SIP Account

Registration Status

Displays the SIP account status. The field is automatically populated.

The status can be the following:

SIP User ID

Specifies the SIP user ID used to log in to the phone.

You can also type the SIP user ID, which is a combination of the following values:

The default value is empty.

Authentication User ID

Specifies the authentication ID.

You can also type the authentication user ID in this field if authentication is enabled on the SIP server.

The authentication user ID is a combination of the following values:

Authentication Password

Specifies the authentication password.

You can also type the password in this field if authentication is enabled on the SIP server.

SIP Global Settings

SIP Domain

Specifies the SIP domain used for SIP registration.

The valid value is a string of 0 to 255 ASCII characters.

Enable PPM as source of Proxy Server

Specifies whether PPM is used as a source of SIP proxy server information.

The options are:

UDP Transport

Specifies whether UDP transport is allowed.

The options are:

Proxy Policy

Specifies whether SIP proxy servers are read-only or can be edited.

The options are:

SIP Proxy Server

Specifies a list of SIP controller designators.

You can set the SIP controllers in FQDNs and IP addresses format. If you set both, FQDNs takes precedence, IP addresses are ignored. When SIP proxy selection policy is Automatic, set the SIP controller in FQDN format in SIP_CONTROLLER_LIST only. If you set the FQDN in the SIP_CONTROLLER_LIST_2, the phone ignores the value.

Ensure to set SIP_CONTROLLER_LIST in FQDN format in at least one of the remote sources (LLDP,DHCP,Settings file).

If remote sources have any SIP controller in FQDN format, the phone ignores the SIP controllers in IP address format from remote and PPM sources. The help section displays such IP addresses in the precendence table.

For any SIP controller FQDN, the phone does not support DNS query of NAPTR and SRV records. The DNS server with the FQDN, resolves to A/AAAA records depending on the IP Mode of the device.

In IPv6 deployments if you specify SIP controllers in FQDN format, use SIP_CONTROLLER_LIST and not SIP_CONTROLLER_LIST_2.

FQDN_IP_MAP parameter has no effect, if you use SIP controllers in FQDN format.

SIP Proxy Server (Automatic)

Specifies the SIP proxy server settings as received from the 46xxsettings.txt file or PPM.

Register to Proxy Server

Specifies whether the phone registers simultaneously to a proxy server.

The options are:

Number of proxy server to register simultaneously

Specifies the number of SIP proxy controllers that the phone can register simultaneously.

The options are:

Number of Line Appearances

Specifies the number of line appearances that the phone will display. For each displayed line appearance there is a specific line appearance index.

The options range from 1 to 10 line appearances. The default value is 3.

Authentication User-ID Field

Controls the display of the User ID input field on the phone Login Screen, and Authentication User ID on the Web UI SIP Account tab.

The options are:

Registration Interval

Specifies the time interval in seconds between two registrations to the SIP proxy.

The valid value is an integer from 30 to 86,400. The default value is 900 seconds.

Un-registration Wait Timer (seconds)

Specifies the time in seconds during which the phone waits before terminating all SIP dialog and SIP registrations.

The valid value is an integer from 4 to 3,600. The default value is 32 seconds.

Registration Wait Timer (seconds)

Specifies the time in seconds during which the phone waits for a response message from registration. If no response message is received within this time, the phone tries to register again.

The valid value is an integer from 4 to 3,600. The default value is 32 seconds.

Signaling IP Preference

This parameter is used by SIP signaling only on a dual mode phone (phone with both IPv4 and IPv16 addresses configured) to select the preferred SIP controller IP addresses.

The default value is IPv4.

Media IP Preference

Specifies the preference of SDP media group lines and the SDP answer/offer format when phone is in dual mode.

The default value is IPv4.

Codecs and DTMF

OPUS

Specifies whether the OPUS codec capability of the phone is enabled or disabled.

The options are:

G.722

Specifies whether the G.722 codec is enabled.

The options are:

G.726

Specifies whether the G.726 codec is enabled.

The options are:

G.729

Specifies whether the G.729A codec is enabled.

The options are:

G.711u law

Specifies whether the G.711u law codec is enabled.

The options are:

G.711a law

Specifies whether the G.711a law codec is enabled.

The options are:

Send DTMF

Specifies whether the phone sends DTMF tones in-band as regular audio, or out-of-band using RFC 2833 procedures.

The options are:

OPUS Payload

Dynamically specifies the RTP payload type to be used for OPUS codec. The parameter is used when the media request is sent to the far-end in an INVITE or 200 OK when INVITE with no Session Description Protocol (SDP) is received.

The valid value is an integer from 96 to 127. The default value is 116.

G.726 Payload

Specifies the RTP payload type to be used for the G.726 codec.

The valid value is an integer from 96 to 127. The default value is 110.

DTMF Payload

Specifies the RTP payload type to be used for RFC 2833 signaling.

The valid value is an integer from 96 to 127. The default value is 120.

Codec Priority

Specifies the preferred priority of codecs. To set the parameter see

Assigning Codec Priority

RTP

Play Tone till RTP

Specifies whether the locally generated ringback tone stops when SDP is received for an early media session, or whether it continues until RTP is actually received from the far-end party.

The options are:

Symmetric RTP

Specifies whether the phone must receive RTP if the UDP source port number is not same as the UDP destination port number.

The options are:

SRTP

Media Encryption

Specifies the crypto suite and session parameters for media encryption.

The options are:

Encrypt RTCP

Specifies whether RTCP packets are encrypted or not.

The options are:

Enforce "SIPS" URI for SRTP

Specifies whether a SIPS URI must be used for SRTP.

The options are:

SDP Negotiation Capability

Specifies the Session Description Protocol (SDP) negotiation capability.

Voice Quality Monitoring

RTCP_XR

Specifies whether and how VoIP Metrics Report Block as defined in RTP Control Protocol Extended Reports (RTCP XR) (RFC 3611) is sent.

The options are:

RTCP Monitor Address

Specifies the IP or DNS address of the RTCP monitor.

The valid value is a string of up to 255 ASCII characters. The default value is empty.

RTCP Monitor Port

Specifies the RTCP monitor port number.

Valid value is an integer from 0 to 65535. The default value is 5005.

RTCP Monitoring Report Period

Specifies the time interval in seconds for sending out RTCP monitoring reports.

Valid value is an integer from 5 to 30. The default value is 5 seconds.

RTCP Publish Address

This parameter is not supported in Avaya Aura®environment.

Timers and Count

SIP Timer T1

Specifies an estimate in milliseconds for the Round Trip Time (RTT).

The valid value is an integer from 500 to 10,000.

The default value is 500 milliseconds.

SIP Timer T2

Specifies the maximum retransmit interval in milliseconds for non-INVITE requests and INVITE responses.

The valid value is an integer from 2,000 to 40,000.

The default value is 4,000 milliseconds.

SIP Timer T4

Specifies the maximum duration in milliseconds for which a message remains in the network.

The valid value is an integer from 2,500 to 60,000.

The default value is 5,000 milliseconds.

INVITE Response Timeout

Specifies the maximum number of seconds that the phone waits for another response after receiving a SIP 100 Trying response.

The valid value is an integer from 30 to 180.

The default value is 60 seconds.

Failed Session Removal Timer

Specifies the time in seconds to automatically remove a failed call session.

The valid value is an integer from 5 to 999.

The default value is 30 seconds.

Outbound Subscription Duration Request

Specifies the outbound subscription request duration in seconds.

The valid value is an integer from 60 to 31,536,000.

The default value is 86,400 seconds.

Controller Search Interval

Specifies the time in seconds that the phone waits to complete the maintenance check for monitored controllers.

The valid value is an integer from 4 to 3,600.

The default value is 16 seconds.

Active subscription wait time for "avaya-cm-feature-status"

Specifies the time in seconds that the phone waits to validate an active subscription when it subscribes to the avaya-cm-feature-status package.

The valid value is an integer from 16 to 3,600.

The default value is 32 seconds.

Remote Data Source initial retry time

Specifies the number of seconds that the phone waits for the first time before trying to contact the PPM server again after a failed attempt. Each subsequent retry is delayed by double the previous delay time.

The valid value is an integer from 2 to 3600.

The default value is 2 seconds.

Remote Data Source maximum retry time

Specifies the maximum delay interval in seconds after which the phone stops to contact the PPM server.

The valid value is an integer from 2 to 3,600.

The default value is 600 seconds.

Remote Data Source initial retry attempts

Specifies the number of attempts the PPM adaptor must try to download from PPM before it stops connecting to the PPM server.

The valid value is an integer from 1 to 30.

The default value is 15 attempts.

Local Port

RTP Port (minimum)

Specifies the lower limit of a port range.

The valid value is an integer from 1024 to 65,503.

The default value is 2048.

RTP Port (range)

Specifies the port range to be used by the following connections:

The valid value is an integer from 32 to 63487.

The default value is 40.

SIP Signaling Port (minimum)

Specifies the lower limit of a port range to be used for SIP signaling.

The valid value is an integer from 5062 to 65503.

The default value is 1024.

SIP Signaling Port (range)

Specifies the port range to be used for SIP signaling.

The valid value is an integer from 32 to 64511.

The default value is 64511.

Miscellaneous

Conference Factory URI

Specifies the URI for Avaya Aura® Conferencing or network conferencing in Open SIP environments.

The valid value is a string of up to 255 ASCII characters.

Subscribe Event Packages

Specifies a comma-separated list of event packages to subscribe to after registration.

Allowed values are:

For IP Office, you must use the following:

For the Open SIP environment, you can use message-summary.

Voice Mail Access Code

Specifies the number to access the voice mail in a non-Avaya environment.

100rel

Specifies whether the 100rel option tag is included in the SIP INVITE header field.

The options are:

Validate Incoming messages

Specifies whether AOR received in Request-URI of an incoming call must be validated with the contact header published by phone during registration.

The options are:

‘Privacy’ header in Incoming message

Specifies whether AOR received in Request-URI of an incoming call must be private in the contact header published by the phone during registration.

The options are:

Validate host in SIP URI

Specifies whether to accept SIP URI with unrecognized host part in INVITE message.

The valid options are: