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Additional pages:
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The Use the Packet Service Profile screen enables you to create or edit a Packet Service Profile. Each Packet Service Profile is configured for a pair of gateways and includes entries for up to four audio/video encoding methods. The pair of gateways can be originating and destination gateways in the same gateway group, or can be originating and destination gateways in an inter-gateway group.
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For egress call legs over IP trunk groups, you can use the Trunk Group screen to assign a packet service profile to an egress IP trunk group.
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On the SBC main screen, go to Configuration > System Provisioning > Category: Trunk Provisioning > Packet Service Profile. or
All > Profiles > Media > Packet Service Profile
The Packet Service Profile window is displayed.
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To edit any of the Packet Service Profiles in the list, click the radio button next to the specific Packet Service Profile name.
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create a new Packet Service Profile:
Click New Packet Service Profile. The Create New Packet Service Profile window is displayed
where you can name and set an initial set of options for the profile using the window shown in the following figure.
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Make the required changes and click Save at the right hand bottom of the panel to save the changes made.
To create a new Packet Service Profile, click New Packet Service Profile tab on the Packet Service Profile List panel.
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The Create New Packet Service Profile window is displayed.
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The following fields are displayed:
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Parameter
Description
Name
The packet service profile entry ID used to identify a particular packet service profile entry.
Silence Factor
The silence factor is the percentage of call time for which silence is expected. It is used to reduce expected call bandwidth.
The default value is 40.
Type of Service
Specifies the type of service (TOS) parameter to be provided in the IP header for voice packets. It is the decimal number that is included as is in the 8-bit TOS field of the IP header. Note that this number should be four times the DSCP value that you want to set in the high order 6 bits of the 8-bit TOS field.
The default value is 0.
Voice Initial Playout Buffer Delay
Specifies a numeric value, in milliseconds (ms), for the voice initial playout buffer delay required to absorb the maximum expected packet jitter across the network, in the range of 1 ms to 50 ms in increments of 1 ms.
The default value is 10 ms.
Peer Absence
Action
AaL1 Payload
Size
Specifies the ATM Adaption Layer Type 1 (AAL-1) payload size. For G.711, the possible values are 40, 44, or 47 bytes.
The default value is 47 bytes.
Preferred RTP Payload Type for DTMF Relay
Specifies the preferred RTP payload type in the RTP header of audio packets for this encoding. (default = 128). This parameter is only used for 8 kHz clock rate. DTMF payload type of each subsequent clock rate (16 kHz, 24 kHz, etc.) is incremented by 1.
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Using the default value of "128" for |
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Refer to If |
Media Packet
COS
Applies a Class of Service (COS) marking to be set in the User Priority field of the Ethernet VLAN tag header on media packets transmitted on a call leg that uses this packet service profile. Has an effect only if the network interface supports 802.1Q tagged Ethernet frames. The default value zero corresponds to best effort. The value range is 0-7.
Honor Remote
Precedence
Specifies whether the audio encoding priority order of the local packet service profile takes precedence over the remote peer's audio encoding priority order when creating the priority order of the audio encodings that are common to both.
Possible choices are:
• disabled
—When selected, Disabled makes the local audio encoding priority order take precedence. Disabled also makes the local Secure RTP/RTCP settings and crypto suite priority order take precedence.
• enabled
—When selected, Enabled makes the remote peer's audio encoding priority order take precedence. For ingress call legs, Enabled also makes the remote peer's Secure RTP/RTCP settings and crypto suite priority order take precedence. The default setting is Disabled.
Send Route
PSPPrecedence
Specifies the audio encoding order preference in outgoing messages only.
• disabled
—When selected, Disable the audio encoding order preference.
• enabled
—When selected, Enable the audio encoding order preference.
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To edit an existing Packet Service Profile:
To copy an existing Packet Service Profile as the basis for a new profile,
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The following fields are displayed:
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Name | The packet service profile entry ID used to identify a particular packet service profile entry. | |||||||||||||||||||||||||||||
Silence Factor | The silence factor is the percentage of call time for which silence is expected. It is used to reduce expected call bandwidth. The default value is 40. | |||||||||||||||||||||||||||||
Type of Service | Specifies the type of service (TOS) parameter to be provided in the IP header for voice packets. It is the decimal number that is included as is in the 8-bit TOS field of the IP header. Note that this number should be four times the DSCP value that you want to set in the high order 6 bits of the 8-bit TOS field. The default value is 0. | |||||||||||||||||||||||||||||
Voice Initial Playout Buffer Delay | Specifies a numeric value, in milliseconds (ms), for the voice initial playout buffer delay required to absorb the maximum expected packet jitter across the network, in the range of 1 ms to 50 ms in increments of 1 ms. The default value is 10 ms. | |||||||||||||||||||||||||||||
Peer Absence Action | Specifies the action to be taken when loss of bearer plane connectivity is detected on the channel. Possible actions are:
Requires the RTCP check box to be selected, which enables RTCP on the channel. The default setting is None. | |||||||||||||||||||||||||||||
AaL1 Payload Size | Specifies the ATM Adaption Layer Type 1 (AAL-1) payload size. For G.711, the possible values are 40, 44, or 47 bytes. | |||||||||||||||||||||||||||||
Preferred RTP Payload Type for DTMF Relay | Specifies the preferred RTP payload type in the RTP header of audio packets for this encoding. (default = 128). This parameter is only used for 8 kHz clock rate. DTMF payload type of each subsequent clock rate (16 kHz, 24 kHz, etc.) is incremented by 1.
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Media Packet COS | Applies a Class of Service (COS) marking to be set in the User Priority field of the Ethernet VLAN tag header on media packets transmitted on a call leg that uses this packet service profile. Has an effect only if the network interface supports 802.1Q tagged Ethernet frames. The default value zero corresponds to best effort. The value range is 0-7. | |||||||||||||||||||||||||||||
Honor Remote Precedence | Specifies whether the audio encoding priority order of the local packet service profile takes precedence over the remote peer's audio encoding priority order when creating the priority order of the audio encodings that are common to both. Possible choices are:
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Send Route PSPPrecedence | Specifies the audio encoding order preference in outgoing messages only. The options are:
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Data Calls | ||||||||||||||||||||||||||||||
Preferred Rtp Data Payload Type | The RTP Payload Type included in the RTP header of the data packet. The value ranges from 0 to 127 and the default value is 56. | |||||||||||||||||||||||||||||
Initial Playout Buffer Delay | Used for G.711 only. This is the initial playout delay for calls with a data bearer channel, for example, ISDN 64K data calls. This value is configured separately from the initial playout delay for voice channels (Voice Initial Playout Buffer Delay) so providers can trade off delay on data calls versus the likelihood of jitter causing delay changes while the playout buffer adapts. Some data bearer calls are very sensitive to delay changes (such as H.320 video conferencing), so a higher initial delay should reduce the chance of jitter bursts causing problems. (range: 5-50 / default = 50). | |||||||||||||||||||||||||||||
Packet Size | Specifies the maximum data packet size (Kilobits). The options are:
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RTCP Options | ||||||||||||||||||||||||||||||
Rtcp | Specifies whether to enable RTCP. The options are:
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Termination For Passthrough | Specifies RTCP termination behavior for pass-through calls. The options are:
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Enable RTCPFor Held Calls | If this option is enabled, the SBC ignores the configured RR/RS values in the Packet Service Profile and send RR/RS = 0 in the offer/answer and disables RTCP when the call is active. When the call is HELD, and a RE-INVITE is sent, the SBC uses the configured values in the Packet Service Profile for RTCP bandwidth and enables RTCP. When the call is RESUMED, the SBC again disables RTCP by sending RR/RS=0 in the RE-INVITE. The value of RR ranges from 100-4000 and the value of RS ranges from 100-3000. If this flag is disabled, the older behavior of SBC is applicable. The options are:
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Packet Loss Threshold | Enter a value of 0, or a value in the range of 400-32767 to specify the
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To copy any of the created Packet Service Profiles and to make any minor changes, click the radio button next to the specific Packet Service Profile to highlight the row.
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Click Copy Packet Service Profile tab on the Packet Service Profile List panel.
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The Copy Selected Packet Service Profile window is displayed along with the field details which can be edited.
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The following fields are displayed:
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Parameter | Description | ||||||||||||||||||||
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| The packet service profile entry ID used to identify a particular packet service profile entry. | ||||||||||||||||||||
| The silence factor is the percentage of call time for which silence is expected. It is used to reduce expected call bandwidth. The default value is 40. | ||||||||||||||||||||
| Specifies the type of service (TOS) parameter to be provided in the IP header for voice packets. It is the decimal number that is included as is in the 8-bit TOS field of the IP header. Note that this number should be four times the DSCP value that you want to set in the high order 6 bits of the 8-bit TOS field. The default value is 0. | ||||||||||||||||||||
| Specifies a numeric value, in milliseconds (ms), for the voice initial playout buffer delay required to absorb the maximum expected packet jitter across the network, in the range of 1 ms to 50 ms in increments of 1 ms. The default value is 10 ms. |
| Specifies the action to be taken when loss of bearer plane connectivity is detected on the channel. Possible actions are: |||||||||||||||||||
| Specifies the ATM Adaption Layer Type 1 (AAL-1) payload size. For G.711, the possible values are 40, 44, or 47 bytes. | ||||||||||||||||||||
| Specifies the preferred RTP payload type in the RTP header of audio packets for this encoding. (default = 128). This parameter is only used for 8 kHz clock rate. DTMF payload type of each subsequent clock rate (16 kHz, 24 kHz, etc.) is incremented by 1.
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| Applies a Class of Service (COS) marking to be set in the User Priority field of the Ethernet VLAN tag header on media packets transmitted on a call leg that uses this packet service profile. Has an effect only if the network interface supports 802.1Q tagged Ethernet frames. The default value zero corresponds to best effort. The value range is 0-7. | ||||||||||||||||||||
| Specifies whether the audio encoding priority order of the local packet service profile takes precedence over the remote peer's audio encoding priority order when creating the priority order of the audio encodings that are common to both. Possible choices are:
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| Specifies the audio encoding order preference in outgoing messages only. The options are:
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Data Calls | |||||||||||||||||||||
Preferred Rtp Data Payload Type | The RTP Payload Type included in the RTP header of the data packet. The value ranges from 0 to 127 and the default value is 56. | ||||||||||||||||||||
Initial Playout Buffer Delay | Used for G.711 only. This is the initial playout delay for calls with a data bearer channel, for example, ISDN 64K data calls. This value is configured separately from the initial playout delay for voice channels (Voice Initial Playout Buffer Delay ) so providers can trade off delay on data calls versus the likelihood of jitter causing delay changes while the playout buffer adapts. Some data bearer calls are very sensitive to delay changes (such as H.320 video conferencing), so a higher initial delay should reduce the chance of jitter bursts causing problems. (range: 5-50 / default = 50). | ||||||||||||||||||||
Packet Size | Specifies the maximum data packet size (Kilobits). The options are:
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Rtcp Options | |||||||||||||||||||||
Rtcp | Specifies if the Rtcp option should be enabled or not. The options are:
| Termination For Passthrough | Specifies RTCP termination behavior for pass-through calls. The options are:
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This parameter is visible only when |
Enable RTCPFor Held Calls
If this flag is enabled, SBC ignores the configured RR/RS values in the Packet Service Profile and send RR/RS = 0 in the offer/answer and disables RTCP when the call is active. When the call is HELD, and a RE-INVITE is sent, SBC uses the configured values in the Packet Service Profile for RTCP bandwidth and enables RTCP. When the call is RESUMED, SBC again disables RTCP by sending RR/RS=0 in the RE-INVITE.
The value of RR ranges from 100-4000 and the value of RS ranges from 100-3000.
If this flag is disabled, the older behavior of SBC is applicable.
options are:
disabled
(default)enabled
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This parameter is visible only when |
Packet Loss Threshold
Enter a value between 0-32767 to specify the Packet Loss Threshold
(number of lost packets/100,000) which will trigger a Packet Loss Action
. This parameter is required if RTCP is enabled. When set to “0”, no packet loss inactivity detection is performed. The value ranges from 0 to 32767 and the default value is 0. The value ranges from 0 to 32767 and the default value is 0.
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Configuring this parameter to a value less than 400 disables threshold detection, so be sure to use a value in the range of 400 to 32767 to enable threshold detection. |
This setting can be used in conjunction with Media Peer Inactivity. To set media amedia peer inactivity timeout value, see the Media Peer Inactivity parameter in parameter on the Media Peer Inactivity page.
For an example configuration of this parameter, see the Packet Service Profile (CLI) page.
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This parameter is visible visible only when Rtcp is enabledenabled. |
Specifies the RTCP bandwidth allocated to active data senders. The value ranges from 100 to 4000 and the default value is 250.
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This parameter is visible visible only when Rtcp is enabledenabled. |
Specifies the RTCP bandwidth allocated for receivers. The value ranges from 100 to 3000 and the default value is 250.
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This parameter is visible visible only when Rtcp is enabledenabled. |
none
None — Take no action.packetLossTrap
Packet Loss Trap — Generate trap.packetLossTrapAndDisconnect
— Generate Packet Loss Trap And Disconnect
— Generate trap and disconnect.
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This parameter is visible visible only when Rtcp is enabledenabled. |
By default, this flag option is enabled to allow SID packets to be sent within a minimal interval during a silence period (at least one SID packet must be sent within a SID maximum packet time frame). The options are:
disable
Disableenable
Enable (default)Codec
CodecDefines the codec entry priorities and codec names. Up to 12 codec configurations are supported by the SBC in PSX and Advanced ERE deployment scenarios (see Routing and Policy Management for a description of the different routing configurations).
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This attribute specifies the codec entry with a priority of "1". For each codec entry, select the desired codec. Codec IDs available by default are:
codecEntry1
Codec Entry1
.codecEntry1
Codec
Entry1
.codecEntry1
Codec
Entry1
.codecEntry1
Codec
Entry1
.codecEntry1
Codec
Entry1
.codecEntry1
Codec
Entry1
.codecEntry1
Codec
Entry1
.codecEntry1
Codec
Entry1
.codecEntry1
Codec
Entry1
.codecEntry1
Codec
Entry1
.codecEntry1
Codec
Entry1
.Transcode
Transcode options:
conditional
Conditional (default)determinedByPspForOtherLeg
only
transcoderFreeTransparency
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If you configure Packet To Packet Control for “Transcode Transcoder Free Transparency”, ensure you also set Late Media Support to ‘passthru ’ Passthru (refer to Trunk Group - SIP Trunk Group and SIP Trunk Group - Media). |
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Apply fax tone treatment. The options are:
disable
Disable (default)enable
EnableEnable this flag to perform transcoding when the ingress and egress call legs use different DTMF relay methods. The options are:
disable
Disable (default)enable
Enable Enable this flag to perform transcoding when the ingress and egress call legs use different packet sizes. The options are:
disable
Disable (default)enable
Enable
Enable this flag to perform transcoding when the ingress and egress call legs use different silence suppression methods. The options are:
disable
Disable (default)enable
Enable this Honor Offer Preference (HOP) flag to honor the codec preference of the peer's offer when the 'Honor Remote Preference' flag on the PSX is enabled. This option is available only when transcode
= conditional
Transcode is Conditional. (See the table below describing SBC behavior when this flag option is enabled/disabled). The options are:
disable
Disable (default)enable
EnableThe SBC triggers a new offer towards the other side when an answer is received for a re-INVITE from this side. The re-INVITE generated on the other side carries all possible codecs in Route Packet Service Profile that causes the most preferred codec of the other side peer to be modified. Enable this Honor Answer Preference (HAP) flag to lock down the most preferred codec towards the peer irrespective of re-INVITE received for for mid-call modification from this side. (See the table below describing SBC behavior when this flag option is enabled/disabled). The options are:
disable
Disable (default)enable
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HOP Flag State | HAP Flag State | SBC Behavior |
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enableEnable | disableDisable | The SBC selects a codec order of precedence in the offered SDP, irrespective of whether it is a pass-through or transcoded codec (if transcoding is defined for that codec). The SBC as part of media lock-down may send a re-INVITE to egress peer. Note that the preference on the answerer side is given to a pass-through codec. |
enableEnableenable | Enable | The SBC gives preference to HAP over HOP in case of conflict. The Honor Remote Preference (HRP) flag on the answerer leg decides the preference order. Based on that preference list, the SBC selects a codec with highest preference from answer SDP that can be used even if it requires transcoding. Note that this may cause the selection of a codec on the other side leg not to be honored. This happens in case of a pass-through call. |
disableDisable | enableEnable | The SBC gives preference to answerer codec order that is created based on HRP flag. The most preferred codec is chosen as received in the answer SDP, irrespective of whether it is a pass-through or a transcoded codec (if transcoding is defined for that codec). |
amr | efr | evrc | g711a |
g711u | g722 | g726 | g729 |
g7221 | g7222 | g7231 | ilbc |
t38 |
Enable flag Enable this option to disallow data calls. The options are:
disable
Disable (default)enable
EnableFlag to specify Specifies whether digit detection is enabled on digits sent to the network. The options are:
disable
Disable (default)enable
EnableINFO: See Digit Detect Send Enabled Settings for KPML table below to understand which PSP leg to enable this flag for the desired KPML functionality.
Use Direct Media
Enable flag to use direct media as needed. The options are:
disable
(default)enable
Validate Peer Support For Dtmf Events
Flag to validate peer support for DTMF events. Enable this flag for all peer devices that support RFC 4733. The options are:
disable
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Interwork Dtmf Without Transcoding
Enable flag to interwork DTMF with out-of-band RFC2833 without using transcoding. The options are:
disable
(default)enable
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Dscp Passthrough
When enabled on both the Ingress and Egress call leg, the DSCP value in the IP header of the media packets is transparently passed through the system. Once media is received from the peer, any value set in the Type Of Service
field on the Packet Service Profile has no effect when Dscp Passthrough
is configured on both legs for the associated call. The options are:
disable
(default)enable
Ssrc Randomize
Enable flag to generate a new SSRC (using a random value) along with a new timestamp on a new RTP stream whenever a resource is reactivated (due to change in codec, etc.). SSRC randomization reduces the probability of collision in large groups and simplifies the process of group sampling that depends on uniform distribution of SSRCs. The options are:
disable
(default)enable
Data Rate Management Type
The following Data Rate Management Types are supported:
type1LocalGenerationOfTcf
– Type 1 data rate management requires that the Training Check Frame (TCF) training signal is generated locally by the receiving gateway. Data rate management is performed by the emitting gateway based on training results from both PSTN connections. Type 1 is used for TCP implementations and is optionally used with UDP implementations.type2TransferOfTcf
– (default) Type 2 data rate management requires that the TCF is transferred from the sending gateway to the receiving gateway rather than having the receiving gateway generate it locally. Speed selection is done by the gateways in the same way as they would on a regular PSTN connection. Data rate management type 2 requires the use of UDP and is not recommended for use with TCP.Low Speed Number Of Redundant Packets
Max Bit Rate
Use this object to select the T.38 Maximum Bit Rate which controls and manipulates bits 11, 12, 13, and 14 in the DIS command received by the SBC from either the TDM circuit interface or the T.38 packet interface:
2.4Kbits_s
– For modem type ITU-T V.27ter fall-back mode.4.8Kbits_s
– For modem type ITU-T V.27ter.9.6Kbits_s
– For modem types ITU-T V.27ter and V.29.14.4Kbits_s
– (default) For modem types ITU-T V.27ter, V.29, and V.17. This setting is used to constrain the type of modem modulation schemes.Number Of Redundant Packets
Ecm Preferred
Use this flag to allocate DSP resources, when available, for T.38 Error Correction Mode (ECM) calls.
disable
– (default) use normal resource allocation.enable
Secure Rtp Rtcp
Crypto Suite Profile
Allow Fallback
Enable flag to allow fallback to standard RTP/RTCP when crypto attribute negotiation fails. The options are:
disable
(default)enable
Enable Srtp
Enable this flag to enable secure RTP/RTCP. The options are:
disable
(default)enable
Reset ROCOn Key Change
Enable flag to reset the SRTP Roll Over Counter when the session key changes. The options are:
disable
(default)enable
Reset Enc Dec ROCOn Dec Key Change
Enable flag to reset Roll Over Counter for both encryption and decryption when decryption key changes. The options are:
disable
(default)enable
Update Crypto Keys On Modify
For an SRTP call, if this flag is enabled in Packet Service Profile and call leg mode is changed from sendonly/inactive/recvonly to sendrecv, the SBC generates a new set of crypto attributes. The options are:
disable
(default)enable
Max Video Bandwidth
The maximum allowable session bandwidth (in Kbps) for a call that includes video streams. This value includes the bandwidth for all streams in the call (audio, video, BFCP, and so on). If "0" is set as the value, video calls are not allowed; and only audio calls can be set up following the normal allocation process (range: 0-50000 Kbps / default = 10).
Video Bandwidth Reduction Factor
The amount, as a percentage, to reduce the session bandwidth allocation for calls that include video streams. This setting only affects the internal allocation of bandwidth used for the calls (does not affect the signaling). For example, if the reduction factor is "20", the bandwidth allocated for calls is reduced by 20%. In other words, if the normal bandwidth allocation for calls is 1000 Kbps, a 20% reduction equates to a new 800 Kbps bandwidth. (range: 0-100 / default = 0).
Ipv4Tos
Ipv6Traffic Class
IPv6 traffic class. (range: 0-255 / default = 0).
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Ieee8021QVLan Cos
Codec List Profile
Audio Only If Video Is Prevented
By default, this flag is enabled to allow call to continue with the audio only portion if the video cannot be established for any reason. The options are:
disable
enable
(default)Unknown Codec Packet Size
Unknown Codec Bit Rate
Dtls Crypto Suite Profile
Allow Dtls Fallback
When enabled, specifies a fall back to standard RTP when crypto attribute negotiation fails. The options are:
Disable (default)
Enable
Enable Dtls Srtp
When enabled, this parameter enables the secure RTP. The options are:
Disable (default)
Enable
Dtls Srtp Relay
When enabled, the Relays DTLS-sRTP audio and video streams are enabled on the SBC. The options are:
Disable (default)
Enable
Dtls Sctp Relay
When enabled, the Relays DTLS/SCTP streams are enabled on the SBC. The options are:
Disable (default)
Enable
Make the required changes to the required fields and click Save to save the changes. The copied Packet Service Profile is displayed at the bottom of the original Packet Service Profile in the Packet Service Profile List panel.
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To delete any of the created Packet Service Profile, click the radio button next to the specific Packet Service Profile which you want to delete.
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Click Delete at the end of the highlighted row. A delete confirmation message appears seeking your decision.
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Click Yes to remove the specific Packet Service Profile from the list.
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In this section:
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Info | ||
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Additional pages:
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flag for the desired KPML functionality. | |||||||||||||||||||||
Use Direct Media | Enable this option to use direct media as needed. The options are:
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Validate Peer Support For Dtmf Events | Enable this option to validate peer support for DTMF events. Enable this option for all peer devices that support RFC 4733. The options are:
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Interwork Dtmf Without Transcoding | Enable this option to interwork DTMF with out-of-band RFC2833 without using transcoding. The options are:
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Dscp Passthrough | When enabled on both the Ingress and Egress call leg, the DSCP value in the IP header of the media packets is transparently passed through the system. Once media is received from the peer, any value set in the Type Of Service field on the Packet Service Profile has no effect when Dscp Passthrough is configured on both legs for the associated call. The options are:
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Ssrc Randomize | Enable flag to generate a new SSRC (using a random value) along with a new timestamp on a new RTP stream whenever a resource is reactivated (due to change in codec, etc.). SSRC randomization reduces the probability of collision in large groups and simplifies the process of group sampling that depends on uniform distribution of SSRCs. The options are:
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HDCodec Preferred | Enable flag to set HD codecs as preferred codec over non-HD codecs even if transcoding is required. When flag is disabled, continue with existing PSP/IPSP behavior. The options are:
When enabled,
Note:
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Prefer NBPassthru Over HDTranscode | Enable this option to allow the SBC to choose NB-NB pass-through over HD-HD transcoded call. The options are:
When disabled, the SBC prefers HD-HD transcoded call over NB-NB pass-thru. Note:
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Match Offered Codec Group If Nb Only | Enable this option to allow the SBC to send only NB in the outgoing offer if only NB is received in the ingress offer. Otherwise, do nothing. While sending the offer, this option is ignored if either HD-only or (HD+NB) is received in incoming offer. The options are:
If this option is disabled, the SBC uses existing behavior. Note: If Transcoder Free Transparency is enabled, this option is ignored. | ||||||||||||||||||||
Force Route PSPOrder | Enable this option to send the outgoing offer in the same order as in the egress route Packet Service Profile, irrespective of HD/NB priorities. The options are:
Note:
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Generate and Signal SSRCAnd Cname | Enable this flag to generate an SSRC value and associated attributes and include them in SDP signaling and RTP/RTCP streams. Options are:
Note: This flag takes precedence over the Packet Service Profile Ssrc Randomize flag. | ||||||||||||||||||||
Allow Mid Call SSRCModification | Enable this flag so that in call hold/resume scenarios the SBC modifies the SSRC and associated attributes after the call resumes. The SBC sends both the previous and updated SSRC in SDP signaling and includes the new SSRC iin RTP/RTCP streams. Options are:
Note: You must enable the Generate and Signal SSRCAnd Cname flag before you can enable this flag. Note: If you enable the IP Signaling profile common IP attributes flag Minimize Relaying Of Media Changes From Other Call Leg All, you must also enable the Relay Data Path Mode Change From Other Call Leg flag to have the SSRC modification processing take effect. | ||||||||||||||||||||
Reserve BW For Preferred Audio Common Codec | Reserves bandwidth on the basis of the preferred common codec, and polices on the worst case codec. This applies to both known and unknown codecs. The options are:
Note: This option is active for a call when both PSPs have this option enabled. If this option is disabled in either of the PSPs, the option is not applied. | ||||||||||||||||||||
Police On Heaviest Audio Codec | When enabled, the SBC reserves bandwidth based on the worst-case common codec on trunk groups and interfaces, but polices on the maximum bandwidth for all codecs from the Offer or Answer in a pass-through call. The options are:
Note: This configuration applies to all pass-through calls. It works independently from Audio Transparency feature and | ||||||||||||||||||||
T140Call | Specifies whether text media calls, using T.140 codec, are allowed. The options are:
For more information on text codecs, refer to Text Codecs. | ||||||||||||||||||||
Allow Audio Transcode For Multi Stream Call | Use this option to enable audio transcoding for multi-stream calls. The options are:
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The Packet Service Profile screen enables you to create or edit a Packet Service Profile. Each Packet Service Profile is configured for a pair of gateways and includes entries for up to four audio/video encoding methods. The pair of gateways can be originating and destination gateways in the same gateway group, or can be originating and destination gateways in an inter-gateway group.
For egress call legs over IP trunk groups, you can use the Trunk Group screen to assign a packet service profile to an egress IP trunk group.
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On the SBC main screen, go to Configuration > System Provisioning > Category: Trunk Provisioning > Packet Service Profile. The Packet Service Profile window is displayed.
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To edit any of the Packet Service Profiles in the list, click the radio button next to the specific Packet Service Profile name.
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The Edit Selected Packet Service Profile window is displayed below.
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Make the required changes and click Save at the right hand bottom of the panel to save the changes made.
To create a new Packet Service Profile, click New Packet Service Profile tab on the Packet Service Profile List panel.
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The Create New Packet Service Profile window is displayed.
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The following fields are displayed:
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Parameter
Description
Name
The packet service profile entry ID used to identify a particular packet service profile entry.
Silence Factor
The silence factor is the percentage of call time for which silence is expected. It is used to reduce expected call bandwidth.
The default value is 40.
Type of Service
Specifies the type of service (TOS) parameter to be provided in the IP header for voice packets. It is the decimal number that is included as is in the 8-bit TOS field of the IP header. Note that this number should be four times the DSCP value that you want to set in the high order 6 bits of the 8-bit TOS field.
The default value is 0.
Voice Initial Playout Buffer Delay
Specifies a numeric value, in milliseconds (ms), for the voice initial playout buffer delay required to absorb the maximum expected packet jitter across the network, in the range of 1 ms to 50 ms in increments of 1 ms.
The default value is 10 ms.
Peer Absence
Action
AaL1 Payload
Size
Specifies the ATM Adaption Layer Type 1 (AAL-1) payload size. For G.711, the possible values are 40, 44, or 47 bytes.
The default value is 47 bytes.
Preferred RTP Payload Type for DTMF Relay
Specifies the preferred RTP payload type in the RTP header of audio packets for this encoding. (default = 128). This parameter is only used for 8 kHz clock rate. DTMF payload type of each subsequent clock rate (16 kHz, 24 kHz, etc.) is incremented by 1.
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Using the default value of "128" for |
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Refer to If |
Media Packet
COS
Applies a Class of Service (COS) marking to be set in the User Priority field of the Ethernet VLAN tag header on media packets transmitted on a call leg that uses this packet service profile. Has an effect only if the network interface supports 802.1Q tagged Ethernet frames. The default value zero corresponds to best effort. The value range is 0-7.
Honor Remote
Precedence
Specifies whether the audio encoding priority order of the local packet service profile takes precedence over the remote peer's audio encoding priority order when creating the priority order of the audio encodings that are common to both.
Possible choices are:
• disabled
—When selected, Disabled makes the local audio encoding priority order take precedence. Disabled also makes the local Secure RTP/RTCP settings and crypto suite priority order take precedence.
• enabled
—When selected, Enabled makes the remote peer's audio encoding priority order take precedence. For ingress call legs, Enabled also makes the remote peer's Secure RTP/RTCP settings and crypto suite priority order take precedence. The default setting is Disabled.
Send Route
PSPPrecedence
Specifies the audio encoding order preference in outgoing messages only.
• disabled
—When selected, Disable the audio encoding order preference.
• enabled
—When selected, Enable the audio encoding order preference.
To copy any of the created Packet Service Profiles and to make any minor changes, click the radio button next to the specific Packet Service Profile to highlight the row.
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Click Copy Packet Service Profile tab on the Packet Service Profile List panel.
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The Copy Selected Packet Service Profile window is displayed along with the field details which can be edited.
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The following fields are displayed:
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Parameter | Description | ||||||||||||||||||||
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| The packet service profile entry ID used to identify a particular packet service profile entry. | ||||||||||||||||||||
| The silence factor is the percentage of call time for which silence is expected. It is used to reduce expected call bandwidth. The default value is 40. | ||||||||||||||||||||
| Specifies the type of service (TOS) parameter to be provided in the IP header for voice packets. It is the decimal number that is included as is in the 8-bit TOS field of the IP header. Note that this number should be four times the DSCP value that you want to set in the high order 6 bits of the 8-bit TOS field. The default value is 0. | ||||||||||||||||||||
| Specifies a numeric value, in milliseconds (ms), for the voice initial playout buffer delay required to absorb the maximum expected packet jitter across the network, in the range of 1 ms to 50 ms in increments of 1 ms. The default value is 10 ms. |
| Specifies the action to be taken when loss of bearer plane connectivity is detected on the channel. Possible actions are: |||||||||||||||||||
| Specifies the ATM Adaption Layer Type 1 (AAL-1) payload size. For G.711, the possible values are 40, 44, or 47 bytes. | ||||||||||||||||||||
| Specifies the preferred RTP payload type in the RTP header of audio packets for this encoding. (default = 128). This parameter is only used for 8 kHz clock rate. DTMF payload type of each subsequent clock rate (16 kHz, 24 kHz, etc.) is incremented by 1.
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| Applies a Class of Service (COS) marking to be set in the User Priority field of the Ethernet VLAN tag header on media packets transmitted on a call leg that uses this packet service profile. Has an effect only if the network interface supports 802.1Q tagged Ethernet frames. The default value zero corresponds to best effort. The value range is 0-7. | ||||||||||||||||||||
| Specifies whether the audio encoding priority order of the local packet service profile takes precedence over the remote peer's audio encoding priority order when creating the priority order of the audio encodings that are common to both. Possible choices are:
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| Specifies the audio encoding order preference in outgoing messages only. The options are:
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Data Calls | |||||||||||||||||||||
Preferred Rtp Data Payload Type | The RTP Payload Type included in the RTP header of the data packet. The value ranges from 0 to 127 and the default value is 56. | ||||||||||||||||||||
Initial Playout Buffer Delay | Used for G.711 only. This is the initial playout delay for calls with a data bearer channel, for example, ISDN 64K data calls. This value is configured separately from the initial playout delay for voice channels (Voice Initial Playout Buffer Delay ) so providers can trade off delay on data calls versus the likelihood of jitter causing delay changes while the playout buffer adapts. Some data bearer calls are very sensitive to delay changes (such as H.320 video conferencing), so a higher initial delay should reduce the chance of jitter bursts causing problems. (range: 5-50 / default = 50). | ||||||||||||||||||||
Packet Size | Specifies the maximum data packet size (Kilobits). The options are:
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Rtcp | Specifies if the Rtcp option should be enabled or not. The options are:
| Termination For Passthrough | Specifies RTCP termination behavior for pass-through calls. The options are:
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This parameter is visible only when |
Enable RTCPFor Held Calls
If this flag is enabled, SBC ignores the configured RR/RS values in the Packet Service Profile and send RR/RS = 0 in the offer/answer and disables RTCP when the call is active. When the call is HELD, and a RE-INVITE is sent, SBC uses the configured values in the Packet Service Profile for RTCP bandwidth and enables RTCP. When the call is RESUMED, SBC again disables RTCP by sending RR/RS=0 in the RE-INVITE.
The value of RR ranges from 100-4000 and the value of RS ranges from 100-3000.
If this flag is disabled, the older behavior of SBC is applicable.
options are:
disabled
(default)enabled
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This parameter is visible only when |
Packet Loss Threshold
Enter a value between 0-32767 to specify the Packet Loss Threshold
(number of lost packets/100,000) which will trigger a Packet Loss Action
. This parameter is required if RTCP is enabled. When set to “0”, no packet loss inactivity detection is performed. The value ranges from 0 to 32767 and the default value is 0. The value ranges from 0 to 32767 and the default value is 0.
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Configuring this parameter to a value less than 400 disables threshold detection, so be sure to use a value in the range of 400 to 32767 to enable threshold detection. |
This setting can be used in conjunction with Media Peer Inactivity
. To set media peer inactivity timeout value, see Media Peer Inactivity
parameter in Media Peer Inactivity page.
For an example configuration of this parameter, see Packet Service Profile (CLI) page.
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This parameter is visible only when |
Rr Bandwidth
Specifies the RTCP bandwidth allocated to active data senders. The value ranges from 100 to 4000 and the default value is 250.
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This parameter is visible only when |
Rs Bandwidth
Specifies the RTCP bandwidth allocated for receivers. The value ranges from 100 to 3000 and the default value is 250.
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This parameter is visible only when |
Packet Loss Action
none
— Take no action.packetLossTrap
— Generate trap.packetLossTrapAndDisconnect
— Generate trap and disconnect.
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This parameter is visible only when |
Silence Insertion Descriptor
G711Sid Rtp Payload Type
Heartbeat
By default, this flag is enabled to allow SID packets to be sent within a minimal interval during a silence period (at least one SID packet must be sent within a SID maximum packet time frame). The options are:
disable
enable
(default)Codec
Defines the codec entry priorities and codec names. Up to 12 codec configurations are supported by the SBC in PSX and Advanced ERE deployment scenarios (see Routing and Policy Management for a description of the different routing configurations).
Codec Entry1
This attribute specifies the codec entry with a priority of "1". For each codec entry, select the desired codec. Codec IDs available by default are:
G711-DEFAULT
G711SS-DEFAULT
G723-DEFAULT
G723A-DEFAULT
G726-DEFAULT
G729A-DEFAULT
G729AB-DEFAULT
Codec Entry2
codecEntry1
.Codec Entry3
codecEntry1
.Codec Entry4
codecEntry1
.Codec Entry5
codecEntry1
.Codec Entry6
codecEntry1
.Codec Entry7
codecEntry1
.Codec Entry8
codecEntry1
.Codec Entry9
codecEntry1
.Codec Entry10
codecEntry1
.Codec Entry11
codecEntry1
.Codec Entry12
codecEntry1
. Packet to Packet Control
Transcode
Transcode options:
conditional
(default)determinedByPspForOtherLeg
only
transcoderFreeTransparency
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Conditions in Addition To No Common Codec
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Apply Fax Tone Treatment
Apply fax tone treatment. The options are:
disable
(default)enable
Different DTMF Relay
Enable this flag to perform transcoding when the ingress and egress call legs use different DTMF relay methods. The options are:
disable
(default)enable
Different Packet Size
Enable this flag to perform transcoding when the ingress and egress call legs use different packet sizes. The options are:
disable
(default)enable
Different Silence Suppression
Enable this flag to perform transcoding when the ingress and egress call legs use different silence suppression methods. The options are:
disable
(default)enable
Honor Offer Preference
Enable this Honor Offer Preference (HOP) flag to honor the codec preference of the peer's offer when the 'Honor Remote Preference' flag on the PSX is enabled. This option is available only when transcode
= conditional
. (See the table below describing SBC behavior when this flag is enabled/disabled). The options are:
disable
(default)enable
Honor Answer Preference
The SBC triggers a new offer towards the other side when an answer is received for a re-INVITE from this side. The re-INVITE generated on the other side carries all possible codecs in Route Packet Service Profile that causes the most preferred codec of the other side peer to be modified. Enable this Honor Answer Preference (HAP) flag to lock down the most preferred codec towards the peer irrespective of re-INVITE received for mid-call modification from this side. (See the table below describing SBC behavior when this flag is enabled/disabled). The options are:
disable
(default)enable
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HOP Flag State | HAP Flag State | SBC Behavior |
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enable | disable | The SBC selects a codec order of precedence in the offered SDP, irrespective of whether it is a pass-through or transcoded codec (if transcoding is defined for that codec). The SBC as part of media lock-down may send a re-INVITE to egress peer. Note that the preference on the answerer side is given to a pass-through codec. |
enable | enable | The SBC gives preference to HAP over HOP in case of conflict. The Honor Remote Preference (HRP) flag on the answerer leg decides the preference order. Based on that preference list, the SBC selects a codec with highest preference from answer SDP that can be used even if it requires transcoding. Note that this may cause the selection of a codec on the other side leg not to be honored. This happens in case of a pass-through call. |
disable | enable | The SBC gives preference to answerer codec order that is created based on HRP flag. The most preferred codec is chosen as received in the answer SDP, irrespective of whether it is a pass-through or a transcoded codec (if transcoding is defined for that codec). |
Codecs Allowed for Transcoding
This Leg
Other Leg
amr | efr | evrc | g711a |
g711u | g722 | g726 | g729 |
g7221 | g7222 | g7231 | ilbc |
t38 |
Disallow Data Calls
Enable flag to disallow data calls. The options are:
disable
(default)enable
Digit Detect Send Enabled
Flag to specify whether digit detection is enabled on digits sent to the network. The options are:
disable
(default)enable
INFO: See Digit Detect Send Enabled Settings for KPML table below to understand which PSP leg to enable this flag for the desired KPML functionality. |
Use Direct Media
Enable flag to use direct media as needed. The options are:
disable
(default)enable
Validate Peer Support For Dtmf Events
Flag to validate peer support for DTMF events. Enable this flag for all peer devices that support RFC 4733. The options are:
disable
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Interwork Dtmf Without Transcoding
Enable flag to interwork DTMF with out-of-band RFC2833 without using transcoding. The options are:
disable
(default)enable
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Dscp Passthrough
When enabled on both the Ingress and Egress call leg, the DSCP value in the IP header of the media packets is transparently passed through the system. Once media is received from the peer, any value set in the Type Of Service
field on the Packet Service Profile has no effect when Dscp Passthrough
is configured on both legs for the associated call. The options are:
disable
(default)enable
Ssrc Randomize
Enable flag to generate a new SSRC (using a random value) along with a new timestamp on a new RTP stream whenever a resource is reactivated (due to change in codec, etc.). SSRC randomization reduces the probability of collision in large groups and simplifies the process of group sampling that depends on uniform distribution of SSRCs. The options are:
disable
(default)enable
The following Data Rate Management Types data rate management types are supported:
type1LocalGenerationOfTcf
Type1 Local Generation Of Tcf – Type 1 data rate management requires that the Training Check Frame (TCF) training signal is generated locally by the receiving gateway. Data rate management is performed by the emitting gateway based on training results from both PSTN connections. Type 1 is used for TCP implementations and is optionally used with UDP implementations.type2TransferOfTcf
Type2 Transfer Of Tcf – (default) Type 2 data rate management management requires that the TCF is transferred from the sending gateway to the receiving gateway rather than having the receiving gateway generate it locally. Speed selection is done by the gateways in the same way as they would on a regular PSTN connection. Data rate management type 2 requires the use of UDP and is not recommended for use with TCP.Use this object to select the T.38 Maximum Bit Rate maximum bit rate which controls and manipulates bits 11, 12, 13, and 14 in the DIS command received by the SBC from either the TDM circuit interface or the T.38 packet interface:
Use this flag to allocate DSP resources, when available, for T.38 Error Correction Mode (ECM) calls.
disable
Disable – (default) use normal resource allocation.enable
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Enable flag this option to allow fallback to standard RTP/RTCP when crypto attribute negotiation fails. The options are:
disable
Disable (default)enable
EnableEnable this flag option to enable secure RTP/RTCP. The options are:
disable
Disable (default)enable
EnableEnable flag this option to reset the SRTP Roll Over Counter when the session key changes. The options are:
disable
Disable (default)enable
EnableEnable flag this option to reset the Roll Over Counter for both encryption and decryption when decryption key changes. The options are:
disable
Disable (default)enable
EnableFor an SRTP call, if this flag option is enabled in the Packet Service Profile and the call leg mode is changed from sendonly/inactive/recvonly to sendrecv, the SBC generates a new set of crypto attributes. The options are:
disable
Disable (default)enable
EnableThe maximum allowable session bandwidth (in Kbps) for a call that includes video streams. This value includes the bandwidth for all streams in the call (audio, video, BFCP, and so on). If "0" is set as the value, video calls are not allowed; and only audio calls can be set up following the normal allocation process (range: 0-50000 Kbps / default = 10).
The amount, as a percentage, to reduce the session bandwidth allocation for calls that include video streams. This setting only affects the internal allocation of bandwidth used for the calls (does not affect the signaling). For example, if the reduction factor is "20", the bandwidth allocated for calls is reduced by 20%. In other words, if the normal bandwidth allocation for calls is 1000 Kbps, a 20% reduction equates to a new 800 Kbps bandwidth. (range: 0-100 / default = 0).
IPv6 traffic class. (range: 0-255 / default = 0).
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By default, this flag option is enabled to allow a call to continue with the audio portion only portion if the video cannot be established for any reason. The options are:
disable
Disable enable
Enable (default)
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When enabled, specifies a fall back to standard RTP when crypto attribute negotiation fails. The options are:
When enabled, this parameter enables the secure RTP. The options are:
When enabled, the Relays relay of DTLS-sRTP SRTP audio and video streams are is enabled on the SBC. The options are:
When enabled, the Relays relay of DTLS/SCTP streams are is enabled on the SBC. The options are:
Make the required changes to the required fields and click Save to save the changes. The copied Packet Service Profile is displayed at the bottom of the original Packet Service Profile in the Packet Service Profile List panel.the original Packet Service Profile in the Packet Service Profile List panel.
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To delete any of the created a Packet Service Profile, click :
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Click Yes to remove the specific Packet Service Profile from the list.