Re-Invite Supported

Default = Off.

When enabled, the IP Office can use Re-Invite during a call to change the characteristics of the call. For example, when the target of an incoming call or a transfer does not support the codec originally negotiated on the trunk.

• Requires the ITSP to also support Re-Invite.

• This setting must be enabled for video support.

Codec Selection

Default = System Default

Set the supported codecs. Where possible, Avaya recommend that you use the same set of codecs for all IP Office systems, lines, and extensions.

The options are:

• System Default - Use the codec list set in the system settings.

• Custom - Configure a list of codec preferences for the line.

  • You can move codecs between the Unused and Selected sets and change the order of the codecs.

  • The codecs available are set by System Settings > System > VoIP. The possible codecs are:

    • OPUS - Supported on Linux-based IP Office systems only.

    • G.711 ALAW/G.711 ULAW

    • G.729

    • G.723.1 - Supported on IP500 V2 systems only.

    • G.722 64K - Supported by Linux-based IP Office systems and on IP500 V2 systems with IP500 VCM, IP500 VCM V2 or IP500 Combo cards.

Fax Transport Support

Default = None.

If enabled, when the IP Office detects fax tone, it will renegotiate the call codec as configured below.

• This option requires Re-Invite Supported.

• IP Office systems in a network support fax relay for fax calls between systems.

• IP500 V2 systems can terminate T38 fax calls.

• Linux-based IP Office systems can route the calls between trunks/terminals with compatible fax types.

The supported options are:

• None - Do not support fax.

• G.711 - Use G.711 to send and receive faxes.

• T38 - Use T38 to send and receive faxes.

• T38 Fallback - Use T38 to send and receive faxes. If the call destination does not support T38, the IP Office will send a re-invite to change the transport method to G.711.

Call Initiation Timeout (s)

Default = 4 seconds. Range = 1 to 99 seconds.

Sets how long the IP Office system should wait for a response to an attempt to initiate a call before following the alternate routes set in an ARS form.

DTMF Support

Default = RFC2833 (IP500 V2), RFC2833/RFC4733 (Linux-Based Server)

Sets how the IP Office signals DTMF key press digits to the remote end. The options are:

• In Band - Send digits as tones within the call audio.

• RFC2833 or RFC2833/RF4733 - Send digits using a separate audio stream from the call audio. If not supported by the far end, the line reverts to using In Band signaling.

• Info - Send the digits in SIP INFO packets.

Media Security

Default = Enforced.

These setting control whether SRTP is used for this line and the settings used for the SRTP. The options are:

• Same as System: Matches the system setting at System Settings > System > VoIP Security.

• Disabled: Use RTP.

• Preferred: Attempt to use SRTP. If SRTP call setup is unsuccessful, fall back to RTP.

• Enforced: Use SRTP. If SRTP call setup is unsuccessful, the call fails.

  • For calls using Dial Emergency, the IP Office will switch to RTP if SRTP call setup fails.

Advanced Media Security Options

Default = Same as System.

Sets the requirements for SRTP when enabled.

• Same as System:

  Use the same settings as configured on System Settings > System > VoIP Security.

• Encryptions: Default = RTP

  Sets which parts of a session SRTP protects using encryption.

• Authentication: Default = RTP and RTCP

  Sets which parts of the session SRTP protects using authentication.

• Replay Protection SRTP Window Size: Default = 64. Not adjustable.

  The IP Office will accept authenticated packets that have a sequence number that is higher than or within 64 packets of the highest-numbered packet already received.

• Crypto Suites: Default = SRTP_AES_CM_128_SHA1_80.

  Set the crypto suites SRTP uses for encryption. The options are SRTP_AES_CM_128_SHA1_32 and SRTP_AES_CM_128_SHA1_80.