Last Updated : Jun 01, 2016 |

Navigation: Line | SIP Line | VoIP

This form is used to configure the VoIP settings applied to calls on the SIP trunk.

Configuration Settings

These settings are mergeable. 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 | System | VoIP | 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.

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 = Disabled.

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 | System | VoIP | 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 | System | VoIP | 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.

Local Hold Music

Default = Off.

When enabled, if the far end puts the call on HOLD, the system plays music received from far end (SIP Line) to the other end. RTCP reports are sent towards SIP Line. When disabled, the system plays local music to the other endpoint and no RTCP packets are sent to SIP trunk.

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 Lockdown

Default = Off.

After making a SIP offer with a list of codecs, the IP Office expects an answer with a single codec selected from the list. User agents that send an answer with multiple codecs expect to switch to any of those codecs during the call without further negotiation, which the IP Office does not support. Instead, loss of speech occurs if the user agent changes codec without renegotiating.

  • If enabled, when the IP Office receives an answer with multiple codecs, the IP Office sends a re-INVITE and a SIP offer with just one codec.

  • This option requires Re-Invite Supported enabled.

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.

PRACK/100rel Supported

Default = Off.

When selected, supports Provisional Reliable Acknowledgment (PRACK) on SIP trunks. Enable this parameter when you want to ensure that provisional responses, such as announcement messages, have been delivered. Provisional responses provide information on the progress of the request that is in process.  For example, while a cell phone call is being connected, there may be a delay while the cell phone is located; an announcement such as “please wait while we attempt to reach the subscriber” provides provisional information to the caller while the request is in process. PRACK, which is defined in RFC 3262, provides a mechanism to ensure the delivery of these provisional responses.

Force direct media with phones

Default = On

When enabled, if an Avaya IP phone dials digits during a direct media call, the IP Office changes the call to indirect media and sends the digits as RFC2833. 15-seconds after the last digit, the IP Office changes the call back to direct media.

  • This setting is requires the line to have Re-Invite Supported and Allow Direct Media Path enabled, and DTMF Support set to RFC2833/RF4733.

G.711 Fax ECAN

Default = Off

When enabled, if the IP Office detects a fax call, it switches to G.711 with echo cancellation (ECAN) based on the 'G.711 Fax ECAN field, NLP disabled, a fixed jitter buffer, and silence suppression is disabled. You can use this to avoid an ECAN mismatch with the trunk provider.

  • This setting is only available on IP500 V2 systems when Fax Transport Support is set to G.711 or T38 Fallback.