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Administering Avaya G430 Branch Gateway
Table of Contents
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Detailed CLI output per RTP session
Last Updated : Mar 24, 2020 |
Prolog information
Metadata
Keywords:
The following table describes the fields in the show rtp-stat detailed command output according to the numbered labels in the example.
|
Field
|
Label
|
Description
|
From the CLI example
|
|---|---|---|---|
|
Session-ID
|
1
|
An arbitrary index number for the session in the session table
|
Session-ID: 35
|
|
Status
|
2
|
The status of the session. Possible values:
|
Status: Terminated
|
|
QOS
|
3
|
The QoS status of the session. Possible values:
|
QOS: Faulted
|
|
EngineId
|
4
|
The ID of the VoIP engine. The Branch Gateway has one VoIP engine.
|
EngineId: 0
|
|
Start-Time
|
5
|
The date of the RTP session
|
2015-08-23
|
|
6
|
The start time of the RTP session
|
Start-Time: 2015-08-23,11:09:07
|
|
|
End-Time
|
7
|
The end time of the RTP session
|
End-Time: 2015-08-23,11:13:40
|
|
Duration
|
8
|
The duration of the RTP session
|
Duration: 00:04:33
|
|
CName
|
9
|
format: gwt@<MGP-address>
|
CName: gwp@135.8.118.252
|
|
Phone
|
10
|
The local extension number and conference ID in format <conference ID>:<extension number>.
Conference calls can involve more than one entry in the session table. Multiple sessions belonging to the same conference call can usually be identified by a common conference ID.
Notes:
|
Phone: 69:2011
|
|
Local-Address
|
11
|
The PMI. The number after the colon is the UDP port number.
|
Local-Address: 135.8.118.252:2061
|
|
Remote-Address
|
13
|
The remote VoIP engine, gateway PMI, or IP phone address. The number after the colon is the UDP port number.
|
Remote-Address: 135.8.76.107:2061
|
|
12, 14
|
SSRC ID. The number in parentheses is the number of observed SSRC changes during the session.
|
SSRC 2989801899 (0)
|
|
|
Samples
|
15
|
The number of times the application has sampled the VoIP engine (RTP receiver) statistics.
|
Samples: 5415 (5 sec)
|
|
16
|
The sampling interval
|
Samples: 54 (5 sec)16
|
|
|
Codec:
|
17
|
The codec used for the session
|
G723
|
|
18
|
The RTP packet size, in bytes
|
62B
|
|
|
19
|
The RTP packet interval, in ms
|
30mS
|
|
|
20
|
The encryption method
|
Off
|
|
|
Silence suppression (Tx/Rx)
|
21
|
The received silence suppression method
|
Silence-suppression(Tx/Rx) Disabled21/Not-Supported
|
|
22
|
The transmitted silence suppression method
|
Silence-suppression(Tx/Rx) Disabled/Not-Supported22
|
|
|
Play-Time
|
23
|
The overall time the codec played valid received frames
|
Play-Time 272.610sec
|
|
Codec
Loss codec-loss%
|
24
|
The last value of codec loss sampled. Codec loss is the percentage of time the codec played fill frames due to lack of valid RTP frames. Possible causes include jitter and packet loss.
|
Loss 0.0%24 #1
|
|
#codec-loss-events
|
25
|
The codec loss event counter
|
Loss 0.0% #125
|
|
Avg-Loss
|
26
|
The average of all codec loss values sampled during the session
|
Avg-Loss 0.1%
|
|
RTT rtt ms
|
27
|
The last sampling of codec round trip time (RTT), in ms. Codec RTT is the round-trip delay experienced by the user, including internal delay. This value is not entirely accurate since remote internal delays are not always known.
|
RTT 741mS27 #38
|
|
#rtt-events
|
28
|
The codec RTT event counter
|
RTT 741mS #3828
|
|
Avg-RTT
|
29
|
The average of all codec RTT values sampled during the session
|
Avg-RTT 570mS
|
|
Jbuf-under/ overruns
|
30
|
The estimated percentage contribution of jitter-buffer underruns to the average codec loss
|
JBuf-under/overruns 0.1%30/0.0%
|
|
31
|
The estimated percentage contribution of jitter-buffer overruns to the average codec loss
|
JBuf-under/overruns 0.1%/0.0%31
|
|
|
Jbuf-delay
|
32
|
The last jitter buffer delay
|
Jbuf-Delay 22mS
|
|
Max-Jbuf-Delay
|
33
|
The maximum jitter buffer delay during the session
|
Max-Jbuf-Delay 60mS
|
|
Received RTP:
|
|||
|
Packets
|
34
|
The total number of received packets
|
Packets 9236
|
|
Loss
loss%
|
35
|
The last sampled value of network RTP packet loss
|
Loss 0.0%35 #0
|
|
#loss-events
|
36
|
The network RTP packet loss event counter
|
Loss 0.0% #036
|
|
Avg-loss
|
37
|
The average of all network RTP packet loss values during the session
|
Avg-Loss 0.0%
|
|
RTT rtt ms
|
38
|
The network RTT. The RTT is calculated upon RTCP packet reception.
|
RTT 604mS38 #38
|
|
#rtt-events
|
39
|
The network RTT event counter
|
RTT 604mS #3839
|
|
Avg-RTT
|
40
|
The average of all network RTT values during the session
|
Avg-RTT 376mS
|
|
Jitter
jitter ms
|
41
|
The network jitter at the RTP receiver.
Combined with long RTT, a large jitter value may indicate WAN congestion.
|
Jitter 0mS41 #0
|
|
#jitter-event
|
42
|
The RTP receiver network jitter event counter
|
Jitter 0mS #042
|
|
Avg-Jitter
|
43
|
The average of all network jitter values during the session
|
Avg-Jitter 0mS
|
|
TTL (last/min/max)
|
44
|
The last value of TTL, minimum value of TTL, and maximum value of TTL sampled during the session. TTL changes during a session may indicate route flaps in the IP network.
|
TTL(last/min/max) 63/63/63
|
|
Duplicates
|
45
|
This counter increments each time two consecutive RTP packets with the sample RTP sequence number are received. A large number of duplicates may indicate problems in the Layer 2/Ethernet topology (for example, loops).
|
Duplicates 0
|
|
Seq-Fall
|
46
|
This counter increments each time an RTP packet with a sequence number less than the last known sequence is received. Packet resequencing may be caused by switching to a backup WAN interface or route flaps.
|
Seq-Fall 0
|
|
DSCP
|
47
|
The last received DSCP value of the RTP packets
|
DSCP 46
|
|
L2Pri
|
48
|
The last received Layer 2 priority value of an RTP packet (usually IEEE802.1p)
|
L2Pri 12
|
|
RTCP
|
49
|
The total number of received RTCP packets
|
RTCP 54
|
|
Transmitted-RTP:
|
|||
|
VLAN
|
50
|
The VLAN-ID on which the RTP packets are transmitted
|
VLAN 1
|
|
DSCP
|
51
|
The DSCP of RTP packets
|
DSCP 184
|
|
L2Pri
|
52
|
The Layer 2 priority of transmitted RTP packets (usually 802.1p)
|
L2Pri 6
|
|
RTCP
|
53
|
The total number of transmitted RTCP packets
|
RTCP 62
|
|
Remote-Statistics:
(Remote-Statistics items are calculated and evaluated upon reception of RTCP messages)
|
|||
|
Loss
rem-loss%
|
54
|
The network loss experienced by the remote RTP receiver. The local RTP receiver learns about its remote peer statistics from RTCP packets.
|
Loss 0.0%54 #0
|
|
#rem-loss-ev
|
55
|
The number of samples that were over the rem-loss threshold
|
Loss 0.0% #055
|
|
Avg-Loss
|
56
|
The average network loss experienced by the remote RTP receiver
|
Avg-Loss 0.0%
|
|
Jitter
rem-jitter
|
57
|
The network jitter experienced by the remote RTP receiver
|
Jitter 0mS57 #0
|
|
#rem-jitter-ev
|
58
|
The number of samples that were over the remote jitter threshold
|
Jitter 0mS #058
|
|
Avg-jitter
|
59
|
The average remote jitter
|
Avg-Jitter 0mS
|
|
Echo Cancellation:
|
|||
|
Loss
loss dbm
|
60
|
The echo cancellation loss on the TDM bus. A high value (that is, a low absolute value) may indicate impairment of DCP terminals.
|
Loss 45dB60 #1
|
|
#loss-ev
|
61
|
A counter that increments each time the echo-cancellation loss is sampled below its threshold
|
Loss 45dB #161
|
|
Len
|
62
|
The last echo-cancellation tail length used for this session
|
Len 32mS
|
|
RSVP:
|
|||
|
Status
|
63
|
The current (last) RSVP reservation state at the end of the session
|
Status Disabled
|
|
Failures
|
64
|
The total number of reservation failures during the session
|
Failures 0
|