Parameter | Description |
|---|
| Fields common to the Create New and Copy Windows |
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: - None—No action taken.
- Trap—Generate an SNMP trap.
- Trap and Disconnect—Generate an SNMP trap and disconnect the call.
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.
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. | Info |
|---|
| Using the default value of "128" for Preferred Rtp Payload Type For Dtmf Relay implies that the preferred DTMF value (from the system configurable) is used for this profile. |
| Info |
|---|
| Refer to the Interwork Dtmf Without Transcoding parameter on the Packet Service Profile - Flags page for valid values when the Interwork Dtmf Without Transcoding parameter is enabled. If Interwork Dtmf Without Transcoding is enabled, ensure Preferred Rtp Payload Type For Dtmf Relay is set to a valid value (96-127). If the Preferred Rtp Payload Type For Dtmf Relay value is invalid (set to "128"), the system may fail to pick up the value configured using the "set system dspPad rtpDtmfRelay" option because DSPs are not used for the call. |
| Info |
|---|
| The uses the Preferred RTP Payload Type For DTMF Relay configuration as the payload type for 8 kHz DTMF when not used by any other codec.For wideband (16 kHz) DTMF, the uses the next available payload type (to the configured Preferred RTP Payload Type For DTMF Relay value). |
|
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 (default) — 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 — 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.
|
Send Route PSPPrecedence | Specifies the audio encoding order preference in outgoing messages only. The options are: - Disabled (default) —disable the audio encoding order preference.
- Enabled — enable the audio encoding order preference.
|
| Fields that appear only on the Copy Window |
| 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: |
| RTCP Options |
|
| Rtcp | Specifies whether to enable RTCP. The options are: - Disable (default)
- Enable – RTCP is used for the call.
|
| Termination For Passthrough | Specifies RTCP termination behavior for pass-through calls. The options are: - Disable (default) — RTCP is relayed between the endpoints for pass-through calls.
- Enable — Enable this option (as well as the Rtcp option) on one leg to terminate RTCP sessions on each leg for pass-through calls. If RTCP and RTCP termination is enabled on one leg of a pass-through call, RTCP is terminated and originated for that leg. If RTCP is enabled on both legs on the pass-through call, irrespective of Termination For Passthrough settings, RTCP is always relayed.
| Info |
|---|
| This parameter is visible only when Rtcp is enabled. |
|
| 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: - Disabled (default)
- Enabled
| Info |
|---|
| This parameter is visible only when Rtcp is enabled. |
|
| Packet Loss Threshold | Enter a value of 0, or a value in the range of 400-32767 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 default value is 0. | Info |
|---|
| Configuring this parameter to a value less than 400 disables threshold detection; 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 a media peer inactivity timeout value, see the Media Peer Inactivity parameter on the System - Media - Media Peer Inactivity page. For an example configuration of this parameter, see the Packet Service Profile - CLI page. | Info |
|---|
| This parameter is visible only when Rtcp is enabled. |
|
| Rr Bandwidth | Specifies the RTCP bandwidth allocated to active data senders. The value ranges from 100 to 4000, and the default value is 250. | Info |
|---|
| This parameter is visible only when Rtcp is enabled. |
|
| Rs Bandwidth | Specifies the RTCP bandwidth allocated for receivers. The value ranges from 100 to 3000, and the default value is 250. | Info |
|---|
| This parameter is visible only when Rtcp is enabled. |
|
| Packet Loss Action | Specifies what Packet loss action to take when packet threshold is exceeded. The options are: | Info |
|---|
| This parameter is visible only when Rtcp is enabled. |
|
| 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 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: |
| 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). | Excerpt Include |
|---|
| Supported Codecs and Transcoding |
|---|
| Supported Codecs and Transcoding |
|---|
| nopanel | true |
|---|
|
|
| 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 | This attribute specifies the codec entry with a priority of "2". For each codec entry, select the codec entry ID as specified for Codec Entry1. |
| Codec Entry3 | This attribute specifies the codec entry with a priority of "3". For each codec entry, select the codec entry ID as specified for Codec Entry1. |
| 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 Codec Entry1. |
| 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 Codec Entry1. |
| 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 Codec Entry1. |
| 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 Codec Entry1. |
| 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 Codec Entry1. |
| 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 Codec Entry1. |
| 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 Codec Entry1. |
| 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 Codec Entry1. |
| 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 Codec Entry1. |
| Packet to Packet Control |
|
| Transcode | Transcode options: - Conditional (default)
- Determined By Psp For Other Leg
- Only
- Transcoder Free Transparency
|
| Conditions in Addition To No Common Codec | The performs transcoding when any of the specified conditions are met, including no common codec on ingress and egress legs. |
| Apply Fax Tone Treatment | Apply fax tone treatment. The options are: |
| Different DTMF Relay | Enable this flag to perform transcoding when the ingress and egress call legs use different DTMF relay methods. The options are: |
| Different Packet Size | Enable this flag to perform transcoding when the ingress and egress call legs use different packet sizes. The options are: |
| Different Silence Suppression | Enable this flag to perform transcoding when the ingress and egress call legs use different silence suppression methods. The options are: |
| Honor Offer Preference | Enable Honor Offer Preference (HOP) to honor the codec preference of the peer's offer when Honor Remote Preference on the PSX is enabled. This option is available only when Transcode is Conditional. (See the table below describing SBC behavior when this option is enabled/disabled). The options are: |
| 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 Honor Answer Preference (HAP) 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 option is enabled/disabled). The options are: |
Table 2: honorAnswerPreference honorAnswerPreference vs. honorOfferPreference | HOP Flag State | HAP Flag State | SBC Behavior |
|---|
| 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 | Use this parameter to specify codecs allowed for transcoding, and for which call leg. |
| This Leg | (see codec list below) |
| Other Leg | (see codec list below) |
| amr | efr | evrc | g711a | | g711u | g722 | g726 | g729 | | g7221 | g7222 | g7231 | ilbc | | t38 |
|
|
|
|
| Flags |
|
|---|
| Disallow Data Calls | Enable this option to disallow data calls. The options are: |
| Digit Detect Send Enabled | Specifies 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 this option to use direct media as needed. The options are: |
| 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: - Disable (default) – does not validate the presence of DTMF events in the offer or answer from the peer. If DTMF relay is enabled, transmits DTMF digits received from the other leg to this peer using the named event RTP payload.
- Enable – validates the presence of DTMF events in the offer or answer from peers that support RFC 4733. If DTMF Relay is enabled and events 0-15 are received (with no other combination or subset of events), forwards the events in the egress leg to this peer using the named event RTP payload. When is configured for a pass-through call and it receives DTMF events other than 0-15 from the ingress peer, it does not offer any DTMF events to the egress endpoint.
|
| Interwork Dtmf Without Transcoding | Enable this option to interwork DTMF with out-of-band RFC2833 without using transcoding. The options are: | Info |
|---|
| If you enable Interwork Dtmf Without Transcoding, ensure Preferred Rtp Payload Type For Dtmf Relay is set to a valid value (96-127). If the Preferred Rtp Payload Type For Dtmf Relay value is invalid (set to "128"), the system may fail to pick up the value configured using the "set system dspPad rtpDtmfRelay" command because DSPs are not used for the call. |
|
| 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: |
| 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: |
| 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, | If... | Then... |
|---|
| the ingress Offer contains any valid HD codecs | HD codecs are sorted to the top of the list while sending out the Offer. | | all NB codecs are present | SBC reorders the codec entries with NB first, followed by HD codecs. |
| Codec selection priority from Answer message |
|---|
- HD-HD pass-through
- HD-HD transcoding
- NB-NB pass-through
- NB-NB transcoding
- HD-NB transcoding
|
Note: - If Force Route PSPOrder is enabled, this option does not affect the ordering of outgoing offer.
- If Transcoder Free Transparency is enabled, this option is ignored.
|
| 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: - This option is valid only if HDCodec Preferred is enabled, and it is applied when selecting a codec from answer.
- If Transcoder Free Transparency is enabled, this option is ignored.
|
| 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: - If this flag is enabled, the HDCodec Preferred option does not affect the ordering of outgoing offer.
- If Transcoder Free Transparency is enabled, this option is ignored.
|
| 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 in 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 Reserve BW For Preferred Audio Common Codec |
| 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: |
| Ssrc Randomize for Srtp | Enable this option to determine how the SBC handles SSRCs in SRTP media flows. When enabled, the SBC: - generates and replaces the SSRC for both pass-through and transcoded SRTP media flows
- generates a new SSRC value when a mid-call modification occurs (such as hold/resume)
- replaces the CNAME in the SRTCP SDES block
- replaces the SSRC in the SRTCP report blocks
When disabled, the SBC relays the SSRC for pass-through media flows it receives from the peer. The options are: |
| T38 |
|
|---|
| Data Rate Management Type | The following data rate management types are supported: - 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.
- Type2 Transfer Of Tcf – (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 | 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: - 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 | 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). |
| Ecm |
|
|---|
| 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 | | Include Page |
|---|
| _PSP_DTLS_SRTP_limitation |
|---|
| _PSP_DTLS_SRTP_limitation |
|---|
|
|
| Crypto Suite Profile | Enter the name of a crypto suite profile. Refer to Security Profiles - Crypto Suite Profile. |
| Flags |
|
|---|
| Allow Fallback | Enable this option to allow fallback to standard RTP/RTCP when crypto attribute negotiation fails. The options are: |
| Enable Srtp | Enable this option to enable secure RTP/RTCP. The options are: |
| Reset ROCOn Key Change | Enable this option to reset the SRTP Roll Over Counter when the session key changes. The options are: |
| Reset Enc Dec ROCOn Dec Key Change | Enable this option to reset the Roll Over Counter for both encryption and decryption when decryption key changes. The options are: |
| Update Crypto Keys On Modify | For an SRTP call, if this 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: |
| 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). | Info |
|---|
| IPv6Traffic Class is not supported with H.323 calls. |
|
| 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 option is enabled to allow a call to continue with the audio portion only if the video cannot be established for any reason. The options are: |
| 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 | | Include Page |
|---|
| _PSP_DTLS_SRTP_limitation |
|---|
| _PSP_DTLS_SRTP_limitation |
|---|
|
|
|---|
| Dtls Crypto Suite Profile | Enter the name of a crypto suite profile. Refer to Media - Packet Service Profile - DTLS. |
| Dtls Flags |
|
|---|
| Allow Dtls Fallback | When enabled, specifies a fall back to standard RTP when crypto attribute negotiation fails. The options are: |
| Enable Dtls Srtp | When enabled, this parameter enables secure RTP. The options are: |
| Dtls Srtp Relay | When enabled, relay of DTLS-SRTP audio and video streams is enabled on the SBC. The options are: |
| Dtls Sctp Relay | When enabled, relay of DTLS/SCTP streams is enabled on the SBC. The options are: |