IP Office Line VoIP Settings

Last Updated : Apr 21, 2016 |

Navigation: System Settings > Line > Add/Edit Trunk Line > IP Office Line > VoIP Settings

Configuration Settings

These settings can be edited Online. Changes to these settings do not require a reboot of the system.

Field

Description

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.

Media Security

Default = Same as System.

Secure RTP (SRTP) can be used between IP Offices to add additional security. These settings 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.

VoIP Silence Suppression

Default = Off

When enabled, when the IP Office detects silence during an IP call, it does not send any audio data.

  • Lines between IP Office systems using G.711 ignore this feature.

  • On trunks between networked IP Office systems, you must enable the setting at both ends.

Out Of Band DTMF

Default = On.

Out of Band DTMF is set to on and cannot be changed.

Allow Direct Media Path

Default = On

This settings controls whether calls between IP endpoints and/or lines must go through the IP Office.

  • If disabled, calls go through the IP Office and use its resources. RTP relay allows calls between devices using the same audio codec to not require a voice compression channel.

  • If enabled, calls can take routes other than through the IP Office system. Both ends must support direct media and have matching VoIP settings. For example, both ends must use the same protocol (SIP or H.323), same addressing (IPv4 or IPv6), and so on. Otherwise, the call goes through the IP Office system.

    • For extensions, disabling Requires DTMF allows the extension to attempt direct media even if the other end has differing DTMF settings.