TR-106 Data Model Template for TR-069-Enabled Devices BBF TR TR-104 Provisioning Parameters for VoIP CPE BBF TR RFC 3261 SIP: Session Initiation Protocol IETF RFC http://www.ietf.org/rfc/rfc3261.txt RFC 3435 Media Gateway Control Protocol (MGCP) Version 1.0 IETF RFC http://www.ietf.org/rfc/rfc3435.txt RFC 3550 RTP: A Transport Protocol for Real-Time Applications IETF RFC http://www.ietf.org/rfc/rfc3550.txt RFC 2198 RTP Payload for Redundant Audio Data IETF RFC http://www.ietf.org/rfc/rfc2198.txt?number=2198 RFC 2833 RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals IETF RFC http://www.ietf.org/rfc/rfc2833.txt?number=2833 RFC 2833 Codes for the Representation of Names of Countries and Their Subdivisions ISO ITU-T H.235.1 H.323 security framework: Baseline security profile ITU-T http://www.itu.int/rec/T-REC-H.235.1/en Number of entries in VoiceService table. The top-level object for VoIP CPE. Number of instances of VoiceProfile. The overall capabilities of the VoIP CPE. Maximum total number of distinct voice profiles supported. Maximum total number of lines supported across all profiles. This parameter is applicable only for a VoIP endpoint. Maximum number of voice sessions supported for any given line across all profiles. A value greater than one indicates support for CPE provided conference calling. This parameter is applicable only for a VoIP endpoint. Maximum total number of voice sessions supported across all lines and profiles. (This might differ from {{param|MaxLineCount}} if each line can support more than one session for CPE provided conference calling. This value MAY be less than the product of {{param|MaxLineCount}} and {{param|MaxSessionsPerLine}}.) {{list}} Each item is a supported signaling protocol. {{pattern}} Each entry MAY be appended with a version indicator in the form "/X.Y". For example: "SIP/2.0" The list MAY include vendor-specific protocols, which MUST be in the format defined in {{bibref|TR-106}}. For example: "X_EXAMPLE-COM_MyProt" {{list}} Each item is a geographic region supported by the CPE. Each item is the list MUST be an alpha-2 (two-character alphabetic) country code as specified by {{bibref|ISO3166}}. If {{empty}} indicates that the CPE does not support region-based customization via {{object|.VoiceProfile.{i}.}}. Support for RTCP. If {{true}} indicates support for {{object|.VoiceProfile.{i}.RTP.RTCP.}}. This parameter is applicable only for a VoIP endpoint. Support for SRTP. If {{true}} indicates support for {{object|.VoiceProfile.{i}.RTP.SRTP.}}. If {{true}} also indicates that {{param|SRTPKeyingMethods}} and {{param|SRTPEncryptionKeySizes}} are present. This parameter is applicable only for a VoIP endpoint. {{list}} Each item is a keying protocol supported by this endpoint for SRTP. {{enum}} This list MAY include vendor-specific keying methods, which MUST use the format defined in {{bibref|TR-106}}. This parameter is applicable only if {{param|SRTP}} is {{true}}. {{list}} Each item is a supported SRTP encryption key size. This parameter is applicable only if {{param|SRTP}} is {{true}}. Support for RTP payload redundancy as defined in {{bibref|RFC2198}}. If {{true}} indicates support for {{object|.VoiceProfile.{i}.RTP.Redundancy.}}. This parameter is applicable only for a VoIP endpoint. If {{true}} indicates that the CPE is constrained such that transmitted call control packets use the same DSCP marking as transmitted RTP packets. If {{true}}, the CPE MUST NOT support {{param|.VoiceProfile.{i}.MGCP.DSCPMark}}, {{param|.VoiceProfile.{i}.H323.DSCPMark}}, or {{param|.VoiceProfile.{i}.SIP.DSCPMark}} for call control. This parameter is applicable only for a VoIP endpoint. If {{true}} indicates that the CPE is constrained such that transmitted call control packets use the same Ethernet tagging (VLAN ID Ethernet Priority) as transmitted RTP packets. If {{true}}, the CPE MUST NOT support the {{param|.VoiceProfile.{i}.MGCP.VLANIDMark}}, {{param|.VoiceProfile.{i}.H323.VLANIDMark}}, {{param|.VoiceProfile.{i}.SIP.VLANIDMark}} {{param|.VoiceProfile.{i}.MGCP.EthernetPriorityMark}}, {{param|.VoiceProfile.{i}.H323.EthernetPriorityMark}}, or {{param|.VoiceProfile.{i}.SIP.EthernetPriorityMark}} for call control. This parameter is applicable only for a VoIP endpoint. If {{true}} indicates the CPE is capable of supporting the PSO_Activate Facility Action, which allows a call to be switched to a PSTN FXO (Foreign eXchange Office) line. This parameter is applicable only for a VoIP endpoint. Support for T.38 fax. If {{true}} indicates support for {{object|.VoiceProfile.{i}.FaxT38.}}. This parameter is applicable only for a VoIP endpoint. Support for fax pass-through. If {{true}} indicates support for {{param|.VoiceProfile.{i}.FaxPassThrough}}. This parameter is applicable only for a VoIP endpoint. Support for modem pass-through. If {{true}} indicates support for the {{param|.VoiceProfile.{i}.ModemPassThrough}}. This parameter is applicable only for a VoIP endpoint. Support for tone generation. If {{true}} indicates support for {{object|.VoiceProfile.{i}.Tone.}}. If {{true}} also indicates that {{param|ToneDescriptionsEditable}}, {{param|PatternBasedToneGeneration}}, and {{param|FileBasedToneGeneration}} are present. This parameter is applicable only for a VoIP endpoint. If {{true}} indicates that {{object|.VoiceProfile.{i}.Tone.Description.}} and {{object|.VoiceProfile.{i}.Tone.Pattern.}} are editable (if entries can be added, removed, or modified). This parameter is applicable only if {{param|ToneGeneration}} is {{true}}. Support for tone generation by pattern specification. If {{true}} indicates support for {{object|.VoiceProfile.{i}.Tone.}}. If {{param|ToneGeneration}} is {{true}}, at least one of {{param}} and {{param|FileBasedToneGeneration}} MUST also be {{true}}. This parameter is applicable only if {{param|ToneGeneration}} is {{true}}. Support for tone generation by file playback. If {{true}} indicates support for {{object|.VoiceProfile.{i}.Tone.}}. If {{true}} also indicates that {{param|ToneFileFormats}} is present. If {{param|ToneGeneration}} is {{true}}, at least one of {{param|PatternBasedToneGeneration}} and {{param}} MUST also be {{true}}. This parameter is applicable only if {{param|ToneGeneration}} is {{true}}. {{list}} Each item is a supported tone file format. The specified file formats are raw codec data files, using one of the codecs listed below. {{enum}} The list MAY include vendor-specific -specific extensions, which MUST use the format defined in {{bibref|TR-106}}. Example: "G.711MuLaw, MP3, X_EXAMPLE-COM_MyFileFormat" If the CPE does not support tone files, this parameter MUST be {{empty}}. This parameter is applicable only if {{param|FileBasedToneGeneration}} is {{true}}. Support for ring generation. If {{true}} indicates support for control of ring generation via {{object|.VoiceProfile.{i}.Line.{i}.Ringer.}}. If {{true}} also indicates that {{param|RingDescriptionsEditable}}, {{param|PatternBasedRingGeneration}}, and {{param|FileBasedRingGeneration}} are present. This parameter is applicable only for a VoIP endpoint. If {{true}} indicates that {{object|.VoiceProfile.{i}.Line.{i}.Ringer.Description.}} and {{object|.VoiceProfile.{i}.Line.{i}.Ringer.Pattern.}} are editable (if entries can be added, removed, or modified). This parameter is applicable only if {{param|RingGeneration}} is {{true}}. Support for ring generation by pattern specification. If {{true}} indicates support for {{object|.VoiceProfile.{i}.Line.{i}.Ringer.Pattern.}}. If {{true}} also indicates that {{param|RingPatternEditable}} is present. This parameter is applicable only if {{param|RingGeneration}} is {{true}}. If {{true}} indicates that {{object|.VoiceProfile.{i}.Line.{i}.Ringer.Pattern.}} is editable (if entries can be added, removed, or modified). This parameter is applicable only if {{param|PatternBasedRingGeneration}} is {{true}}. Support for ring generation by file playback. If {{true}} indicates support for specification of ringer files in {{object|.VoiceProfile.{i}.Line.{i}.Ringer.Description.}}. If {{true}} also indicates that {{param|RingFileFormats}} is present. This parameter is applicable only if {{param|RingGeneration}} is {{true}}. {{list}} Each item is a supported ring file format. {{enum}} The list MAY include vendor-specific-specific extensions, which MUST use the format defined in {{bibref|TR-106}}. Example: "MIDI, AMR, X_EXAMPLE-COM_MyFileFormat" If the CPE does not support ring files, this parameter MUST be {{empty}}. This parameter is applicable only if {{param|FileBasedRingGeneration}} is {{true}}. MMF RTTTL or RTX Support for a configurable digit map string. If {{true}} indicates full support for {{param|.VoiceProfile.{i}.DigitMap}}. Support for a configurable numbering plan. If {{true}} indicates support for a configurable numbering plan via {{object|.VoiceProfile.{i}.NumberingPlan.}}. This parameter is applicable only for a VoIP endpoint. Support for a configurable button map. If {{true}} indicates support for a configurable button map via {{object|.VoiceProfile.{i}.ButtonMap}}. This parameter is applicable only for a VoIP endpoint. Support for remotely accessible voice-port tests. If {{true}} indicates support for {{object|.PhyInterface.{i}.Tests.}}. This parameter is applicable only for a VoIP endpoint. SIP-specific capabilities. Applicable only if the value of {{param|.Capabilities.SignalingProtocols}} includes {{pattern|SIP|.Capabilities.SignalingProtocols}}. The role of this VoIP CPE. {{enum}} A single VoiceService instance MUST have only one role. If a device includes the capabilities for more than one role, each role MUST be represented as separate VoiceService instances. {{list}} Each item is a supported SIP extension method. SIP extension methods MUST be in the form of the method name in upper case. The list MAY include vendor-specific extensions, which MUST use the format defined in {{bibref|TR-106}}. Examples: : "REFER" : "INFO" : "X_EXAMPLE-COM_MyExt" {{list}} Each item is a supported SIP transport protocol. {{enum}} The list MAY include vendor-specific transports, which MUST use the format defined in {{bibref|TR-106}}. {{list}} Each item is a supported URI scheme beyond the URI schemes required by the SIP specification. Each URI scheme is given by the URI prefix, without the colon separator. Example: "tel, fax" Support for SIP event subscription. If {{true}} value indicates support for {{object|.VoiceProfile.{i}.SIP.EventSubscribe.}} and {{object|.VoiceProfile.{i}.Line.{i}.SIP.EventSubscribe.{i}.}}. Support for SIP response map. If {{true}} indicates support for {{object|.VoiceProfile.{i}.SIP.ResponseMap.}}. This parameter is applicable only for a VoIP endpoint. {{list}} Each item is a supported authentication protocol for TLS transport. {{enum}} The list MAY include vendor-specific protocols, which MUST use the format defined in {{bibref|TR-106}}. Support for this parameter is applicable only if the value of {{param|Transports}} includes {{enum|TLS|Transports}}. {{list}} Each item represents a supported TLS authentication key size. Support for this parameter is applicable only if the value of {{param|Transports}} includes {{enum|TLS|Transports}} and {{param|TLSAuthenticationProtocols}} is present and non-empty and includes at least one value other than {{enum|Null|TLSAuthenticationProtocols}}. {{list}} Each item is a supported authentication protocol for TLS transport. {{enum}} The list MAY include vendor-specific protocols, which MUST use the format defined in {{bibref|TR-106}}. Support for this parameter is applicable only if the value of {{param|Transports}} includes "{{enum|TLS|Transports}}. {{list}} Each item is a supported TLS encryption key size. Support for this parameter is applicable only if the value of {{param|Transports}} includes {{enum|TLS|Transports}} and {{param|TLSEncryptionProtocols}} is present and non-empty and includes at least one value other than {{enum|Null|TLSEncryptionProtocols}}. {{list}} Each item is a supported authentication protocol for TLS transport. {{enum}} The list MAY include vendor-specific protocols, which MUST use the format defined in {{bibref|TR-106}}. Support for this parameter is applicable only if {{param|Transports}} includes the value {{enum|TLS|Transports}} and {param|TLSEncryptionProtocols}} is present and non-empty and includes at least one value other than {{enum|Null|TLSEncryptionProtocols}}. MGCP-specific capabilities. Applicable only if the value of {{param|.Capabilities.SignalingProtocols}} includes the value {{pattern|MGCP|.Capabilities.SignalingProtocols}} or {{pattern|MGCP-NCS|.Capabilities.SignalingProtocols}}. {{list}} Each item is a supported optional MGCP package. MGCP packages are listed using the uppercase package abbreviation. The list MAY include vendor-specific extensions, which MUST use the format defined in {{bibref|TR-106}}. Examples: : "BP" : "X_EXAMPLE-COM_MyExt" H.323-specific capabilities. Applicable only if the value of {{param|.Capabilities.SignalingProtocols}} includes {{pattern|H\.323|.Capabilities.SignalingProtocols}}. Support for H323 fast start. If {{true}} indicates support for fast start. {{list}} Each item is a supported authentication method. {{enum}} The list MAY include vendor-specific protocols, which MUST use the format defined in {{bibref|TR-106}}. Diffie-Hellman password with symmetric encryption password with hashing certificate with signature IPSEC based connection TLS Table to describe the set of supported codecs. The table MUST include a distinct entry for each supported combination of these {{param|Codec}} and {{param|BitRate}}. Applicable only for a VoIP endpoint. Unique identifier for each entry in this table. Identifier of the type of codec. {{enum}} The parameter MAY instead be a vendor-specific codec, which MUST be in the format defined in {{bibref|TR-106}}. For example: "X_EXAMPLE-COM_MyCodec" Bit rate, specified in {{units}}. The value MUST be among the values appropriate for the specified codec. {{list}} Each item is a supported packetization period, in milliseconds, or a continuous range of packetization periods. Ranges are indicated as a hyphen-separated pair of unsigned integers. Examples: : "20" indicates a single discrete value. : "10, 20, 30" indicates a set of discrete values. : "5-40" indicates a continuous inclusive range. : "5-10, 20, 30" indicates a continuous range in addition to a set of discrete values. A range MUST only be indicated if all values within the range are supported. If {{true}} indicates support for silence suppression for this codec. Object associated with a collection of voice lines with common characteristics. Support for adding and removing profiles is conditional on whether more than one profile is supported as indicated by {{param|.Capabilities.MaxProfileCount}}. By default, a single VoiceProfile object SHOULD be present in a VoiceService, initially in the disabled state. Enables or disables all lines in this profile, or places them into a quiescent state. {{enum}} If the value is {{enum|Quiescent}}, in-progress sessions remain intact, but no new sessions are allowed. If the value is set to {{enum|Quiescent}} in a CPE that does not support {{enum|Quiescent}}, the CPE MUST treat it as if the value is {{enum|Disabled}}. When written as {{true}}, forces the all lines in this profile to be reset, causing it to re-initialize and perform all start-up actions such as registration. Always {{false}} when read. Number of instances of {{object|.VoiceProfile.{i}.Line.{i}.}} within this {{object}}. Applicable only for a VoIP endpoint. Human-readable string to identify the profile instance. The protocol to be used for this profile. Limit on the number of simultaneous voice sessions across all lines in this {{object}}. Must be less than or equal to {{param|.Capabilities.MaxSessionCount}}. (This MAY be greater than {{param|NumberOfLines}} if each line can support more than one session, for example for CPE provided conference calling.) Method by which DTMF digits MUST be passed. {{enum}} If {{param|DTMFMethodG711}} is non-empty, then this parameter applies only when the current codec is not G.711. This parameter is applicable only for a VoIP endpoint. Applicable only if the value of {{param|SignalingProtocol}} is {{pattern|SIP|SignalingProtocol}}. Method by which DTMF digits MUST be passed if the the current codec is G.711. {{enum}} If {{empty}} indicates that the value of {{param|DTMFMethod}} is to apply whether or not the the the current codec is G.711. This parameter is applicable only for a VoIP endpoint. Applicable only if the value of {{param|SignalingProtocol}} is {{pattern|SIP|SignalingProtocol}}. The geographic region associated with this profile. This MAY be used by the CPE to customize localization settings. If {{empty}} indicates that the region is unspecified and the CPE SHOULD use default localization settings. This parameter is applicable only if {{param|.Capabilities.Regions}} is non-empty. Digit map controlling the transmission of dialed digit information. The string defines the criteria to be met as digits are collected before an outgoing request (e.g., a SIP INVITE) can be initiated. The syntax of this parameter is exactly the syntax used by MGCP as defined in {{bibref|RFC3435|Section 2.1.5}}. This parameter is applicable only if the device supports a dialing mechanism for which a dialing plan is needed (for example, a device with an explicit Dial button may not need to be aware of the dialing plan) and if the device does not already support a dialing plan mechanism for this profile (e.g., in-band via MGCP). This object is supported only if {{param|.Capabilities.DigitMap}} is {{true}}. Applicable only for a VoIP endpoint. Enables use of {{param|DigitMap}}. When {{true}}, {{object|.VoiceProfile.{i}.NumberingPlan.}}, if present, MUST be ignored. This parameter is required if and only if both {{param|DigitMap}} and {{object|.VoiceProfile.{i}.NumberingPlan}} are present. Applicable only for a VoIP endpoint. Enable or disable use of STUN to allow operation through NAT. Note: enabling STUN is to be interpreted as enabling the use of STUN for discovery, not use as a keep-alive mechanism. Domain name or IP address of the STUN server. For bandwidth-based admission control, indicates the amount of upstream bandwidth, in {{units}}, that must be left available for non-voice traffic when determining whether a session can proceed. This parameter is appropriate only in implementations in which the actual bandwidth can be known, such as a VoIP device embedded in a DSL B-NT. For bandwidth-based admission control, indicates the amount of downstream bandwidth, in {{units}}, that must be left available for non-voice traffic when determining whether a session can proceed. This parameter is appropriate only in implementations in which the actual bandwidth can be known, such as a VoIP device embedded in a DSL B-NT. Specifies whether or not the CPE SHOULD fail over to PSTN service, if available, on loss of connectivity to the VoIP service. This parameter is appropriate only in implementations in which PSTN fail-over is possible. Specifies the behavior of the CPE for pass-through of fax data. {{enum}} If this parameter is supported, the value of {{param|.Capabilities.FaxPassThrough}} MUST be {{true}}. This parameter is appropriate only for a VoIP endpoint. Prevents the CPE from switching to a fax pass-through mode. Allows the CPE to automatically detect fax data to determine whether or not to switch to a fax pass-through mode. Forces the CPE to switch to a fax pass-through mode regardless of whether fax signaling is detected. Specifies the behavior of the CPE for pass-through of modem data. {{enum}} If this parameter is supported, the value of {{param|.Capabilities.ModemPassThrough}} MUST be {{true}}. This parameter is appropriate only for a VoIP endpoint. Prevents the CPE from switching to a modem pass-through mode. Allows the CPE to automatically detect modem data to determine whether or not to switch to a modem pass-through mode. Forces the CPE to switch to a modem pass-through mode regardless of whether modem signaling is detected. Information regarding the organization providing service for this voice profile instance. Human-readable string identifying the service provider. URL of the service provider for this profile instance. Phone number to contact the service provider for this profile instance. Email address to contact the service provider for this profile instance. Voice profile parameters that are specific to SIP user agents. Creation of this object occurs on specification of {{pattern|SIP|.VoiceProfile.{i}.SignalingProtocol}}. Host name or IP address of the SIP proxy server. All SIP signaling traffic MUST be sent to the host indicated by this parameter and the port indicated by {{param|ProxyServerPort}} unless {{param|OutboundProxy}} is non-empty or a different route was discovered during normal operations SIP routing operation. Regardless of which host the traffic gets sent to ({{param}} or {{param|OutboundProxy}}), the value of this parameter MUST be used to derive the URI placed into the SIP Route header field of all requests originated by this end-point unless a different proxy host was discovered dynamically during normal SIP routing operations. Destination port to be used in connecting to the SIP server. Transport protocol to be used in connecting to the SIP server. Host name or IP address of the SIP registrar server. If this parameter is {{empty}}, the CPE MUST obtain all of the registrar server configuration information, including host name or IP address, port, and transport protocol, from the values in {{param|ProxyServer}}, {{param|ProxyServerPort}}, and {{param|ProxyServerTransport}} and MUST ignore the values in {{param}}, {{param|RegistrarServerPort}} and {{param|RegistrarServerTransport}}. Destination port to be used in connecting to the SIP registrar server. If {{param|RegistrarServer}} is empty the CPE MUST obtain all of the registrar server configuration information, including host name or IP address, port, and transport protocol, from the values in {{param|ProxyServer}}, {{param|ProxyServerPort}}, and {{param|ProxyServerTransport}} and MUST ignore the values in {{param|RegistrarServer}}, {{param}} and {{param|RegistrarServerTransport}}. Transport protocol to be used in connecting to the registrar server.{{enum}} If {{param|RegistrarServer}} is empty the CPE MUST obtain all of the registrar server configuration information, including host name or IP address, port, and transport protocol, from the values in {{param|ProxyServer}}, {{param|ProxyServerPort}}, and {{param|ProxyServerTransport}} and MUST ignore the values in {{param|RegistrarServer}}, {{param|RegistrarServerPort}} and {{param}}. CPE domain string. If {{empty}}, the CPE SHOULD use its IP address as the domain. Port used for incoming call control signaling. Transport protocol to be used for incoming call control signaling. Host name or IP address of the outbound proxy. If the value is not {{empty}}, the SIP endpoint MUST send all SIP traffic (requests and responses) to the host indicated by this parameter and the port indicated by {{param|OutboundProxyPort}}. This MUST be done regardless of the routes discovered using normal SIP operations, including use of Route headers initialized from Service-Route and Record-Route headers previously received. The OutboundProxy value is NOT used to generate the URI placed into the Route header of any requests. Destination port to be used in connecting to the outbound proxy. This parameter MUST be ignored unless the value of {{param|OutboundProxy}} is non-empty. Text string to be used in the Organization header. Period over which the user agent must periodically register, in {{units}}. Value of SIP timer T1, in {{units}}, as defined in {{bibref|RFC3261}}. Value of SIP timer T2, in {{units}}, as defined in {{bibref|RFC3261}}. Value of SIP timer T4, in {{units}}, as defined in {{bibref|RFC3261}}. Value of SIP timer A, in {{units}}, as defined in {{bibref|RFC3261}}. Value of SIP timer B, in {{units}}, as defined in {{bibref|RFC3261}}. Value of SIP timer C, in {{units}}, as defined in {{bibref|RFC3261}}. Value of SIP timer D, in {{units}}, as defined in {{bibref|RFC3261}}. Value of SIP timer E, in {{units}}, as defined in {{bibref|RFC3261}}. Value of SIP timer F, in {{units}}, as defined in {{bibref|RFC3261}}. Value of SIP timer G, in {{units}}, as defined in {{bibref|RFC3261}}. Value of SIP timer H, in {{units}}, as defined in {{bibref|RFC3261}}. Value of SIP timer I, in {{units}}, as defined in {{bibref|RFC3261}}. Value of SIP timer J, in {{units}}, as defined in {{bibref|RFC3261}}. Value of SIP timer K, in {{units}}, as defined in {{bibref|RFC3261}}. Invite request Expires header value, in {{units}}. Re-invite request Expires header value, in {{units}}. Register request Expires header value, in {{units}}. Register request Min-Expires header value, in {{units}}. Register retry interval, in {{units}}. Type of inbound authentication, if any, required. If inbound authentication is required, the username credentials. If inbound authentication is required, the password credentials. When {{true}}, in the SDP included in an OK response to an Invite, the first listed codec MUST be the highest priority codec among those offered in the Invite, based on the priorities specified in {{object|.VoiceProfile.{i}.Line.{i}.Codec.List.{i}.}}. The list of codecs in the SDP MAY also include other lower priority codecs. When {{false}}, there is no specific requirement for choosing the codecs listed in the SDP included in an OK response. Diffserv code point to be used for outgoing SIP signaling packets. VLAN ID (as defined in {{bibref|802.1Q-2005}}) to be used for outgoing SIP signaling packets for this profile. A value of -1 indicates the default value is to be used. If either {{param}} or {{param|EthernetPriorityMark}} are greater than zero, then the outgoing frames MUST be tagged. Otherwise, the outgoing frames MAY be tagged or untagged. Ethernet priority code (as defined in {{bibref|802.1Q-2005}}) to be used for outgoing SIP signaling packets for this profile. A value of -1 indicates the default value is to be used. If either {{param|VLANIDMark}} or {{param}} are greater than zero, then the outgoing frames MUST be tagged. Otherwise, the outgoing frames MAY be tagged or untagged. Indicates the number of EventSubscribe objects. Indicates the number of SIPResponseMap objects. Table to specify the SIP events to which the CPE MUST subscribe. If supported, the value of {{param|.Capabilities.SIP.EventSubscription}} MUST be {{true}} and {{param|.VoiceProfile.{i}.SIP.SIPEventSubscribeNumberOfElements}} MUST be present. SIP event name to appear in the EVENT header of the SIP SUBSCRIBE request. Host name or IP address of the event notify server. Destination port to be used in connecting to the event notifier. Transport protocol to be used in connecting to the event notifier. Subscription refresh timer, in {{units}}. Each entry in this table specifies the tone and message to be provided to the user for a particular SIP Response received (normally 4xx and 5xx). If supported, the value of {{param|.Capabilities.SIP.ResponseMap}} MUST be {{true}} and {{param|.VoiceProfile.{i}.SIP.SIPResponseMapNumberOfElements}} MUST be present. Applicable only for a VoIP endpoint. The SIP Response code number. The message to be provided on the screen or display of the VoIP device when the SIP response is received. If this parameter is not {{empty}}, display of this text preempts the value of {{param|.VoiceProfile.{i}.Tone.Description.{i}.ToneText}} associated with {{param|Tone}}. If this parameter is {{empty}}, the value of {{param|.VoiceProfile.{i}.Tone.Description.{i}.ToneText}} associated with {{param|Tone}}, if any, is displayed instead. This parameter is applicable only for VoIP devices capable of text display. The tone to be played to the user when the SIP response is received. The value corresponds to EntryID of an entry in {{object|.VoiceProfile.{i}.Tone.Description.}}. A value of zero, or any value that is not valid, results in no tone played. If the value of {{param|.Capabilities.ToneGeneration}} is {{false}}, no tone is played regardless of the value of this parameter. Voice profile parameters that are specific to MGCP call signaling. Creation of this object occurs on specification of {{pattern|MGCP|.VoiceProfile.{i}.SignalingProtocol}}. Host name or IP address of the main MGCP call agent. Destination port to be used in connecting with the main MGCP call agent. Host name or IP address of the backup MGCP call agent. Destination port to be used in connecting with the backup MGCP call agent. Message retransfer interval, in {{units}}. Max number of message retransfers. Register mode. Port listening for incoming call control signaling. CPE domain string. If {{empty}}, the CPE SHOULD use its IP address. User string used in accessing the call agent. Diffserv code point to be used for outgoing MGCP signaling packets. VLAN ID (as defined in {{bibref|802.1Q-2005}}) to be used for outgoing MGCP signaling packets for this profile. A value of -1 indicates the default value is to be used. If either {{param}} or {{param|EthernetPriorityMark}} are greater than zero, then the outgoing frames MUST be tagged. Otherwise, the outgoing frames MAY be tagged or untagged. Ethernet priority code (as defined in {{bibref|802.1D-2004}}) to be used for outgoing MGCP signaling packets for this profile. A value of -1 indicates the default value is to be used. If either {{param|VLANIDMark}} or {{param}} are greater than zero, then the outgoing frames MUST be tagged. Otherwise, the outgoing frames MAY be tagged or untagged. Indicates whether or not piggyback events are allowed to the MGCP call agent. Indicates whether or not to send RSIP immediately on restart. Voice profile parameters that are specific to H.323 call signaling. Creation of this object occurs on specification of {{pattern|H\.323|.VoiceProfile.{i}.SignalingProtocol}} as the Signaling Protocol. Host name or IP address of H.323 Gatekeeper. Destination port to be used in connecting to the H.323 Gatekeeper. Gatekeeper ID. Defines the TimeToLive specification in the registration with the Gatekeeper in {{units}}. Enables or disables usage of H.235 security baseline security profile as defined in {{bibref|ITU-H.235.1}} baseline security profile. Password to be used when H.235 is enabled. In ITU-T based H.235 authentication, the sendersID is the ID of the gateway as received from the Gatekeeper. As long as the endpointID is not received from the Gatekeeper, the sendersID will be applied as configured here. The generalID is the GatekeeperID. Diffserv code point to be used for outgoing H.323 signaling packets. VLAN ID (as defined in {{bibref|802.1Q-2005}}) to be used for outgoing H.323 signaling packets for this profile. A value of -1 indicates the default value is to be used. If either {{param}} or {{param|EthernetPriorityMark}} is greater than zero, then the outgoing frames MUST be tagged. Otherwise, the outgoing frames MAY be tagged or untagged. Ethernet priority code (as defined in {{bibref|802.1D-2004}}) to be used for outgoing H.323 signaling packets for this profile. A value of -1 indicates the default value is to be used. If either {{param|VLANIDMark}} or {{param}} is greater than zero, then the outgoing frames MUST be tagged. Otherwise, the outgoing frames MAY be tagged or untagged. Voice profile parameters related to the voice stream sent via RTP. Applicable only for a VoIP endpoint. Base of port range to be used for incoming RTP streams for this profile. Top of port range to be used for incoming RTP streams for this profile. Diffserv code point to be used for outgoing RTP packets for this profile. It is RECOMMENDED that by default the DSCP for RTP traffic be set to the value to indicate EF traffic. VLAN ID (as defined in {{bibref|802.1Q-2005}}) to be used for outgoing RTP packets for this profile. A value of -1 indicates the default value is to be used. If either {{param}} or {{param|EthernetPriorityMark}} is greater than zero, then the outgoing frames MUST be tagged. Otherwise, the outgoing frames MAY be tagged or untagged. Ethernet priority code (as defined in {{bibref|802.1D-2004}}) to be used for outgoing RTP packets for this profile. A value of -1 indicates the default value is to be used. If either {{param|VLANIDMark}} or {{param}} is greater than zero, then the outgoing frames MUST be tagged. Otherwise, the outgoing frames MAY be tagged or untagged. Payload type to be used for RTP telephone events. This parameter indicates the payload type to be used for DTMF events if transmission of DTMF information is in use according to {{bibref|RFC2833}}. Voice profile parameters related to RTCP. If supported, the value of {{param|.Capabilities.RTCP}} MUST be {{true}}. Applicable only for a VoIP endpoint. Enable or disable RTCP. Transmission repeat interval, in {{units}}. Local Cname (canonical name). Voice profile parameters for secure voice transmission via SRTP. If supported, the value of {{param|.Capabilities.SRTP}} MUST be {{true}}. Applicable only for a VoIP endpoint. Enable or disable the use of SRTP. If RTCP is enabled, a true value of this parameter also implies the use of SRTCP. {{list}} Each item is a keying method that may be used. By default this parameter MUST have the value of {{param|.Capabilities.SRTPKeyingMethods}}- {{list}} Each item is an encryption key size that may be used. By default this parameter MUST have the value of {{param|.Capabilities.SRTPEncryptionKeySizes}}. Voice profile parameters for RTP payload redundancy as defined by {{bibref|RFC2198}}. If supported, the value of {{param|.Capabilities.RTPRedundancy}} MUST be {{true}}. Applicable only for a VoIP endpoint. Enable or disable the use of RTP payload redundancy as defined by {{bibref|RFC2198}}. The Payload Type of RTP packet as defined in {{bibref|RFC2198}}. Values SHOULD be within the range of dynamic Payload Types (96-127). Block Payload Type of redundancy packet. Specifies the redundancy number for fax and modem pass-through data transmissions. A non-negative value indicates that {{bibref|RFC2198}} is to be used for fax and modem pass-through data. The value indicates the number of redundant copies to be transmitted (the total number transmitted is one plus this value). A value of -1 indicates {{bibref|RFC2198}} is not to be used for fax and modem pass-through data. If {{param|ModemRedundancy}} is present, then {{param}} applies only to fax transmissions, but not to modem transmissions. Specifies the redundancy number for modem pass-through data transmissions. A non-negative value indicates that {{bibref|RFC2198}} is to be used for modem pass-through data. The value indicates the number of redundant copies to be transmitted (the total number transmitted is one plus this value). A value of -1 indicates {{bibref|RFC2198}} is not to be used for modem pass-through data. Specifies the redundancy number for DTMF transmissions. A non-negative value indicates that {{bibref|RFC2198}} is to be used for DTMF. The value indicates the number of redundant copies to be transmitted (the total number transmitted is one plus this value). A value of -1 indicates {{bibref|RFC2198}} is not to be used for DTMF. Specifies the redundancy number for general voice transmissions. A non-negative value indicates that {{bibref|RFC2198}} is to be used for voice. The value indicates the number of redundant copies to be transmitted (the total number transmitted is one plus this value). A value of -1 indicates {{bibref|RFC2198}} is not to be used for voice. The maximum number of sessions using {{bibref|RFC2198}} payload redundancy simultaneously in this VoiceProfile. A value of zero indicates no explicit limit on the number of sessions using redundancy. This object contains information related the numbering plan. This object is applicable only if the device supports a dialing mechanism for which a number plan is needed (for example, a device with an explicit Dial button may not need to be aware of the dialing plan) and if the device does not already support a numbering plan mechanism for this profile (e.g., in-band via MGCP). If supported, the value of {{param|.Capabilities.NumberingPlan}} MUST be {{true}}. Applicable only for a VoIP endpoint. This is the minimum number of digits that must be collected before an outgoing request (e.g., a SIP INVITE) can be initiated. If "End of Dialing" (refer to the definition of the InterDigitTimer) occurs before the minimum number of digits has been reached then the number will be considered incomplete and no request will be initiated. In practice, searching {{object|.VoiceProfile.{i}.NumberingPlan.PrefixInfo.{i}.}} should only commence once the minimum number of digits (as specified by this parameter) has been received. This is the maximum number of digits that may be collected before an outgoing request (e.g., a SIP INVITE) must be initiated. Any additional dialed digits will be ignored. This parameter is only used in the case that no match in {{object|.VoiceProfile.{i}.NumberingPlan.PrefixInfo.{i}.}} has been found. This timer is the maximum allowable time (expressed in {{units}}) between the dialing of digits. This timer is restarted every time a digit is dialed. Expiration of this timer indicates "End of Dialing". This timer is the maximum allowable time (expressed in {{units}}) between the dialing of digits once the minimum number of digits defined on a prefix based has been reached. This timer is only applicable to "open numbering", where the exact number of digits for a prefix is not known. The tone that should be provided to the user when the number dialed is determined to be invalid. The value corresponds to an instance of an EntryID in {{object|.VoiceProfile.{i}.Tone.Description.}}. A value of zero, or any value that does not match a valid EntryID, results in no tone played. If {{param|.Capabilities.ToneGeneration}} is equal to false, no tone is played regardless of the value of this parameter. This is the maximum number of instances of {{object|.VoiceProfile.{i}.NumberingPlan.PrefixInfo.{i}.}} that can be supported. Indicates the number of instances of {{object|.VoiceProfile.{i}.NumberingPlan.PrefixInfo.{i}.}}. Each entry in this table contains information related to an individual prefix in the numbering plan. It is anticipated that once the minimum number of digits has been received, the VoIP device will search this prefix list every time a new digit is received. If no new entry is found, then the object that was previously found will be used instead. If supported, {{param|.VoiceProfile.{i}.NumberingPlan.PrefixInfoMaxEntries}} and {{param|.VoiceProfile.{i}.NumberingPlan.PrefixInfoNumberOfEntries}} MUST be present. The defaults given for this object apply only to explicit creation of an instance of this object and not to automatic creation of instances of this object due to creation of a parent object. This is a string representation of a range of prefixes. Each prefix consists of a "From" part consisting of 1 to n digits (string representation) followed by an optional "To" part consisting of exactly one digit prefixed by a "-" symbol. It should be noted that only the characters "0-9", "*": and "#" can be used to represent the "From" and "To" parts of the prefix range. A further constraint is that the "To" digit MUST always be numerically greater than the last digit of the "From" part. Examples: : 02 : 031-5 : 032 : 0325 : *#34 : #22 This is the minimum number of allowable digits for the prefix range. Once the minimum number of digits has been reached, {{param|.VoiceProfile.{i}.NumberingPlan.InterDigitTimerOpen}} will be used instead of {{param|.VoiceProfile.{i}.NumberingPlan.InterDigitTimerStd}}. If the minimum number of digits has been reached and the inter-digit timer expires, an outgoing request should be initiated. This is the maximum number of allowable digits for the prefix range. Once the number of digits received reaches this value an outgoing request should be initiated. If this parameter has a non-zero value, the specified number of digits will be removed from the internal digit buffer (which contains the dialed digits) from the position specified by {{param|PosOfDigitsToRemove}}. Subsequently a search of {{object}} for a matching prefix using the modified number should be performed. Note that this parameter does not have any impact on the number sent in the outgoing request - but is instead only used for searching within the Numbering Plan. This parameter has no effect if it is set to 0. This parameter is provided to handle Carrier override and other codes that may prefix standard numbers and to ensure that the correct "End of Dialing" can be specified without significant data duplication. This parameter is used in conjunction with {{param|NumberOfDigitsToRemove}}. It specifies the position within the internal digit buffer from which the digits are to be removed. The tone to be played by the VoIP device when the user has dialed exactly the same digits as defined in the prefix. The VoIP device will cease playing the tone once an additional digit has been dialed. The value corresponds to an instance of and EntryID in {{object|.VoiceProfile.{i}.Tone.Description.}}. A value of zero, or any value that does not match a valid EntryID, results in no tone played. If {{param|.Capabilities.ToneGeneration}} is equal to false, no tone is played regardless of the value of this parameter. This is a string representing a Facility Action implemented by the VoIP device. {{bibref|TR-104|Appendix A}} lists a set of defined values for this string. The parameter MAY instead indicate a vendor-specific FacilityAction, which MUST use the format defined in {{bibref|TR-106}}. An empty or unrecognized string (i.e., a Facility Action not supported by the CPE) should be treated as a normal outgoing request. Optional argument associated with {{param|FacilityAction}}. The interpretation of the argument is dependent on a specific value of {{param|FacilityAction}}. Where used, the value is specified in {{bibref|TR-104|Appendix A}} in the definition of the corresponding {{param|FacilityAction}} value. This object defines the contents of the tones and announcements generated locally by the VoIP device. If this object is supported, {{param|.Capabilities.ToneGeneration}} MUST be {{true}}. Applicable only for a VoIP endpoint. Indicates the number of entries in {{object|.VoiceProfile.{i}.Tone.Event.{i}.}}. Indicates the number of entries in {{object|.VoiceProfile.{i}.Tone.Description.{i}.}}. Indicates the number of entries in {{object|.VoiceProfile.{i}.Tone.Pattern.{i}.}}. Table of events for which a tone is defined. The table is pre-populated with the list of events for which the CPE supports definition of tones. If this table is supported, {{param|.VoiceProfile.{i}.Tone.EventNumberOfEntries}} MUST be present. The event for which the tone is to apply. {{enum}} The parameter MAY instead indicate a vendor-specific event name, which MUST use the format defined in {{bibref|TR-106}}. The EntryID of the entry in {{object|.VoiceProfile.{i}.Tone.Description.{i}.}} for the tone to be associated with the given event. A value of zero indicates no tone is to be played for this event. Each entry in this table defines the contents of an individual tone. If ability to add, delete, and modify entries in this table is supported, {{param|.Capabilities.ToneDescriptionsEditable}} MUST be {{true}}. If this table is supported, {{param|.VoiceProfile.{i}.Tone.DescriptionNumberOfEntries}} MUST be present. The defaults given for this object apply only to explicit creation of an instance of this object and not to automatic creation of instances of this object due to creation of a parent object. Unique identifier of this tone. Assigned by the CPE upon creation of the entry. Enables or disables the tone entry. If a disabled tone entry is referenced, the result is that no tone is played. Name of the tone. This parameter is required to be editable only {{param|.Capabilities.ToneDescriptionsEditable}} is equal to {{true}}. EntryID of the entry in {{object|.VoiceProfile.{i}.Tone.Pattern.}} that begins the tone pattern for this tone. If the tone is specified by a tone file instead of a tone pattern, this parameter MUST be set to zero. This parameter is applicable only if {{param|.Capabilities.PatternBasedToneGeneration}} is equal to {{true}}. This parameter is required to be editable only if {{param|.Capabilities.ToneDescriptionsEditable}} is equal to {{true}}. This is the file name of a tone file that has been downloaded to the CPE. The download may have occurred via the TR-069 Download mechanism or by some other means. If the tone is specified by a tone pattern instead of a tone file, this parameter MUST be {{empty}}. This parameter is applicable only if {{param|.Capabilities.FileBasedToneGeneration}} is equal to {{true}}. This parameter is required to be editable only if {{param|.Capabilities.ToneDescriptionsEditable}} is equal to {{true}}. The default number of times the data in {{param|ToneFile}} should be repeated. If the value 0 (zero) is specified then {{param|ToneFile}} should be played indefinitely. This parameter is required to be editable only if {{param|.Capabilities.ToneDescriptionsEditable}} is equal to {{true}}. The text to be displayed by on the screen of the VoIP device when the tone is played and no specific error message has been provided. This parameter is applicable only for VoIP devices capable text display. Each entry in the table defines a single phase in an overall tone pattern. Each phase identifies the entry that corresponds to the next phase. Each entry in the table refers to the entry that corresponds to the next phase of the pattern. The table MAY be set up such that entries form loops, or MAY end after a finite sequence. If this object is supported, {{param|.Capabilities.PatternBasedToneGeneration}} MUST be equal to {{true}}, and {{param|.VoiceProfile.{i}.Tone.PatternNumberOfEntries}} MUST be present. If ability to add, delete, and modify entries in this table is supported, {{param|.Capabilities.ToneDescriptionsEditable}} MUST be equal to {{true}}. The defaults given for this object apply only to explicit creation of an instance of this object and not to automatic creation of instances of this object due to creation of a parent object. Identifier of a tone-pattern entry. This parameter is required to be editable only if {{param|.Capabilities.ToneDescriptionsEditable}} is equal to {{true}}. Note: when {{param|.Capabilities.ToneDescriptionsEditable}} is {{true}}, this parameter is editable so that the {{param|NextEntryID}} values for each table entry can be pre-assigned for a series of associated table entries rather than requiring the ACS to set the value according to an ID assigned dynamically upon creation of each entry. Whether or not a tone is on during this phase of the pattern. If the value is {{false}}, the frequency and power parameters in this entry MUST be ignored. This parameter is required to be editable only if {{param|.Capabilities.ToneDescriptionsEditable}} is equal to {{true}}. First tone frequency in {{units}}. A value of zero indicates this tone component is not used. This parameter is required to be editable only if {{param|.Capabilities.ToneDescriptionsEditable}} is equal to {{true}}. First tone power level in units of 0.1 {{units}}. This parameter is required to be editable only if {{param|.Capabilities.ToneDescriptionsEditable}} is equal to {{true}}. Second tone frequency in {{units}}. A value of zero indicates this tone component is not used. This parameter is required to be editable only if {{param|.Capabilities.ToneDescriptionsEditable}} is equal to {{true}}. Second tone power level in units of 0.1 {{units}}. This parameter is required to be editable only if {{param|.Capabilities.ToneDescriptionsEditable}} is equal to {{true}}. Third tone frequency in {{units}}. A value of zero indicates this tone component is not used. This parameter is required to be editable only if the parameter is supported for reading and {{param|.Capabilities.ToneDescriptionsEditable}} is equal to {{true}}. Third tone power level in units of 0.1 {{units}}. This parameter is required to be editable only if the parameter is supported for reading and {{param|.Capabilities.ToneDescriptionsEditable}} is equal to {{true}}. Fourth tone frequency in {{units}}. A value of zero indicates this tone component is not used. This parameter is required to be editable only if the parameter is supported for reading and {{param|.Capabilities.ToneDescriptionsEditable}} is equal to {{true}}. Fourth tone power level in units of 0.1 {{units}}. This parameter is required to be editable only if the parameter is supported for reading and {{param|.Capabilities.ToneDescriptionsEditable}} is equal to {{true}}. Modulation frequency in {{units}}. A value of zero indicates this tone component is not used. This parameter is required to be editable only if {{param|.Capabilities.ToneDescriptionsEditable}} is equal to {{true}}. Modulation power level in units of 0.1 {{units}}. This parameter is required to be editable only if {{param|.Capabilities.ToneDescriptionsEditable}} is equal to {{true}}. The duration of this phase of the tone pattern, in {{units}}. A value of zero indicates an unlimited duration. This parameter is required to be editable only if {{param|.Capabilities.ToneDescriptionsEditable}} is equal to {{true}}. The {{param|EntryID}} for the next phase of the tone pattern, after the specified {{param|Duration}} of this phase has completed. A value of zero indicates that the tone pattern is to terminate after the current phase is completed. This parameter is required to be editable only if {{param|.Capabilities.ToneDescriptionsEditable}} is equal to {{true}}. This object is provided to permit the purpose of the CPE buttons and function keys to be defined via the ACS. Support of this object is appropriate only for a device that has programmable buttons in its user interface. If this object is supported, {{param|.Capabilities.ButtonMap}} MUST be equal to {{true}}. Applicable only for a VoIP endpoint. Indicates the number of Button objects. Each entry in this table specifies the purpose of each programmable CPE button / function key and whether the user has permission to reprogram the button. Name of the Button. This is an optional parameter that should only be specified for buttons related to a particular Facility Action (e.g., invocation of "Call Return") implemented by the VoIP device. {{bibref|TR-104|Appendix A}} lists a set of defined values for this string. The parameter MAY instead indicate a vendor-specific FacilityAction, which MUST use the format defined in {{bibref|TR-106}}. An empty or unrecognized string (i.e. a Facility Action not supported by the CPE) should be treated as no Facility Action to be taken. Note that If this parameter is specified (non-empty) then {{param|QuickDialNumber}} SHOULD be {{empty}}. Optional argument associated with the specified {{param|FacilityAction}}. The interpretation of the argument is dependent on the specific FacilityAction. Where used, the value is specified in {{bibref|TR-104|Appendix A}} in the definition of the corresponding {{param|FacilityAction}} value. This is a string representing a quick dial destination number. Only the characters '0-9', '*' and '#' can be used. Note that If this parameter is specified (non-empty) then {{param|FacilityAction}} should be {{empty}}. This string represents the message to be displayed on the screen when the button or function key is pressed. This parameter indicates whether the user has permission to program the button or function key. If this parameter is set to {{true}} then {{param|FacilityAction}}, {{param|QuickDialNumber}} and {{param|ButtonMessage}} MUST all be {{empty}}. T.38 Fax information for devices that support T.38 relay. If this object is supported, {{param|.Capabilities.FaxT38}} MUST be equal to {{true}}. Applicable only to a VoIP endpoint. Enable or disable the use of T.38. Maximum data rate for fax. The rate at which high speed data will be sent across the network, in {{units}}. Specifies the packet-level redundancy for high-speed data transmissions (i.e., T.4 image data). Specifies the packet-level redundancy for low-speed data transmissions (i.e., T.30 handshaking information). The method with which data is handled over the network. Object associated with a distinct voice line. Support for adding and removing lines is conditional on whether the CPE supports more than one line in total as indicated by {{param|.Capabilities.MaxLineCount}}. By default, on creation of a given {{object|.VoiceProfile.{i}.}}, a single Line object MUST be present, initially with value {{enum|Disabled|Enable}}. Applicable only for a VoIP endpoint. Enables or disables this line, or places it into a quiescent state. {{enum}} In the {{enum|Quiescent}} state, in-progress sessions remain intact, but no new sessions are allowed. If this parameter is set to {{enum|Quiescent}} in a CPE that does not support the {{enum|Quiescent}} state, it MUST treat it the same as the {{enum|Disabled}} state (and indicate {{enum|Disabled|Status}} in {{param|Status}}). Directory number associated with this line. May be used to identify the line to the user. In case of H.323 signaling, this MUST be an E.164 number. Indicates the status of this line. Indicates the call state for this line. {{list}} Each item corresponds to the value of a particular instance of {{param|.PhyInterface.{i}.InterfaceID}}. Whether or not ringing has been locally muted. Applicable only if the line is associated with a single telephony device for which ringing can be muted. Percent value of current ringer volume level. Applicable only if the line is associated with a single telephony device for which the ringer volume can be controlled. Voice line parameters that are specific to SIP call signaling. Username used to authenticate the connection to the server. Password used to authenticate the connection to the server. URI by which the user agent will identify itself for this line. If empty, the actual URI used in the SIP signaling SHOULD be automatically formed by the CPE as: "sip:UserName@Domain" Where UserName is username given for this line in {{param|AuthUserName}}, and Domain is the domain given for this profile in {{param|.VoiceProfile.{i}.SIP.UserAgentDomain}}. If {{param|.VoiceProfile.{i}.SIP.UserAgentDomain}} is {{empty}}, then the IP address of the CPE SHOULD be used for the domain. If URI is non-empty, but is a SIP or SIPS URI that contains no "@" character, then the actual URI used in the SIP signaling SHOULD be automatically formed by the CPE by appending this parameter with an "@" character followed by the value of {{param|.VoiceProfile.{i}.SIP.UserAgentDomain}}. If {{param|.VoiceProfile.{i}.SIP.UserAgentDomain}} is {{empty}}, then the IP address of the CPE SHOULD be used for the domain. Indicates the number of EventSubscribe objects. Table of SIP Events automatically populated by the CPE with each of the SIP event subscriptions in {{object|.VoiceProfile.{i}.SIP.EventSubscribe.{i}.}}. This table allows specification of the authentication credentials needed for each event subscription. If this table is supported, {{param|.Capabilities.SIP.EventSubscription}} MUST be equal to {{true}} and {{param|.VoiceProfile.{i}.Line.{i}.SIP.SIPEventSubscribeNumberOfElements}} MUST be present. SIP event name corresponding to the value given in {{object|.VoiceProfile.{i}.SIP.EventSubscribe.{i}.}}. Username used to authenticate the connection to the event notify server. Password used to authenticate the connection to the event notify server. Voice line parameters that are specific to MGCP call signaling. Used to identify the line when using MGCP signaling. If empty, the CPE SHOULD use the default names "aaln/1", etc. Voice line parameters that are specific to H.323 call signaling. The H.323 ID assigned to the line. This object defines the ring sequences generated by the VoIP device. If this object is supported, {{param|.Capabilities.RingGeneration}} MUST be equal to {{true}}. Number of entries in {{object|.VoiceProfile.{i}.Line.{i}.Ringer.Event.{i}.}}. Number of entries in {{object|.VoiceProfile.{i}.Line.{i}.Ringer.Description.{i}.}}. Number of entries in {{object|.VoiceProfile.{i}.Line.{i}.Ringer.Pattern.{i}.}}. Table of events for which a ring pattern is defined. The table is pre-populated with the complete list of events for which the CPE supports definition of ring patterns. If this table is supported, {{param|.VoiceProfile.{i}.Line.{i}.Ringer.EventNumberOfEntries}} MUST be present. The event for which the ring pattern is to apply. {{enum}} The parameter MAY instead indicate a vendor-specific event name, which MUST use the format defined in {{bibref|TR-106}}. The value of an instance of {{param|.VoiceProfile.{i}.Line.{i}.Ringer.Description.{i}.EntryID}} for the ring to be associated with the given event. A value of zero indicates ringing is to be disabled for this event. Each entry in this table defines the contents of an individual ring specification. If ability to add, delete, and modify entries in this table is supported, {{param|.Capabilities.RingDescriptionsEditable}} MUST be equal to {{true}}. If this table is supported, the parameter DescriptionNumberOfEntries in the parent object MUST be present. The defaults given for this object apply only to explicit creation of an instance of this object and not to automatic creation of instances of this object due to creation of a parent object. Unique identifier of this ring description. Assigned by the CPE upon creation of the entry. Enables or disables the ring description entry. If a disabled ring description entry is referenced, the result is that no ring is played. Name of the ring description. This parameter is required to be editable only if {{param|.Capabilities.RingDescriptionsEditable}} is equal to {{true}}. The instance of {{param|.VoiceProfile.{i}.Line.{i}.Ringer.Pattern.{i}.EntryID}} that begins the ring pattern for this ring description. If the ring is specified by a ring file instead of a ring pattern, this parameter MUST be set to zero. This parameter is applicable only if {{param|.Capabilities.PatternBasedRingGeneration}} is equal to {{true}}. This parameter is required to be editable only if {{param|.Capabilities.RingDescriptionsEditable}} is equal to {{true}}. This is the file name of a ring file that has been downloaded to the CPE. The download may have occurred via the TR-069 Download mechanism or by some other means. If the ring is specified by a ring pattern instead of a ring file, this parameter MUST be empty. This parameter is applicable only if {{param|.Capabilities.FileBasedRingGeneration}} is equal to {{true}}. This parameter is required to be editable only {{param|.Capabilities.RingDescriptionsEditable}} is equal to {{true}}. Each entry in the table defines a single phase in an overall ring pattern. Each phase identifies the entry that corresponds to the next phase. Each entry in the table refers to the entry that corresponds to the next phase of the pattern. The table MAY be set up such that entries form loops, or MAY end after a finite sequence. If this object is supported, {{param|.Capabilities.PatternBasedRingGeneration}} MUST be equal to {{true}} and {{param|.VoiceProfile.{i}.Line.{i}.Ringer.PatternNumberOfEntries}} MUST be present. If ability to add, delete, and modify entries in this table is supported, {{param|.Capabilities.RingPatternEditable}} MUST be equal to {{true}}. The defaults given for this object apply only to explicit creation of an instance of this object and not to automatic creation of instances of this object due to creation of a parent object. Identifier of a ring-pattern entry. This parameter is required to be editable only if {{param|.Capabilities.RingPatternEditable}} is equal to {{true}}. Note: when {{param|.Capabilities.RingPatternEditable}} is {{true}}, this parameter is editable so that the {{param|NextEntryID}} values for each table entry can be pre-assigned for a series of associated table entries rather than requiring the ACS to set the value according to an ID assigned dynamically upon creation of each entry. If {{true}}, indicates the ringer is to be on for the specified period. {{false}} indicates the ringer is to be off for the specified period. This parameter is required to be editable only if {{param|.Capabilities.RingPatternEditable}} is equal to {{true}}. The duration of this phase of the ring pattern, in {{units}}. A value of zero indicates an unlimited duration. This parameter is required to be editable only if {{param|.Capabilities.RingPatternEditable}} is equal to {{true}}. The value of {{param|EntryID}} for the next phase of the ring pattern, after the value specified by {{param|Duration}} of this phase has completed. A value of zero indicates that the ring pattern is to terminate after the current phase is completed. This parameter is required to be editable only if {{param|.Capabilities.RingPatternEditable}} is equal to {{true}}. Voice line parameters related to optional endpoint based calling features. Enable or disable the transmission of caller ID information on outgoing calls. Enable or disable the transmission of caller ID name information on outgoing calls. String used to identify the caller. Enable or disable call waiting in the endpoint. This parameter should not be present if the CPE does not support endpoint managed call waiting. Status of endpoint managed call waiting, if supported. {{enum}} This parameter should not be present if the CPE does not support endpoint managed call waiting. Indicates the maximum number of simultaneous sessions that may be conferenced together by the endpoint. This value SHOULD be less than the value of {{param|.Capabilities.MaxSessionsPerLine}}. This parameter should not be present if the CPE does not support endpoint managed conference calling. Status of endpoint managed conference calling, if supported. {{enum}} This parameter should not be present if the CPE does not support endpoint managed conference calling. Number of active sessions on this line. This parameter should not be present if the CPE does not support endpoint managed conference calling. No Call in Progress Single call in progress Conference call in progress Enable or disable call forwarding by the endpoint. This parameter should not be present if the CPE does not support endpoint based call forwarding. Directory number to which all incoming calls to this line should be forwarded if {{param|CallForwardUnconditionalEnable}} is {{true}}. This parameter should not be present if the CPE does not support endpoint based call forwarding Enable or disable call forwarding-on-busy by the endpoint. This parameter should not be present if the CPE does not support endpoint based call forwarding. Directory number to which all incoming calls to this line should be forwarded if {{param|CallForwardOnBusyEnable}} is {{true}} and the line is busy. This parameter should not be present if the CPE does not support endpoint based call forwarding Enable or disable call forwarding-on-no-answer by the endpoint. This parameter should not be present if the CPE does not support endpoint based call forwarding. Directory number to which all incoming calls to this line should be forwarded if {{param|CallForwardOnNoAnswerEnable}} is {{true}} and there is no local answer. This parameter should not be present if the CPE does not support endpoint based call forwarding Number of rings before considering there to be no answer for call forwarding-on-no-answer. This parameter should not be present if the CPE does not support endpoint based call forwarding Enable or disable call transfer by the endpoint. This parameter should not be present if the CPE does not support endpoint based call transfer. Enable or disable Message Waiting Indication by the endpoint. This parameter should not be present if the CPE does not support MWI. Indicates whether or not a message is currently waiting on this line as known by the CPE. This parameter should not be present if the CPE does not support MWI. Enable or disable Anonymous Call Block capability in the endpoint. This parameter should not be present if the CPE does not support endpoint based Anonymous Call Block capability. Enable or disable Anonymous Call capability in the endpoint. This parameter should not be present if the CPE does not support endpoint based Anonymous Call capability. Enable or disable Do Not Disturb capability in the endpoint. This parameter should not be present if the CPE does not support endpoint based Do Not Disturb capability. Enable or disable Call Return capability in the endpoint. This parameter should not be present if the CPE does not support endpoint based Call Return capability. Enable or disable Repeat Dial capability in the endpoint. This parameter should not be present if the CPE does not support endpoint based Repeat Dial capability. Voice line parameters related to voice processing capabilities. Gain in {{units}} to apply to the transmitted voice signal prior to encoding. This gain is a modifier of the default transmit-gain, which is unspecified. Gain in {{units}} to apply to the received voice signal after decoding. This gain is a modifier of the default receive-gain, which is unspecified. Enable or disable echo cancellation for this line. Indication of whether or not echo cancellation is currently in use for this line. Tail length in {{units}} of the echo canceller associated with this line (whether or not it is currently in use). This object indicates the state of the transmit and receive codec for this voice line instance. The codec currently in use for the outgoing voice stream. The codec currently in use for the incoming voice stream. Codec bit rate in {{units}} for the codec currently in use for the outgoing voice stream. Codec bit rate in {{units}} for the codec currently in use for the incoming voice stream. Whether or not silence suppression is in use for the outgoing voice stream. Whether or not silence suppression is in use for the incoming voice stream. Current outgoing packetization period in {{units}}. Table to describe the set of codecs enabled for use with this line. Each entry in this table refers to a distinct combination of codec and bit rate. When an instance of {{object|.VoiceProfile.{i}.Line.{i}.}} is created, this object MUST be populated with the set of supported codecs matching {{object|.Capabilities.Codecs.}}. The ACS MAY restrict and/or prioritize the codec support for this profile using this object. Applicable only for a VoIP endpoint. Unique identifier for each entry in this table. The value MUST match that of the corresponding entry in {{object|.Capabilities.Codecs.}} table. Identifier of the codec type. The value MUST match that of the corresponding entry in {{object|.Capabilities.Codecs.}}. Bit rate, in {{units}}. The value MUST match that of the corresponding entry in {{object|.Capabilities.Codecs.}} {{list}} Each item is a supported packetization period, in milliseconds, or continuous ranges of packetization periods as defined in {{param|.Capabilities.Codecs.{i}.PacketizationPeriod}}. The set of packetization periods may be restricted by modifying the value of this parameter to a more restricted set of values than is listed in {{param|.Capabilities.Codecs.{i}.PacketizationPeriod}}. The CPE MUST ignore any values or portions of ranges outside of those specified in {{param|.Capabilities.Codecs.{i}.PacketizationPeriod}}. Indicates support for silence suppression for this codec. If silence suppression is supported, it can be disabled for this codec/bit-rate by setting this parameter to {{false}}. Enable or disable the use of this combination of codec parameters. Indicates the priority for this combination of codec parameters, where 1 is the highest priority. Where the priority differs between entries in this table, the CPE SHOULD use the highest priority (lowest numbered) entry among those supported by the remote endpoint and consistent with the available bandwidth. Where the priorities are equal among multiple entries, the CPE MAY apply a local criterion for choosing among them. Information on each active session associated with this voice line instance. The time that the session started, in UTC. Duration time of the current session, in seconds. The IP address of far end VoIP device. The UDP port used for current RTP session in the far end device. The local UDP port used for current RTP session. Statistics for this voice line instance. When set to one, resets the statistics for this voice line. Always False when read. Total number of RTP packets sent for this line. Total number of RTP packets received for this line. Total number of RTP payload bytes sent for this line. Total number of RTP payload bytes received for this line. Total number of RTP packets that have been lost for this line. Total number of times the receive jitter buffer has overrun for this line. Total number of times the receive jitter buffer has underrun for this line. Total incoming calls received. Total incoming calls answered by the local user. Total incoming calls that successfully completed call setup signaling. Total incoming calls that failed to successfully complete call setup signaling. Total outgoing calls attempted. Total outgoing calls answered by the called party. Total outgoing calls that successfully completed call setup signaling. Total outgoing calls that failed to successfully complete call setup signaling. Total calls that were successfully connected (incoming or outgoing), but dropped unexpectedly while in progress without explicit user termination. Cumulative call duration in seconds. The number of seconds the CPE is unable to maintain a connection to the server. SHOULD not include time in which overall network connectivity is unavailable. Applies only to SIP. Current receive packet loss rate in percent, calculated as defined in {{bibref|RFC3550|Section6.4}} Current far end receive packet lost rate in percent, calculated as defined in {{bibref|RFC3550|Section6.4}}. Current receive interarrival jitter in {{units}}. Calculated from J(i) as defined in {{bibref|RFC3550|Section6.4}}, with units converted to {{units}}. Current Interarrival jitter in {{units}} as reported from the far-end device via RTCP. Calculated from J(i) as defined in {{bibref|RFC3550|Section64.}}, with units converted to {{units}}. Current round trip delay in {{units}} calculated as defined in {{bibref|RFC3550|Section6.4}}. Average receive interarrival jitter in {{units}} since the beginning of the current call. Calculated as the average of D(i,j) as defined in {{bibref|RFC3550|Section6.4}}, with units converted to {{units}}. Average far-end interarrival jitter in {{units}} since the beginning of the current call. Calculated as the average of the interarrival jitter values reported by the far-end, with units converted to {{units}}. Average round trip delay in {{units}} since the beginning of the current call. Average of the {{param|RoundTripDelay}} statistic accumulated each time the delay is calculated. Each instance is associated with a distinct physical FXS (Foreign eXchange Station) port. Instances of this object are statically created by the CPE. Applicable only for a VoIP Endpoint. The physical port number on the device. The unique identifier of the physical port. This value MAY be used in {{param|.VoiceProfile.{i}.Line.{i}.PhyReferenceList}} to indicate which physical ports are associated with a line. A description of the physical port. Voice port tests. If this object is supported, {{param|.Capabilities.VoicePortTests}} MUST be equal to {{true}}. Indicates the current test state. {{enum}} Value MAY be set to {{enum|Requested}} to initiate a diagnostic test. When writing, the only allowed value is {{enum|Requested}}. To ensure the use of the proper test parameters (the writable parameters in this object), the test parameters MUST be set either prior to or at the same time as (in the same SetParameterValues) setting the value {{enum|Requested}}. When requested, the CPE SHOULD wait until after completion of the communication session with the ACS before starting the test. When the test initiated by the ACS is completed (successfully or not), the CPE MUST establish a new connection to the ACS to allow the ACS to view the results, indicating the Event code "8 DIAGNOSTICS COMPLETE" in the Inform message. Indicates which test to perform. {{enum}} The phone connectivity test indicates that the CPE should determine if one or more phones associated with this physical port are properly connected. This test is appropriate only for CPE that connect to phones of any type. The parameter MAY instead indicate a vendor-specific test, which MUST use the format defined in {{bibref|TR-106}}. For example: "X_EXAMPLE-COM_MyTest" Indicates whether or not at least one phone associated with this physical port is properly connected. This parameter is applicable only if {{enum|PhoneConnectivityTest|TestSelector}} is supported. Support for creation and deletion of Profiles is REQUIRED only if more than one Profile is supported as indicated by {{param|.Capabilities.MaxProfileCount}}. Support for creation and deletion of Lines is REQUIRED only if more than one Line is supported as indicated by {{param|.Capabilities.MaxLineCount}}. This parameter is REQUIRED to be writable only if there is more than one entry in this table. This parameter is REQUIRED to be writable only if there is more than one entry in this table.