted as discarded if it cannot be played out in time. Common causes for these packets are QoS and voice traffic delivery problems to the ADTRAN or duplicate rtp streams destined for the same endpoint in the ADTRAN. In the case of duplicate rtp streams, “ip rtp symmetric filter” is explained at the end of this answer.
Discarded Packets
If a packet arrives before or after the window of the jitter buffer (based on timestamp), it will be considered an early or a late arrival. All early and late arrivals will are logged as discarded packets and will not be played out. If the jitter buffer is set to adaptive and the IAD continues to log packets outside of the jitter window, the DSP will increase the window size in 10ms increments.
Delay (Avg Max)
The Avg and Max delay indicate the size of the jitter buffer. By default, the jitter buffer in the IAD is set to adaptive with a nominal depth of 50ms. If the IAD continues to log packets outside of the 50ms window, the jitter buffer depth will increase up to a max of 100ms by default. The nominal and max jitter buffer size can be configured per FXS/SIP user or E&M/PRI trunk.
TA912(config-9001)#rtp packet-delay nominal 50
TA912(config-9001)#rtp packet-delay maximum 100
The jitter window can also be set to a fixed value by changing the delay mode to “fixed” and configuring the nominal packet delay to the desired window size.
TA912(config-9001)#rtp delay-mode fixed
TA912(config-9001)#rtp packet-delay nominal 150
Note:
ip rtp symmetric-filter
This helps in instances when there are multiple rtp streams with the same ip and port destined for the same endpoint in the ADTRAN. This can happen if a call, that is currently inactive, is still active somewhere in the network. The out-of-order packets will increment in this case. The end result can be 1-way audio.
If you have any additional questions, please let me know.
Geoff
