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Overview

Use the Packet Service Profile screen 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.


Note

The PSX supports configuring up to 12 codecs in the Packet Service Profile and Preferred Packet Service Profile. The SBC supports receiving all 12 codecs from the PSX in the PSP and Preferred PSP. This applies to interworking with an external PSX (Advanced ERE deployment scenario). See Routing and Policy Management for deployment scenario details.

Additionally, the SBC supports up to 12 codecs over Gateway links to SBCs and/or GSXs.

Note

An SBC-POL-RTU license is needed to enable more than four codecs.

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.


IMPORTANT

Ribbon recommends using the Transparency Profile to configure transparency on the SBC Core for new deployments, as well as applying additional transparency configurations to existing deployments. Do not use IP Signaling Profile flags in these scenarios because the flags will be retired in upcoming releases.

Refer to the SBC SIP Transparency Implementation Guide for additional information.

Note

Avoid using both the Silence Suppression (SS) and non-SS variant of the same codec in one Packet Service Profile (PSP) because doing so can lead to extra offer-anwer handshakes and trigger a race condition while the SBC attempts to identify a common codec instead of simply transcoding the call.

Example using G.729A on the ingress trunk group, and using G.729AB on the egress trunk group to avoid this situation:

  • Ingress PSP: 711, 729A
  • Egress PSP: 711, 729AB
  • Transcode : Conditional
  • Conditions In Addition To No Common Codec : Different Silence Suppression,
  • Ingress This Leg: 711, 729
  • Egress This Leg: 711, 729

Granular Control of HD Codec Offer or Answer

Both the SBC ERE and PSX provide control over HD codec prioritization through Packet Service Profile options. ERE functionality is described in this section.

  • Prioritization of an HD codec over a non-HD codec in the offer when the peer supports both HD and NB codecs, even if transcoding is required
  • Prioritization of a HD codec over a non-HD codec in the answer
  • Support of a narrowband (NB) codec list only toward a peer supporting HD and NB codecs, for backward compatibility

The following SBC flags are configurable in the Packet Service Profile to control HD codec prioritization: 

  • HDCodec Preferred
  • Prefer NBPassthru Over HDTranscode
  • Match Offered Codec Group If Nb Only
  • Force Route PSPOrder

To View Packet Service Profiles

On the SBC main screen, choose a path:

  • Configuration > Profile Management > Category: Media Profiles > Packet Service Profile 
  • Configuration > System Provisioning > Category: Trunk Provisioning > Packet Service Profile 
  • All > Profiles > Media > Packet Service Profile

The Packet Service Profile window is displayed.

To Create a Packet Service Profile

To create a new Packet Service Profile:

  1. 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.
    (For information on the fields, see the table Packet Service Profile Parameters below)


  2. Click Save.
  3. You can then configure many additional options in the profile using the windows positioned below the Packet Service Profile window in the EMA navigation hierarchy. Select the Packet Service Profile by name at the top of each subordinate window to continue configuring that profile. Refer to the following pages for information on using these windows to continue configuring a new Packet Service profile.

To Edit a Packet Service Profile

To edit an existing Packet Service Profile:

  1. Click the radio button next to the profile you want to edit. The Edit Selected Packet Service Profile window opens. You can edit the set of options that are shown when a profile is created. 
  2. Click Save.
  3. You can edit additional options in the profile using the windows positioned below the Packet Service Profile window in the EMA navigation hierarchy. Refer to the pages listed above for information on using these windows to continue editing a Packet Service profile.

To Copy a Packet Service Profile

To copy an existing Packet Service Profile as the basis for a new profile,

  1. Click the radio button next to the profile you want to copy.
  2. Click Copy Packet Service Profile. The Copy Selected Packet Service Profile window opens. As shown in the following figure, you can specify a  name for the copy and edit the full set of profile options.
  3. Configure the profile copy, as needed, and then click Save.



Packet Service Profile Parameters

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.

NOTE:

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.

NOTE:

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.

NOTE:

The 

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 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 

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 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.

  • 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 DelayUsed 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).

  • 10
  • 20
  • 30
  • 40
RTCP Options
Rtcp

Specifies whether to enable RTCP.

  • Disable (default)
  • Enable – RTCP is used for the call.
Termination For Passthrough
Specifies RTCP termination behavior for pass-through calls.
  • 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.

NOTE:

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.


  • Disabled (default)
  • Enabled
NOTE:

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. 

NOTE:

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.

NOTE:

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.

NOTE:

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.

NOTE:

This parameter is visible only when Rtcp is enabled.

Packet Loss Action
Specifies what Packet loss action to take when packet threshold is exceeded.
  • None — Take no action.
  • Packet Loss Trap — Generate trap.
  • Packet Loss Trap And Disconnect — Generate trap and disconnect.

NOTE:

This parameter is visible only when Rtcp is enabled.

 Silence Insertion Descriptor
G711Sid Rtp Payload TypeSpecifies 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).

  • 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).


Note

The PSX supports configuring up to 12 codecs in the Packet Service Profile and Preferred Packet Service Profile. The SBC supports receiving all 12 codecs from the PSX in the PSP and Preferred PSP. This applies to interworking with an external PSX (Advanced ERE deployment scenario). See Routing and Policy Management for deployment scenario details.

Additionally, the SBC supports up to 12 codecs over Gateway links to SBCs and/or GSXs.

Note

An SBC-POL-RTU license is needed to enable more than four codecs.

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 - Entry12Use these attributes, as needed, to specify codec entries with priority 2-12, respectively. For each codec entry, select the codec entry IDs as configured for Codec Entry1.
 Packet to Packet Control
 Transcode

 Transcode options:

  • Conditional (default)
  • Determined By Psp For Other Leg
  • Only
  • Transcoder Free Transparency

NOTE:
If you configure Transcoder Free Transparency, ensure you also set Late Media Support to Passthru (refer to Trunk Group - SIP Trunk Group and SIP Trunk Group - Media).


Conditions in Addition To No Common Codec
 The 
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 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.

  • Disable (default)
  • Enable
Different DTMF Relay

Enable this flag to perform transcoding when the ingress and egress call legs use different DTMF relay methods.

  • Disable (default)
  • Enable
Different Packet Size

 Enable this flag to perform transcoding when the ingress and egress call legs use different packet sizes. 

 

  • Disable (default)
  • Enable

 

Different Silence Suppression

Enable this flag to perform transcoding when the ingress and egress call legs use different silence suppression methods.

  • Disable (default)
  • Enable
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).

  • 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 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).

  • Disable (default)
  • Enable

honorAnswerPreference vs. honorOfferPreference 

HOP Flag StateHAP Flag StateSBC Behavior
EnableDisable

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.

EnableEnable

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.

DisableEnable

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 TranscodingUse 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)
amrefrevrcg711a
g711ug722g726g729
g7221g7222g7231ilbc
t38


Flags
Disallow Data Calls

Enable this option to disallow data calls.

  • Disable (default)
  • Enable
Digit Detect Send Enabled

Specifies whether digit detection is enabled on digits sent to the network.

  • 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 this option to use direct media as needed.

  • Disable (default)
  • Enable
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.

  • Disable (default) – 
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     does not validate the presence of DTMF events in the offer or answer from the peer. If DTMF relay is enabled, 
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     transmits DTMF digits received from the other leg to this peer using the named event RTP payload.
  • Enable – 
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     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), 
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     forwards the events in the egress leg to this peer using the named event RTP payload. When 
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     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.

  • Disable (default)
  • Enable
NOTE:
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.

  • 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.

  • Disable (default)
  • Enable
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.

  • Disable (default)
  • Enable

When enabled,

If...Then...
the ingress Offer contains any valid HD codecsHD codecs are sorted to the top of the list while sending out the Offer.
all NB codecs are presentSBC reorders the codec entries with NB first, followed by HD codecs.
Codec selection priority from Answer message
  1. HD-HD pass-through
  2. HD-HD transcoding
  3. NB-NB pass-through
  4. NB-NB transcoding
  5. 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.

  • Disable (default)
  • Enable

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.

  • Disable (default)
  • Enable

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.

  • Disable (default)
  • Enable

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:

  • Disable (default)
  • Enable

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:

  • Disable (default)
  • Enable

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.

  • Disable (default)
  • Enable

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.

  • Disable (default)
  • Enable

Note: This configuration applies to all pass-through calls. It works independently from Audio Transparency feature and Reserve BW For Preferred Audio Common Codec flag.

T140Call

Specifies whether text media calls, using T.140 codec, are allowed.

  • Disable (default)
  • Enable

For more information on text codecs, refer to Text Codecs.

Allow Audio Transcode For Multi Stream CallUse this option to enable audio transcoding for multi-stream calls.  
  • Disable (default)
  • Enable
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.

  • Disable (default)
  • Enable
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 PacketsUse 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
Note

Within a Packet Service Profile (PSP), configure either DTLS (Datagram Transport Layer Security) parameters or SRTP (Secure Real-Time Transport Protocol) parameters, but not both.

Crypto Suite ProfileEnter 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.

  • Disable (default)
  • Enable
Enable Srtp

Enable this option to enable secure RTP/RTCP.

  • Disable (default)
  • Enable
Reset ROCOn Key Change

Enable this option to reset the SRTP Roll Over Counter when the session key changes.

  • Disable (default)
  • Enable
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.

  • Disable (default)
  • Enable
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.

  • Disable (default)
  • Enable
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).

Ipv4TosIPv4 type of service. (range: 0-255 / default = 0).
Ipv6Traffic Class

IPv6 traffic class. (range: 0-255 / default = 0).

NOTE:

IPv6Traffic Class is not supported with H.323 calls.

Ieee8021QVLan CosIEEE-802 1Q VLAN Class of Service. (range: 0-7 / default = 0)
Codec List ProfileName 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.

  • Disable
  • Enable (default)
Audio Transparency
Unknown Codec Packet SizeSpecifies the bit rate, in Kilobytes/second, required for bandwidth computation of unknown audio codecs. (range: 1-1000 KB/sec / default = 124)
Unknown Codec Bit RateSpecifies the packet size, in milliseconds, required for Bandwidth computation of unknown audio codecs. (range: 5-100 ms / default = 10)
DTLS
 
Note

Within a Packet Service Profile (PSP), configure either DTLS (Datagram Transport Layer Security) parameters or SRTP (Secure Real-Time Transport Protocol) parameters, but not both.

Dtls Crypto Suite ProfileEnter 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.

  • Disable (default)
  • Enable
Enable Dtls Srtp

When enabled, this parameter enables secure RTP.

  • Disable (default)
  • Enable
Dtls Srtp Relay

When enabled, relay of DTLS-SRTP audio and video streams is enabled on the SBC.

  • Disable (default)
  • Enable
Dtls Sctp Relay

When enabled, relay of DTLS/SCTP streams is enabled on the SBC.

  • 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.

Digit Detect Send Enabled Settings for KPML

Use the following table for guidance in setting the Digit Detect Send Enabled flag on each PSP leg to achieve the desired Key Press Markup Language (KPML) functionality.

Info

Refer to  KPML DTMF Support section on the page DTMF and RTP Relay for feature details.

Digit Detect Send Enabled Settings for KPML

Ingress PSPEgress PSPKPML Subscription Leg
EnableEnableEgress / Ingress
DisableEnableEgress / Ingress with stream reverse
EnableDisableIngress / Egress with stream reverse
DisableDisableNone

To Delete Packet Service Profile

To delete a Packet Service Profile:

  1. Click the radio button next to the Packet Service Profile that you want to delete.
  2. Click the Delete icon (X) at the end of the row.
  3. Confirm the deletion when prompted.



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