Parameter | Length/Range | Description |
|---|
packetServiceProfile
| N/A | Specifies a unique identifier (name) for packet service profile. You can define any name or use Default as the parameter. |
aal1PayloadSize
| 0-999 | The ATM Adaption Layer Type 1 (AAL-1) payload size (default = 47). |
codec
| codecEntry1-codecEntry12 | <codecEntry ID> <codec_name> – Defines the codec entry priorities and codec names. Up to 12 codec configurations are supported by SBC in PSX, ePSX and Advanced ERE deployment scenarios (see Routing and Policy Management for a description of the different routing configurations).
| Note |
|---|
SBC-POL-RTU license is needed to enable more than four codecs. |
The codecEntry IDs are listed below: codecEntry1 – Codec entry with a priority of "1".codecEntry2 – Codec entry with a priority of "2". codecEntry3 – Codec entry with a priority of "3". codecEntry4 – Codec entry with a priority of "4".codecEntry5 – Codec entry with a priority of "5".codecEntry6 – Codec entry with a priority of "6".codecEntry7 – Codec entry with a priority of "7".codecEntry8 – Codec entry with a priority of "8".codecEntry9 – Codec entry with a priority of "9".codecEntry10 – Codec entry with a priority of "10".codecEntry11 – Codec entry with a priority of "11".codecEntry12 – Codec entry with a priority of "12".
For each codecEntry ID, select a codec name. Example default codec names are: G711-DEFAULT (default codec)
G711SS-DEFAULTG711_NONE
G723-DEFAULTG723A-DEFAULTG726-DEFAULTG729A-DEFAULTG729AB-DEFAULT
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dataCalls
| N/A | Data calls pertaining to restricted or unrestricted digital data. initialPlayoutBufferDelay – 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 (voiceInitialPlayoutBufferDelay) 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).packetSize <10 | 20 | 30 | 40> – The maximum data packet size (Kilobits).
preferredRtpDataPayloadType – The RTP Payload Type included in the RTP header of the data packet.
(range: 0-127 / default = 56)
|
dtls | N/A | Use this object to control DTLS-SRTP and DTLS fallback behavior in this Packet Service Profile. dtlsCryptoSuiteProfile <profile name> – Name of DTLS Crypto Suite Profile.dtlsFlags
allowDtlsFallback <disable | enable> – Enable flag to allow fallback to standard RTP when cryptographic attribute negotiation fails (default = disable).
enableDtlsSrtp <disable | enable> – Enable flag to allow Secure RTP.
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flags
| N/A | See Packet Service Profile Flags table below for flag descriptions |
honorRemotePrecedence
| N/A | Flag to set precedence of audio encoding priority order of the local packet service profile over the remote peers audio encoding priority order when creating the priority order of the audio encodings that are common to both. Options are: disable (default) – Applies precedence to local audio encoding priority order, local Secure RTP/RTCP settings and crypto suite priority order.enable – Applies precedence to the remote peer's audio encoding priority order. For ingress call legs, also applies precedence to remote peer's Secure RTP/RTCP settings and crypto suite priority order.
|
mediaPacketCos
| 0-7 | Specifies Class of Service (COS) value to be set in the IEEE 802.1D User Priority field of media packets transmitted on a call leg that uses this Packet Service Profile. This parameter only has an effect if the network interface supports 802.1Q tagged Ethernet frames. (default = 0 which is interpreted as best effort). |
packetToPacketControl
| N/A | Use this object to define the packet-to-packet control parameters: codecsAllowedForTranscoding – Use this parameter to specify codecs allowed for transcoding, and which leg to apply it to.otherLeg <codec> (see codec list below)thisLeg <codec> (see codec list below)
| amr | efr | evrc | g711a | | g711u | g722 | g726 | g729 | | g7221 | g7222 | g7231 | ilbc | | opus | t38 | | |
conditionsInAdditionToNoCommonCodec – The performs transcoding when any of the specified conditions are met, including no common codec on ingress and egress legs.applyFaxToneTreatment– Apply fax tone treatment. different2833PayloadType– Enable this option to allow SBC to transcode media when RFC2833 payload type received from ingress is different from the preferred DTMF payload type configured in egress PSP.differentDtmfRelay – Enable this flag to perform transcoding when the ingress and egress call legs use different DTMF relay methods.differentPacketSize – Enable this flag to perform transcoding when the ingress and egress call legs use different packet sizes.differentSilenceSuppression –Enable this flag to perform transcoding when the ingress and egress call legs use different silence suppression methods.honorAnswerPreference– 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).honorOfferPreference – 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 honorAnswerPreference vs. honorOfferPreference table below describing SBC behavior when this flag is enabled/disabled).
transcode – Transcode options:conditional (default)determinedByPspForOtherLegonlytranscoderFreeTransparency (NOTE: If using this option, you must also set lateMediaSupport to ‘passthru’ ( see sipTrunkGroup media - CLI))
| Div |
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| | Multiexcerpt |
|---|
| MultiExcerptName | TranscodeConditional |
|---|
| The SBC, when configured with the appropriate codecs, transcodes for differences in the codecs. Use ‘transcode conditional’ option to allow the SBC to transcode for certain conditions in addition to differences in codecs. For example, if the codecs are the same on both legs and this option is selected, by applying additional settings for the particular call scenarios, the SBC will trigger transcoding for those scenarios, such as: - Transcode for DTMF differences. For example, when both legs have same codec, but one uses inband for DTMF and the other RFC 2833.
- Transcode for Silence Suppression differences.
- Transcode for Packet Size differences.
- Transcode for RFC2833/4733 Payload differences. For example, when codecs are the same on both legs, but one side uses 101 as the Payload Type for DTMF and the other uses 102.
- Transcode always.
|
|
| Caption |
|---|
| 0 | Table |
|---|
| 1 | honorAnswerPreference vs. honorOfferPreference |
|---|
| 3 | 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). |
|
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peerAbsenceAction
| N/A | Specifies the action to take when mediaPeerInactivity timer expires (see Media System - CLI page). none – (default) Inactivity detection is disabled.peerAbsenceTrap – Choose this option to generate a trap if RTP inactivity is detected.peerAbsenceTrapAndDisconnect – Choose this option to generate a trap and tear down the call if RTP inactivity is detected.
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preferredRtpPayloadTypeForDtmfRelay | 0-128 | The Specifies the preferred RTP payload type to be set 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. | Note |
|---|
Using the default value of "128" for preferredRtpPayloadTypeForDtmfRelay implies that the preferred DTMF value (from the system configurable) is used for | | Note |
|---|
Using the default value of "128" for preferredRtpPayloadTypeForDtmfRelay implies that the preferred DTMF value (from the system configurable) is used for this profile. |
| Note |
|---|
See interworkDtmfWithoutTranscoding note above for valid values to use with this parameter when interworkDtmfWithoutTranscoding is enabledIf interworkDtmfWithoutTranscoding is enabled, ensure preferredRtpPayloadTypeForDtmfRelay is set to a valid value (96-127). If preferredRtpPayloadTypeForDtmfRelay value is invalid (set to "128"), the system may fail to pick up the value configured using "set system dspPad rtpDtmfRelay" command because DSPs are not used for the call. |
|
qosValues | N/A | Use this object to configure the Quality of Service (QoS) DSCP value for this Packet Service Profile. msrpDscp – The DSCP value to use for egressing MSRP packets (range: 0-255 / default = 0).
|
rtcpOptions
| N/A | Use this object to specify Real Time Control Protocol (RTCP) options for the call. RTCP is used to report network traffic congestion data. Various actions (for example call disconnect) may be taken when congestion threshold settings are exceeded. The RTCP options are: rtcp disable (default)enable – RTCP is used for the call.
packetLossAction – Packet loss action to take when packet threshold is exceeded:none – Take no action.packetLossTrap – Generate trap.packetLossTrapAndDisconnect – Generate trap and disconnect.
packetLossThreshold – (default = 0) 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. See Command Example below for an example configuration.
| Note |
|---|
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 mediaPeerInactivity. To set media peer inactivity timeout value, see mediaPeerInactivity parameter in Media System - CLI page. terminationForPassThrough – Specifies RTCP termination behavior for pass-through calls.disable – (default) RTCP is relayed between the end points for pass-through calls. enable – Enable flag (as well as rtcp flag) 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 terminationForPassThrough settings, RTCP is always relayed.
|
secureRtpRtcp
| N/A | Specifies whether secure RTP Real Time Control Protocol (SRTP) is enabled for the call: cryptoSuiteProfile <profile name> – A unique identifier for the crypto suite profile.flags – Possible values are disable/enable. The default value of each flag is "disable".
allowFallback– Enable flag to allow fallback to standard RTP/RTCP when crypto attribute negotiation fails.enableSrtp – Enable this flag to enable secure RTP/RTCP.resetEncDecROCOnDecKeyChange – Enable flag to reset Roll Over Counter for both encryption and decryption when decryption key changes.resetROCOnKeyChange – Enable flag to reset the SRTP Roll Over Counter when the session key changes.updateCryptoKeysOnModify– 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.
|
sendRoutePSPPrecedence
| N/A | Use this flag to allow audio encoding order preference in outgoing messages only. |
silenceFactor
| N/A | The percentage of call time that silence is expected. This parameter is used to reduce expected call bandwidth. (range: 0-50 / default = 40). |
silenceInsertionDescriptor
| N/A | Use this object to define the Silence Insertion Descriptor (SID) attributes. g711SidRtpPayloadType – 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).
|
t38
| N/A | Use this object to specify T.38 data rate attributes using following parameters: dataRateManagementType: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.
ecm – Use this flag to allocate DSP resources, when available, for T.38 Error Correction Mode (ECM) calls.disable – (default) use normal resource allocation.enable
lowSpeedNumberOfRedundantPackets – 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).maxBitRate – 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.
numberOfRedundantPackets – 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).protocolVersion – Use this parameter to specify the T.38 Fax protocol version to use.
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typeOfService
| 0-255 | Use this object to Use this parameter to set six most significant bits of Type of Service byte in an IP header for DSCP marking of MSRP packets. Default value is 0. |
videoCalls
| N/A | Use this object to define video call parameters. audioOnlyIfVideoIsPrevented– 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.codecListProfile – Name of the Codec List profile used to store precedence and purge lists of video codec MIME subtypes.ieee8021QVLanCos – IEEE-802 1Q VLAN Class of Service. (range: 0-7 / default = 0)ipv4Tos – IPv4 type of service. (range: 0-255 / default = 0)ipv6TrafficClass – IPv6 traffic class. (range: 0-255 / default = 0)
| Note |
|---|
ipv6TrafficClass is not supported with H.323 calls.
|
maxVideoBandwith – Maximum bandwidth in Kbps utilized by the video stream. (range: 0-50000 / default = 0).videoBandwidthReductionFactor – The rate in which the maximum video bandwidth is reduced when IP resource allocation is performed. (range: 0-100 / default = 0).
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voiceInitialPlayoutBufferDelay
| 1-50 | The delay (in milliseconds) by the initial playout buffer required to absorb the maximum expected data packet delay across the network. Must be in increments of 1 ms. (default = 10). |