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Add_workflow_for_techpubs
AUTH1UserResourceIdentifier{userKey=8a00a02355cd1c2f0155cd26cb41059c, userName='null'}
JIRAIDAUTHCHOR-10543667
REV5UserResourceIdentifier{userKey=8a00a0c85b2726c2015b58aa779d0003, userName='null'}
REV6UserResourceIdentifier{userKey=8a00a0c85b2726c2015b58aa779d0003, userName='null'}
REV3REV1UserResourceIdentifier{userKey=8a00a02355cd1c2f0155cd26c99e02c0, userName='null'}
REV1UserResourceIdentifier{userKey=8a00a02355cd1c2f0155cd26cef20cbf, userName='null'}

To create To create or modify an Entry to a Call Routing Table:

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Call Routing Entry - Field Definitions

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Admin State

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Specifies the admin state of the Call Route. Valid entry: Enable (enables the call route entry for routing the call, displays in configuration header as  ) or Disable (disables the call route entry from being used, displays in the configuration header as ).

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Specifies the Transformation Table to be used for this routing entry. This drop down list is populated from the entries in the Transformation Table.

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Time of Day Restriction:

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Select an optional time of day table to use with the call route. All Time of Day tables are included in this drop down list. For details on Time of Day, refer to Working with Time of Day.

Destination Type

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Specifies the destination type for calls using this route. Valid selections: Specifies the destination type for calls using this route. Valid selections: Normal, Registrar Table, Deny, or Trunk Group.

Normal. Call routes to normal types such as ISDN or SIP signaling groups. Specify a list of signaling groups through Destination Signaling Groups.

Registrar Table. Call routes to a signaling group that contains the registrar table.

Deny. Call routes to a specific Q.850 cause code are rejected. Through the Deny Q.850 Cause Code field, select the specific Q.850 Cause code. When Deny is selected, the Deny Q.850 Cause Code field is displayed.

Trunk Group. Calls are routed to an incoming trunk group destination using the associated signaling group. This routing entry should be selected in order to route calls to a trunk group. When Trunk Group is selected, Fork Call and Destination Signaling Groups options are not available.

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Note

Available for SBC SWe only.

Info

SIP Media Session and SIP Proxy Media Encryption Signaling Sessions and Enhanced Media Sessions without Transcoding licenses are required for this functionality.

The Proxy SRTP Handling option applies SRTP media encryption to a call; this enables communication between end points that are SRTP incompatible. For example, this enables communication if both Calling and Called user agents both support G.711u codec, but they differ in SRTP capability, or one party in the call does not support SRTP at all. Valid entries: Relay or Local. Default: Relay.

This field is displayed only when Proxy, DSP preferred over Proxy, or Proxy preferred over DSP is selected from the Audio Stream Mode drop down list.

Relay: When Relay is selected, the following options are supported:

  • For Proxy. Proxy or switches the media stream between endpoints.
  • For DSP Preferred over Proxy. Uses DSP resources, rather than proxy or switch the media stream between endpoints.
  • Proxy deferred over DSP. Proxies the media stream between endpoints rather than choosing DSP mode.

Local: When Local is selected, SRTP is negotiated automatically for the following: 

  • Proxy local SRTP. Proxies or switches the media stream between endpoints using SRTP media encryption on a call leg basis.
  • DSP Preferred over Proxy local SRTP. Uses DSP resources in addition to SRTP media encryption, rather than proxy or switch the media stream between endpoints..
  • Proxy Local SRTP over DSP. Proxies the media stream between endpoints rather than choosing DSP mode and uses SRTP media encryption).

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Specifies whether or not to use media transcoding.

Media Transcoding requires a specific Transcoding License (Enhanced Media Sessions with Transcoding). Do not enable Media Transcoding unless your calling configuration requires it and 

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is licensed for the Transcoding feature.

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  • If Media Transcoding is enabled without a Transcoding license (Enhanced Media Sessions with Transcoding), a critical alarm is generated regardless of whether or not the routed calls require transcoding.
  • Transcoding must be enabled on SIP-to-SIP calls. 

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Specifies the Media List to use for this call route. This drop down list is populated with the Media List entries created through the Create Media list option. See Creating and Modifying Media Lists.

If the Media List configuration selected, then the Destination Signaling Group would be selected that has the common media set available. The media order from the call route's media list takes precedence over the Signaling Group's media list. Generally this field should be kept the default value "None" unless the media codec selection has to be controlled and manipulated for this route.

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Numberofcalls
Quality Metrics Number of Calls

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Specifies the number of calls over which the quality metrics are calculated.

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Retry
Retry
Quality Metrics Time Before Retry

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Specifies the period of time in minutes after which a route is tried again after failing quality metrics.

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Specifies the minimum answer/seizure ratio for this rule to be considered for use.

here about Answer Seizure Ratio
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The ASR is a measure of network quality defined by the ITU. The answer/seizure ratio (ASR) is a measurement of network quality and call success rate in telecommunications. It is the percentage of answered telephone calls with respect to the total call volume.

The answer/seizure ratio is defined as 100 times the ratio of sucessfully answered calls divided by the total number of call attempts (seizures).

Busy signals and other rejections by the called number count as call failures. This makes the ASR highly dependent on end-user action or behavior and is out of control by the telecommunications carrier. Low ASR values may be caused by far-end switch congestion, not answering by called parties and busy destination lines.

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parties and busy destination lines.

Enable Min MOS Threshold

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This field enables the Min. MOS score feature (call routes checked for voice quality) to be used. Valid entries: Enabled (enables the Min. MOS Score feature) or Disabled (disables Min. MOS Score feature)

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Min. MOS Score

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The Min. MOS score provides an option to allow the SBC to temporarily enable/disable a route that is having bad voice quality.

Two options are available:

Enabled. A MOS score for quality purposes is added to each call using that specific route. The Quality Metrics Number of Calls field is used to determine the average acceptable value; if the average of acceptable calls falls below, it is considered bad voice quality and the route is then temporarily disabled for calls. The Quality Metrics Time Before Retry field is used when the route is disabled for failing metrics to determine how long before the route is retried.

Disabled. Min. MOS Score feature is disabled and all call routes are enabled.

Enable Max. R/T Delay

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Specifies whether or not to use Round Trip Delay.

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In telecommunications, the round-trip delay time (RTD) or round-trip time (RTT) is the length of time it takes for a signal to be sent plus the length of time it takes for an acknowledgment of that signal to be received. This time delay therefore consists of the transmission times between the two points of a signal.

The RTT was originally estimated in TCP by: RTT = (? - Old_RTT) + ((1 ? ?) - New_Round_Trip_Sample
Where ? is constant weighting factor(0 ? ? < 1). Choosing a value ? close to 1 makes the weighted average immune to changes that last a short time (e.g., a single segment that encounters long delay). Choosing a value for ? close to 0 makes the weighted average respond to changes in delay very quickly.

This was improved by the Jacobson/Karels algorithm, which takes standard deviation into account as well. Once a new RTT is calculated, it is entered into the equation above to obtain an average RTT for that connection, and the procedure continues for every new calculation.

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Specifies whether or not Jitter will be considered as a quality metric for this Call Route.

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Jitter is the undesired deviation from true periodicity of an assumed periodic signal in electronics and telecommunications, often in relation to a reference clock source. Jitter may be observed in characteristics such as the frequency of successive pulses, the signal amplitude, or phase of periodic signals. Jitter is a significant, and usually undesired, factor in the design of almost all communications links (e.g., USB, PCI-e, SATA, OC-48).

Jitter can be quantified in the same terms as all time-varying signals, e.g., RMS, or peak-to-peak displacement. Also like other time-varying signals, jitter can be expressed in terms of spectral density (frequency content).

Jitter period is the interval between two times of maximum effect (or minimum effect) of a signal characteristic that varies regularly with time. Jitter frequency, the more commonly quoted figure, is its inverse. ITU-T G.810 classifies jitter frequencies below 10 Hz as wander and frequencies at or above 10 Hz as jitter.

Jitter may be caused by electromagnetic interference (EMI) and crosstalk with carriers of other signals. Jitter can cause a display monitor to flicker, affect the performance of processors in personal computers, introduce clicks or other undesired effects in audio signals, and loss of transmitted data between network devices. The amount of tolerable jitter depends on the affected application.

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