ZHONE-COM-VOICE-DSP-MIB

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Organization: Zhone Technologies, Inc.

Last Updated: 2003-03-28

Category: Zhone Technologies CPE and Access

Description: Monitors Zhone Technologies voice DSP channel utilization, codec configuration, and voice quality metrics.

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What Is ZHONE-COM-VOICE-DSP-MIB?

ZHONE-COM-VOICE-DSP-MIB is a Zhone Technologies MIB for monitoring Voice Digital Signal Processing (DSP) resources on Zhone's voice application-specific processing cards, used in access/gateway platforms carrying VoIP or TDM voice traffic. It exposes objects for per-channel DSP resource utilization, codec configuration, and voice-quality metrics. Its central monitoring value is hardware/software resource health of the voice cards, covering DSP channel utilization, codec settings in use, and voice-quality indicators, letting operators detect DSP resource exhaustion or degraded call quality before subscribers complain. It is a vendor extension likely paired with Zhone's broader chassis/card inventory MIBs for slot-level context. It is deployed on Zhone (and successor DASAN Zhone) multiservice access platforms delivering voice services over DSP-based line cards. Engineers can download the ZHONE-COM-VOICE-DSP-MIB file directly to load it into their MIB browser.

IPNetwork Monitor allows you to monitor SNMP objects defined in ZHONE-COM-VOICE-DSP-MIB. Use the built-in SNMP Monitoring Browser to explore available variables, view their current values and descriptions, and select the objects you want to monitor. You can then create SNMP monitors, configure thresholds, and receive alerts when monitored values change.

Supported Devices

Zhone Technologies voice DSP/gateway cards
multiservice access platforms

Monitoring Examples

This MIB's sample object list is empty in the source data, so specific OID names cannot be cited here. Based on the module description, an operator would typically poll per-channel DSP utilization and voice-quality metrics on a voice processing card to detect when DSP resources are near capacity. A degraded voice-quality reading on a specific channel would point to a card- or channel-level problem needing attention.

What Can Be Monitored

DSP channel utilization
codec configuration
voice quality metrics

Imported Objects

From SNMPv2-SMI

Counter32
Integer32
MODULE-IDENTITY
NOTIFICATION-TYPE
OBJECT-IDENTITY
OBJECT-TYPE
TimeTicks
Unsigned32

From Zhone

zhoneModules	OBJECT-IDENTITY
zhoneShelfIndex	OBJECT-TYPE
zhoneSlotIndex	OBJECT-TYPE
zhoneVoice	OBJECT-IDENTITY

From Zhone-TC

ZhoneShelfValue
ZhoneSlotValue

How to Use in IPNetwork Monitor

Example using voiceDspResetCount OID:

Select a Zhone Technologies voice DSP/gateway card, multiservice access platform (per-channel DSP-utilization/voice-quality status) as the target host to create a monitor — the SNMP service should be up and running on it. Click New Monitor, then check SNMP Custom on the Favorites tab, click Next, and confirm the host. On the next page, click Select... to open the built-in SNMP MIB Browser and type voiceDspResetCount into the Find box to locate it in the OID tree, selecting the specific row/instance you want to monitor since this is a table column, then select it and click OK. It reports the number of slave DSP resets that have occurred within a voice DSP. On the monitor's Main parameters page you can set the target's SNMP port (default 161), credentials, polling interval, and other settings — see the SNMP Monitor help for details. On the State conditions and Alerting tabs, configure when the monitor should change state and trigger an alert; since this is a Counter32-type OID, Value bounds is the most useful condition here — trigger an alert if the counter increases sharply between polls relative to its normal baseline, since an unexpected spike often reflects a real change in traffic or activity. Click Finish to create the monitor; you can adjust any parameter later.

OIDs

OID symbolic	OID numeric	Type	Access	Description
zhoneVoiceDsp	1.3.6.1.4.1.5504.4.3.3			The MIB module to describe managed objects for Zhone's voice DSP applications. The Voice DSP MIB has the following components: * voiceDspDefaultConfiguration * voiceDspStatusTable * channelStatusTable * voiceDspTraps
voiceDspDefaultConfiguration	1.3.6.1.4.1.5504.4.3.3.1			System-wide defaults for voice DSP variables. These values will be used by voice DSP to set up default values to be used per call.
INT redundancyOverSubscriptionType	1.3.6.1.4.1.5504.4.3.3.1.1	INTEGER	read-write	voice DSP redundancy over-subscription model. It is important to know the concept of active versus standby resources in order to understand how over- subscription works. For each call that is made on our equipment, resources are used. A resource is active if it is currently in use and it is standby if it is being held in case of the active's failure. If an active resourse fails, the standby can take over and the call continues. On the standby resources in a redundancy configuration can be oversubscribed. A service provider system administrator can configure a voice DSP system to use one of the following oversubscription models: * none: implies 1:1 (one to one) oversubscription. When a standby is allocated, the pool of available actives is depleted by one. * low: implies 1:2 oversubscription. * medium: implies 1:4 oversubscription. * high: implies infinite oversubscription. Active pool is never depleted when any standbys are allocated. If all active resources are used for real calls, no standby resources will be available to take over a failed call. The default value of this variable is high. This variable is supported only on BAN with VASP card(s).
INT jitterBufferType	1.3.6.1.4.1.5504.4.3.3.1.2	INTEGER	read-write	Voice DSP channel jitter buffer defaults. There are two types of jitter algorithms: static and dynamic. Dynamic allows the jitter buffer to grow and shrink as inter-arrival jitter changes. A static jitter buffer does not. Some examples of the voice connection types used for; Voice Over ATM (VoATM): pots-to-aal2, isdn-to-aal2, aal2-to-gr303, aal2-to-v52. Voice Over IP (VoIP) : sip-to-ds1, sip-to-gr303, sip-to-v52. Dynamic jitter buffer is not supported in VoATM case. Default jitter buffer type is static for VoATM, and dynamic for VoIP.
U32 jitterBufferSize	1.3.6.1.4.1.5504.4.3.3.1.3	Unsigned32	read-write	Voice DSP jitter buffer size in milliseconds. More specifically, it refers to the amount of memory usage that can store certain milliseconds of voice. There are two types of jitter algorithms: static and dynamic. Dynamic allows the jitter buffer to grow and shrink as inter-arrival jitter changes. A static jitter buffer does not. When the jitterBufferType is dynamic, jitterBufferSize is the initial buffer size the jitter buffer can grow to. Some examples of the voice connection types used for; Voice Over ATM (VoATM): pots-to-aal2, isdn-to-aal2, aal2-to-gr303, aal2-to-v52. Voice Over IP (VoIP) : sip-to-ds1, sip-to-gr303, sip-to-v52. Valid range is 1 to 33 milliseconds for VoATM case. Valid range is 1 to 100 milliseconds for VoIP case. Default jitter buffer size is 10 milliseconds.
U32 interArrvJitThreshold	1.3.6.1.4.1.5504.4.3.3.1.4	Unsigned32	read-write	The inter-arrival jitter threshold per channel used as a trigger value in milliseconds to generate a trap when its value is passed. If this value is 0, the threshold mointoring is disabled for inter-arrival jitter. Valid range is 0 to 100 milliseconds. The default value is 80 milliseconds. This variable is not supported.
U32 pktsLostThreshold	1.3.6.1.4.1.5504.4.3.3.1.5	Unsigned32	read-write	Voice DSP channel packets lost per minute threshold. A trap will be generated when this value is passed. If this value is 0, the threshold mointoring is disabled for packets lost. Valid range for this value is from 0 to 10000 pkts/ min. The default value is 600 pkts/min. This variable is not supported.
INT echoCancellationType	1.3.6.1.4.1.5504.4.3.3.1.6	INTEGER	read-write	This data informs which echo cancellation algorithm to be used by default: 165 or 168 for a particular channel. The numbers 165 and 168 are the ITU designations for the algorithms. The number following the designation describes the echo tail length in milliseconds. Both G.165 and G.168 are ITU-T recommendations. G.168 has slightly stricter technical specs. G.168 defines much better how the tests are performed. If an echo canceller passes the G.165 spec, then it is G.165 compliant. If it passes G.168, it is both G.168 and G.165 compliant. The echo tail length required depends on factors such as network configuration, local loop length, line frequency response, etc. Generally, 16 ms is more than adequate for local loop (ie. Zedge) and 32 ms is a good minimum for Network side. Our products currently support up to 48 ms of echo tail length, MALC supports 16 ms of echo tail length. Valid values for the variable are: * off - echo cancellation is off (on all platforms). * g165EchoTL16 - G.165 echo cancellation with echo tail length of 16ms (on MALC only). * g168EchoTL48 - G.168 echo cancellation with echo tail length of 48ms (on the other platforms, i.e.: BAN, Sechtor 100, Z-Edge64). Following values are not supported: * g165EchoTL32 - G.165 echo cancellation with echo tail length of 32ms. * g165EchoTL48 - G.165 echo cancellation with echo tail length of 48ms. * g168EchoTL32 - G.168 echo cancellation with echo tail length of 32ms. * g168EchoTL64 - G.168 echo cancellation with echo tail length of 64ms. * g168EchoTL80 - G.168 echo cancellation with echo tail length of 80ms. * g168EchoTL96 - G.168 echo cancellation with echo tail length of 96ms. * g168EchoTL112 - G.168 echo cancellation with echo tail length of 112ms. * g168EchoTL128 - G.168 echo cancellation with echo tail length of 128ms. The default value is g165EchoTL16 for MALC, and g168EchoTL48 for the other platforms.
INT silenceSuppressionType	1.3.6.1.4.1.5504.4.3.3.1.7	INTEGER	read-write	This data informs a voice DSP what type of silence suppression algorithm to use on a channel. Silence suppression is used to suppress packet generation during periods of silence of a voice call. The SID (silence descriptor) frame is generated at the start of a silence period, or periodically, and it is used to characterize the power level of the background noise during silence period on the encode side of the voice path. The SID frame is then passed to the decode side through the packet encoding medium (AAL2 or RTP). The decode side then generates comfort noise at an equivalent power level dictated by the SID frame values. Valid values for the variable are: * silSupOff - silence suppression is off. * silSupOnSidOn - send an SID frame at a specified constant interval (e.g. 100ms). Following values are not supported: * silSupOnSidOff - discontinue transmission during silence periods and no SID frame is sent. * silSupOnSidConst - send an initial SID frame and then discontinue transmission. Silence suppression is not supported on MALC. The default value is silSupOff.
INT echoReturnLoss	1.3.6.1.4.1.5504.4.3.3.1.8	INTEGER	read-write	The amount of loss between the transmitted signal and the reflected echo back from the hybrid where the 4-to-2 wire conversion takes place. For MALC: 0dB and 6dB. 6dB is default. For all other platforms: 0dB, 3dB, 6dB. 3db is default.
voiceDspStatusTable	1.3.6.1.4.1.5504.4.3.3.2		not-accessible	Current status information for the voice DSP application. This table gives a current view of how a voice DSP is doing since the voice DSP is started.
voiceDspStatusEntry	1.3.6.1.4.1.5504.4.3.3.2.1		not-accessible	An entry in the voiceDspStatusTable.
U32 voiceDspMaxChannelOnBoard	1.3.6.1.4.1.5504.4.3.3.2.1.1	Unsigned32	read-only	Maximum number of channels a voice DSP can handle at the present time. The valid range for DSP boards we currently support is 0 to 1008.
U32 voiceDspActiveChannelCount	1.3.6.1.4.1.5504.4.3.3.2.1.2	Unsigned32	read-only	Number of channels currently active. The maximum value of this variable should not exceed voiceDspMaxChannelOnBoard.
U32 voiceDspHiWaterChannelCount	1.3.6.1.4.1.5504.4.3.3.2.1.3	Unsigned32	read-only	The highest channel count since this voice DSP has been running. The maximum value of this variable should not exceed voiceDspMaxChannelOnBoard.
C32 voiceDspResetCountEx	1.3.6.1.4.1.5504.4.3.3.2.1.4	Counter32	read-only	Number of slave DSP resets that have occurred within a voice DSP. A reset occurs when a slave DSP crashes. This counter is incremented each time voice DSP detects a slave DSP crash.
channelStatusTable	1.3.6.1.4.1.5504.4.3.3.3		not-accessible	This table gives a detailed snapshot view of all the voice calls the voice DSP is currently processing. Only the current active calls are reported in this table and there is no information about call sessions that are already terminated.
channelStatusEntry	1.3.6.1.4.1.5504.4.3.3.3.1		not-accessible	An entry in the channelStatusTable.
I32 channelId	1.3.6.1.4.1.5504.4.3.3.3.1.1	Integer32	read-only	Logical channel id of a call. The valid range of this id is from 1 to 65536. At any single instance, there should at most one call with the same channel id but over time the same channel id can be used for different calls.
U32 channelVoiceSessionIdHigh	1.3.6.1.4.1.5504.4.3.3.3.1.2	Unsigned32	read-only	The high 32 bits of the session number of a voice call. The session id is used to keep track all resources used by a call. Detailed call performance statistics can be supported in the future with session id uniquely identify each individual call. Voice session id is a unique 64 bit number and it will never be repeated.
U32 channelVoiceSessionIdLow	1.3.6.1.4.1.5504.4.3.3.3.1.3	Unsigned32	read-only	The low 32 bits of the session number of a voice call. The session id is used to keep track all resources used by a call. Detailed call performance statistics can be supported in the future with session id uniquely identify each individual call. Voice session id is a unique 64 bit number and it will never be repeated.
ZHO channelCcrpShelf	1.3.6.1.4.1.5504.4.3.3.3.1.4	ZhoneShelfValue	read-only	The shelf where the Call Control Resource Provider (CCRP) resides. CCRP process acts as a gateway for call signaling between TDM equipment and the ATM AAL2 network. Range of valid values is 1-255. Note 0 is an invalid value.
ZHO channelCcrpSlot	1.3.6.1.4.1.5504.4.3.3.3.1.5	ZhoneSlotValue	read-only	The slot where the Call Control Resource Provider (CCRP) resides. CCRP process acts as a gateway for call signaling between TDM equipment and the ATM AAL2 network. Range of valid values is 1-17. Note 0 is an invalid value.
ZHO channelTrunkCtrpShelf	1.3.6.1.4.1.5504.4.3.3.3.1.6	ZhoneShelfValue	read-only	The shelf where the Trunk Call Temrination Resource Provider (TrunkCtrp) resides. TrunkCtrp is a proccess through which call signaling travels to and from TDM equipment. The CCRP above does not communicate directly with TDM; it does so through this CTRP. Range of valid values is 1-255. Note 0 is an invalid value.
ZHO channelTrunkCtrpSlot	1.3.6.1.4.1.5504.4.3.3.3.1.7	ZhoneSlotValue	read-only	The slot where the Trunk Call Temrination Resource Provider (TrunkCtrp) resides. TrunkCtrp is a proccess through which call signaling travels to and from TDM equipment. The CCRP above does not communicate directly with TDM; it does so through this CTRP. Range of valid values is 1-17. Note 0 is an invalid value.
ZHO channelPktCtrpShelf	1.3.6.1.4.1.5504.4.3.3.3.1.8	ZhoneShelfValue	read-only	The shelf where the Packet Call Temrination Resource Provider (PktCtrp) resides. PktCtrp is a proccess through which call signaling travels to and from ATM AAL2 network. Range of valid values is 1-255. Note 0 is an invalid value.
ZHO channelPktCtrpSlot	1.3.6.1.4.1.5504.4.3.3.3.1.9	ZhoneSlotValue	read-only	The slot where the Packet Call Temrination Resource Provider (PktCtrp) resides. PktCtrp is a proccess through which call signaling travels to and from ATM AAL2 network. Range of valid values is 1-17. Note 0 is an invalid value.
INT channelActiveCodec	1.3.6.1.4.1.5504.4.3.3.3.1.10	INTEGER	read-only	Types of voice codecs on the current call. G.711, G.726, G.729 and G.723 are ITU-T standards for voice and multimedia modulation and digital encoding. Valid values for this variable can be: * g711Ulaw - G.711 Ulaw coding standard at 64 kbps * g711Alaw - G.711 Alaw coding standard at 64 kbps * g726 - G.726 ADPCM coding stardards at 48/56/64 kbps * g729A - G.729 Annex A CS-ACELP coding standard at 8 kbps * g729B - G.729 Annex B CS-ACELP coding standard with silence suppressionat 8 kbps * g729E - G.729 Annex E CS-ACELP coding standard with silence suppressionat 11.8 kbps * g723R1KBPS5Dot3 - G.723.1 coding standard at 5.3 kbps * g723R1KBPS6Dot3 - G.723.1 coding standard at 6.3 kbps * g723R1AKBPS5Dot3 - G.723.1 Annex A coding standard at 5.3 kbps * g723R1AKBPS6Dot3 - G.723.1 Annex coding standard at 6.3 kbps
TIK channelTimeAlive	1.3.6.1.4.1.5504.4.3.3.3.1.11	TimeTicks	read-only	This indicates how long this channel has been running in TimeTicks. A channel is considered alive when the call parties pick up the phone and data starts to flow and it is no longer alive when the phone is hung up. One TimeTick is one hundredth of a second.
C32 channelPktsSent	1.3.6.1.4.1.5504.4.3.3.3.1.12	Counter32	read-only	Number of packets transmitted.
C32 channelBytesSent	1.3.6.1.4.1.5504.4.3.3.3.1.13	Counter32	read-only	Number of bytes transmitted.
C32 channelPktsReceived	1.3.6.1.4.1.5504.4.3.3.3.1.14	Counter32	read-only	Number of packets received.
C32 channelBytesReceived	1.3.6.1.4.1.5504.4.3.3.3.1.15	Counter32	read-only	Number of bytes received.
C32 channelPktsLost	1.3.6.1.4.1.5504.4.3.3.3.1.16	Counter32	read-only	Number of packets lost.
U32 channelInterArrvJitter	1.3.6.1.4.1.5504.4.3.3.3.1.17	Unsigned32	read-only	The inter arrival jitter in milliseconds.
U32 channelJitterBufferSize	1.3.6.1.4.1.5504.4.3.3.3.1.18	Unsigned32	read-only	Size of jitter buffer in milliseconds. The jitter buffer is used to compensate for inter-arrival packet jitter. It is accomplished by keeping a buffer of packets to allow continuous playout of voice.
voiceDspTraps	1.3.6.1.4.1.5504.4.3.3.4			All voice DSP related traps are defined under voiceDspTraps.
voiceDspTrapsPrefix	1.3.6.1.4.1.5504.4.3.3.4.0			prefix 0 for voiceDspTraps as required by SNMPv2.
NTF voiceDspReset	1.3.6.1.4.1.5504.4.3.3.4.0.1			Notification that a voice DSP just reset.
NTF voiceDspChannelPktsLoss	1.3.6.1.4.1.5504.4.3.3.4.0.2			Notification to be generated when channelPktsPktsLost is more than the voice Dsp default pktsLostThreshold.
NTF voiceDspChannelInterArrvJitterTrigger	1.3.6.1.4.1.5504.4.3.3.4.0.3			Notification to be generated when channelInterArrvJitter is more than the voice Dsp default interArrvJitThreshold.
comVoiceDsp	1.3.6.1.4.1.5504.6.12			Voice DSP MIB for Voice Application Specific Processing cards.

RFC description

Voice DSP configuration and statistics for voice processing cards on Zhone telecommunications equipment.

Start monitoring Zhone Technologies voice DSP/gateway cards, multiservice access platforms (per-channel DSP-utilization/voice-quality status) with a free 30-day trial of IPNetwork Monitor. Import MIBs, browse SNMP OIDs, create custom SNMP monitors, configure alerts, and monitor any SNMP-enabled network device from a single console.

Download ZHONE-COM-VOICE-DSP-MIB