Packet Service Profile - CLI

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).

Use this object to configure Audio Transparency settings.

• unknownCodecBitRate – The bit rate, in Kilobytes/second, required for bandwidth computation of unknown audio codecs. (range: 1-1000 KB/sec / default = 124)
• unknownCodecPacketSize – The packet size, in milliseconds, required for Bandwidth computation of unknown audio codecs. (range: 5-100 ms / default = 10)

NOTE: If the bandwidth is not configured, the unknownCodecPacketSize and unknownCodecBitRate are used for the pass-through call.

codec

codecEntry1-codecEntry12

<codecEntry ID> <codec_name> – Defines the codec entry priorities and codec names. See Codec Parameters table below for details.

Up to 12 codec configurations are supported by the SBC for PSX, and Advanced ERE deployment scenarios (refer to Routing and Policy Management for a description of the different routing configurations).

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)

Use this object to control DTLS-SRTP and DTLS fall-back behavior in this Packet Service Profile.

• dtlsCryptoSuiteProfile <profile name> – Name of  Crypto Suite Profile to attach to this Packet Service Profile.
• dtlsFlags
  
  • allowDtlsFallback <disable | enable> – Enable flag to allow fallback to standard RTP when cryptographic attribute negotiation fails (default = disable).
  • dtlsSctpRelay <disable | enable> –  Enable flag on both legs of the call to relay DTLS/SCTP streams to the endpoint (default = disable).

  • dtlsSrtpRelay <disable | enable> – Enable flag on both legs of the call to relay DTLS/SRTP audio and video streams to the endpoint. (default = disable).

  • enableDtlsSrtp <disable | enable> – Enable flag to allow Secure RTP.

NOTE: DTLS-SRTP relay is a licensed feature and requires an SRTP license to be installed on the SBC.

NOTE: When a session contains DTLS-SRTP video stream or DTLS/SCTP application stream and there is no audio stream specified, the SBC allows the session when the ingress and egress Packet Service Profiles (PSP) are configured as audio pass-through. If ICE is part of session establishment in WebRTC (WRTC) scenario, the relay mechanism implemented for DTLS-SRTP and DTLS/SCTP is supported independent of ICE processing.

Refer to Configuring SBC for WRTC for additional WebRTC configuration details.

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

peerAbsenceAction

N/A

Specifies the action to take when mediaPeerInactivity timer expires (refer to 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.

0-128

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.

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: If 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.

NOTE: The SBC uses the preferredRtpPayloadTypeForDtmfRelay configuration as the payload type for 8 kHz DTMF when not used by any other codec.

For wideband (16 kHz) DTMF, the SBC uses the next available payload type (to the configured preferredRtpPayloadTypeForDtmfRelay value).

Use this object to configure the Quality of Service (QoS) DSCP value for this Packet Service Profile.

• dtlsSctpDscp – Use this attribute to set the DSCP value in the Differentiated Services Field of the IP header for DTLS/SCTP packets that egress the SBC. (range: 0-255 / default = 0)
• msrpDscp – Use this attribute to set 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. See RTCP Options Parameters table below for details.

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 Cryptographic Suite Profile.
• flags – Possible values are disable/enable. The default value of each flag below is "disable".

  • allowFallback – Enable flag to allow fallback to standard RTP/RTCP when crypto attribute negotiation fails.

  • allowPassthru – Enable flag to allow the SBC to pass-through SRTP media without authenticating, decrypting, or encrypting it internally. The SBC will prioritize SRTP pass-through media over terminated SRTP media. When this flag is disabled, the SBC terminates all SRTP and SRTCP media for authentication, encryption, or decryption.

    NOTE: To use this flag, enableSrtp flag must be enabled.

  • 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.

• disable (default)
• enable

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).
  • disable
  • enable (default)

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.
  
  • 0 (default)
  • 3 

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.
  • disable
  • enable (default)
• codecListProfile – Name of the Codec List profile used to store precedence and purge lists of video codec MIME sub-types.
• 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 – 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).
• videoBandwidthReductionFactor – 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).

voiceInitialPlayoutBufferDelay

1-200

Use this parameter to configure the initial playout buffer delay value (in milliseconds). Select a value just slightly greater than the maximum expected packet jitter across the network.The value must be in increments of 1 ms. (default = 10).

NOTE: Using a value above 50 ms is not recommended during normal operation.