In this section:
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. 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. 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. 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. 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: Both the SBC ERE and PSX provide control over HD codec prioritization through Packet Service Profile options. ERE functionality is described in this section. The following SBC flags are configurable in the Packet Service Profile to control HD codec prioritization: On the SBC main screen, choose a path: The Packet Service Profile window is displayed. To create a new Packet Service Profile: 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. To edit an existing Packet Service Profile: To copy an existing Packet Service Profile as the basis for a new profile, Packet Service Profile Parameters Parameter Description 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: 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. 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. 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: Send Route PSPPrecedence Specifies the audio encoding order preference in outgoing messages only. 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. Specifies the maximum data packet size (Kilobits). Specifies whether to enable RTCP. 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. Enter a value of 0, or a value in the range of 400-32767 to specify the 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. Specifies the RTCP bandwidth allocated to active data senders. The value ranges from 100 to 4000, and the default value is 250. Specifies the RTCP bandwidth allocated for receivers. The value ranges from 100 to 3000, and the default value is 250. 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). 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). 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. An SBC-POL-RTU license is needed to enable more than four codecs. 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: Transcode options: Apply fax tone treatment. Enable this flag to perform transcoding when the ingress and egress call legs use different DTMF relay methods. Enable this flag to perform transcoding when the ingress and egress call legs use different packet sizes. Enable this flag to perform transcoding when the ingress and egress call legs use different silence suppression methods. 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 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). honorAnswerPreference vs. honorOfferPreference 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. 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. 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). Enable this option to disallow data calls. Specifies whether digit detection is enabled on digits sent to the network. 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. Enable this option to use direct media as needed. Enable this option to validate peer support for DTMF events. Enable this option for all peer devices that support RFC 4733. Enable this option to interwork DTMF with out-of-band RFC2833 without using transcoding. 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. Enable this flag to generate a new synchronization source (SSRC) using a random value along with a new timestamp on a new RTP stream whenever reactivating a resource (due to a change in codec or some other event). SSRC randomization reduces the probability of collision in large groups and simplifies the process of group sampling that relies on a uniform distribution of SSRCs. This flag also affects medial behavior following a switchover. Options: 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. When enabled, Note: Enable this option to allow the SBC to choose NB-NB pass-through over HD-HD transcoded call. When disabled, the SBC prefers HD-HD transcoded call over NB-NB pass-thru. Note: 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. If this option is disabled, the SBC uses existing behavior. Note: If Transcoder Free Transparency is enabled, this option is ignored. 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. Note: 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. 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. 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. 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. 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. Note: This configuration applies to all pass-through calls. It works independently from Audio Transparency feature and Specifies whether text media calls, using T.140 codec, are allowed. For more information on text codecs, refer to Text Codecs. Enable this option to determine how the SBC handles SSRCs in SRTP media flows. When enabled, the SBC: When disabled, the SBC relays the SSRC for pass-through media flows it receives from the peer. The following data rate management types are supported: 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: Use this flag to allocate DSP resources, when available, for T.38 Error Correction Mode (ECM) calls. 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. Enable this option to allow fallback to standard RTP/RTCP when crypto attribute negotiation fails. Enable this option to enable secure RTP/RTCP. Enable this option to reset the SRTP Roll Over Counter when the session key changes. Enable this option to reset the Roll Over Counter for both encryption and decryption when decryption key changes. 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 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). 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). IPv6 traffic class. (range: 0-255 / default = 0). 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. Enter the name of a crypto suite profile. Refer to Media - Packet Service Profile - DTLS. 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. When enabled, specifies a fall back to standard RTP when crypto attribute negotiation fails. When enabled, this parameter enables secure RTP. When enabled, relay of DTLS-SRTP audio and video streams is enabled on the SBC. When enabled, relay of DTLS/SCTP streams is enabled on the SBC. 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. 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. Refer to KPML DTMF Support section on the page DTMF and RTP Relay for feature details. Digit Detect Send Enabled Settings for KPML To delete a Packet Service Profile:Overview
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IMPORTANT
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Granular Control of HD Codec Offer or Answer
To View Packet Service Profiles
To Create a Packet Service Profile
(For information on the fields, see the table Packet Service Profile Parameters below)
To Edit a Packet Service Profile
To Copy a Packet Service Profile
Fields common to the Create New and Copy Windows
The default value is 47 bytes.Fields that appear only on the Copy Window Data Calls Preferred Rtp Data Payload Type 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 RTCP Options Rtcp Termination For Passthrough Specifies RTCP termination behavior for pass-through calls.
Enable RTCPFor Held Calls Packet Loss Threshold 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. Rr Bandwidth Rs Bandwidth Packet Loss Action Specifies what Packet loss action to take when packet threshold is exceeded. Packet Loss Trap And Disconnect — Generate trap and disconnect. Silence Insertion Descriptor G711Sid Rtp Payload Type Specifies the G.711 Silence Insertion Descriptor (SID) RTP payloadType. (range: 0-127 / default = 19). Heartbeat Codec
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Codec Entry1 Codec Entry2 - Entry12 Use 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 Conditions in Addition To No Common Codec The SBC performs transcoding when any of the specified conditions are met, including no common codec on ingress and egress legs.Apply Fax Tone Treatment Different DTMF Relay Different Packet Size Different Silence Suppression Honor Offer Preference Honor Answer Preference HOP Flag State HAP Flag State SBC Behavior Enable Disable Enable Enable Disable Enable 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 Digit Detect Send Enabled Use Direct Media Validate Peer Support For Dtmf Events Interwork Dtmf Without Transcoding Dscp Passthrough Ssrc Randomize HDCodec Preferred 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 Prefer NBPassthru Over HDTranscode Match Offered Codec Group If Nb Only Force Route PSPOrder Generate and Signal SSRCAnd Cname Allow Mid Call SSRCModification Reserve BW For Preferred Audio Common Codec Police On Heaviest Audio Codec Reserve BW For Preferred Audio Common CodecT140Call Allow Audio Transcode For Multi Stream Call Use this option to enable audio transcoding for multi-stream calls. Ssrc Randomize for Srtp T38 Data Rate Management Type 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 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 Secure Rtp Rtcp
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Crypto Suite Profile Enter the name of a crypto suite profile. Refer to Security Profiles - Crypto Suite Profile. Flags Allow Fallback Enable Srtp Reset ROCOn Key Change Reset Enc Dec ROCOn Dec Key Change Update Crypto Keys On Modify Video Calls Max Video Bandwidth Video Bandwidth Reduction Factor Ipv4Tos IPv4 type of service. (range: 0-255 / default = 0). Ipv6Traffic Class 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 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
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Dtls Flags Allow Dtls Fallback Enable Dtls Srtp Dtls Srtp Relay Dtls Sctp Relay Digit Detect Send Enabled Settings for KPML
Info
Ingress PSP Egress PSP KPML Subscription Leg Enable Enable Egress / Ingress Disable Enable Egress / Ingress with stream reverse Enable Disable Ingress / Egress with stream reverse Disable Disable None To Delete Packet Service Profile
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