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Additional pages:
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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.
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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. 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 | ||||
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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: | ||||
| Specifies the audio encoding order preference in outgoing messages only. |
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:
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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 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:
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Packet Loss Threshold | Enter a value between 0-32767 to specify the
This setting can be used in conjunction with For an example configuration of this parameter, see Packet Service Profile (CLI) page.
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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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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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Packet Loss Action | Specifies what Packet loss action to take when packet threshold is exceeded. The options are:
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Silence Insertion Descriptor | |||||||||||||||||||||
G711Sid Rtp Payload Type | Specifies the G.711 Silence Insertion Descriptor (SID) RTP payloadType. (range: 0-127 / default = 19). | ||||||||||||||||||||
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:
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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).
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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:
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Codec Entry2 | This attribute specifies the codec entry with a priority of "2". For each codec entry, select the codec entry ID as specified in codecEntry1 . | ||||||||||||||||||||
Codec Entry3 | This attribute specifies the codec entry with a priority of "3". For each codec entry, select the codec entry ID as specified in codecEntry1 . | ||||||||||||||||||||
Codec Entry4 | This attribute specifies the codec entry with a priority of "4". For each codec entry, select the codec entry ID as specified for codecEntry1 . | ||||||||||||||||||||
Codec Entry5 | This attribute specifies the codec entry with a priority of "5". For each codec entry, select the codec entry ID as specified for codecEntry1 . | ||||||||||||||||||||
Codec Entry6 | This attribute specifies the codec entry with a priority of "6". For each codec entry, select the codec entry ID as specified for codecEntry1 . | ||||||||||||||||||||
Codec Entry7 | This attribute specifies the codec entry with a priority of "7". For each codec entry, select the codec entry ID as specified for codecEntry1 . | ||||||||||||||||||||
Codec Entry8 | This attribute specifies the codec entry with a priority of "8". For each codec entry, select the codec entry ID as specified for codecEntry1 . | ||||||||||||||||||||
Codec Entry9 | This attribute specifies the codec entry with a priority of "9". For each codec entry, select the codec entry ID as specified for codecEntry1 . | ||||||||||||||||||||
Codec Entry10 | This attribute specifies the codec entry with a priority of "10". For each codec entry, select the codec entry ID as specified for codecEntry1 . | ||||||||||||||||||||
Codec Entry11 | This attribute specifies the codec entry with a priority of "11". For each codec entry, select the codec entry ID as specified for codecEntry1 . | ||||||||||||||||||||
Codec Entry12 | This attribute specifies the codec entry with a priority of "12". For each codec entry, select the codec entry ID as specified for codecEntry1 . | ||||||||||||||||||||
Packet to Packet Control | |||||||||||||||||||||
Transcode | Transcode options:
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Conditions in Addition To No Common Codec | The
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Apply Fax Tone Treatment | Apply fax tone treatment. The options are:
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Different DTMF Relay | Enable this flag to perform transcoding when the ingress and egress call legs use different DTMF relay methods. The options are:
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Different Packet Size | Enable this flag to perform transcoding when the ingress and egress call legs use different packet sizes. The options are:
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Different Silence Suppression | Enable this flag to perform transcoding when the ingress and egress call legs use different silence suppression methods. The options are:
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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
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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:
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Codecs Allowed for Transcoding | Use this parameter to specify codecs allowed for transcoding, and which leg to apply it to. | ||||||||||||||||||||
This Leg | (see codec list below) | ||||||||||||||||||||
Other Leg | (see codec list below) | ||||||||||||||||||||
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Disallow Data Calls | Enable flag to disallow data calls. The options are:
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Digit Detect Send Enabled | Flag to specify whether digit detection is enabled on digits sent to the network. The options are:
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:
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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:
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Interwork Dtmf Without Transcoding | Enable flag 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
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Ssrc Randomize | Enable flag to generate a new SSRC (using a random value) along with a new time stamp 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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T38 | |||||||||||||||||||||
Data Rate Management Type | The following Data Rate Management Types are supported:
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Low Speed Number Of Redundant Packets | This field specifies the number of redundant IFP messages sent in a UDP packet for T.38 low speed fax transmission, and applies only if the T.38 error correction type is redundancy. (range: 0-2 / default = 1). | ||||||||||||||||||||
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:
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Number Of Redundant Packets | Use this parameter for high-speed fax relay to specify the number of redundant Internet Facsimile Protocol (IFP) messages sent in a User Datagram Packet (UDP) for fax transmission. (range: 0-2 / default = 1). | ||||||||||||||||||||
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Ecm Preferred | Use this flag to allocate DSP resources, when available, for T.38 Error Correction Mode (ECM) calls.
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Secure Rtp Rtcp | |||||||||||||||||||||
Crypto Suite Profile | A unique identifier for the crypto suite profile. | ||||||||||||||||||||
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Allow Fallback | Enable flag to allow fallback to standard RTP/RTCP when crypto attribute negotiation fails. The options are:
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Enable Srtp | Enable this flag to enable secure RTP/RTCP. The options are:
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Reset ROCOn Key Change | Enable flag to reset the SRTP Roll Over Counter when the session key changes. The options are:
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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:
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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:
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Video Calls | |||||||||||||||||||||
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 | IPv4 type of service. (range: 0-255 / default = 0). | ||||||||||||||||||||
Ipv6Traffic Class | IPv6 traffic class. (range: 0-255 / default = 0).
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Ieee8021QVLan Cos | IEEE-802 1Q VLAN Class of Service. (range: 0-7 / default = 0) | ||||||||||||||||||||
Codec List Profile | Name of the Codec List profile used to store precedence and purge lists of video codec MIME subtypes. | ||||||||||||||||||||
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:
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Audio Transparency | |||||||||||||||||||||
Unknown Codec Packet Size | Specifies the bit rate, in Kilobytes/second, required for bandwidth computation of unknown audio codecs. (range: 1-1000 KB/sec / default = 124) | ||||||||||||||||||||
Unknown Codec Bit Rate | Specifies the packet size, in milliseconds, required for Bandwidth computation of unknown audio codecs. (range: 5-100 ms / default = 10) | ||||||||||||||||||||
DTLS | |||||||||||||||||||||
Dtls Crypto Suite Profile | Specifies a unique identifier for the crypto suite profile. | ||||||||||||||||||||
Dtls Flags | |||||||||||||||||||||
Allow Dtls Fallback | When enabled, specifies a fall back to standard RTP when crypto attribute negotiation fails. The options are:
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Enable Dtls Srtp | When enabled, this parameter enables the secure RTP. The options are:
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Dtls Srtp Relay | When enabled, the Relays DTLS-sRTP audio and video streams are enabled on the SBC. The options are:
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Dtls Sctp Relay | When enabled, the Relays DTLS/SCTP streams are enabled on the SBC. The options are:
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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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