SIP Extension VOIP

Last Updated : Jun 20, 2024 |

Navigation: Call Management > Extensions > Edit Extension > SIP VoIP

The IP Office uses these settings for SIP extensions. For example, for J100 Series phones.

Field

Description

IP Address

Default = 0.0.0.0 (Accept any IPv4 address)

If set, the IP Office will only accept registration from a device with the same address.

IP Address (IPv6)

Default = Blank (Accept any IPv6 address)

If set, the IP Office will only accept registration from a device with the same address.

  • Supported on Linux-based IP Office R12.1 systems only. To enable IPv6 support, select LAN1 > LAN Setting (IPv6) > IPv6 and/or LAN2 > LAN Setting (IPv6) > IPv6.

  • The IP Office supports IPv6 addresses in the following formats:

    • Full address: For example, 2001:0000:040F:0000:0000:0000:805B:001B.

    • Replace one series of :0000: parts with ::. For example, 2001:0000:040F::805B:001B.

    • Replace any individual :0000: parts with :0:. For example, 2001:0:040F::805B:001B.

    • Remove leading 0 zeros after any : colon. For example, 2001:0:40F::805B:1B.

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.

Reserve License

Default = None.

By default, the IP Office issues phone licenses in the order that devices register. Using this setting, you can license an extension before the device registers.

The options are:

  • Reserve Avaya IP Endpoint License

  • Reserve 3rd Party IP Endpoint License

  • Both

  • None

Note:

  • Avaya IP phones supported by IP Office require an Avaya IP Endpoint license.

  • Other IP phones require a Third-Party IP Endpoint license.

  • When WebLM licensing is enabled, this field is automatically set to Reserve Avaya IP Endpoint License. The Both and None options are not available.

  • When the Profile of the corresponding user is set to Centralized User, this field is automatically set to Centralized Endpoint License.

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.

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 Transport

Default = RFC2833.

Set how the IP Office signals DTMF key presses to the remote end. The supported options are In Band, RFC2833, or Info.

Requires DTMF

Default = Off.

You can use this setting to attempt direct media between devices with differing DTMF setting. This requires you to enable Ignore DTMF Mismatch for Phones (System Settings > System > VoIP).

  • If disabled, during direct media check the IP Office ignores the DTMF checks if the call is between two VoIP phones.

    • Direct media is still not be possible if other settings differ, for example; codecs, NAT, or security settings.

  • You must enable Requires DTMF if the extension needs to receive DTMF signals.

The IP Office treats SIP soft-phone that do not have an extension record in the IP Office configuration as not requiring DTMF.

Local Hold Music

Default = Off.

When enabled, the extension plays local music when on HOLD.

  • For calls using a SIP line, if the SIP line Local Hold Music setting is enabled (System Settings > Line > Add/Edit Trunk Line > SIP Line > SIP Advanced), you must disable the extension Local Hold Music setting to play far end music to the extension.

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.

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.

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.

3rd Party Auto Answer

Default = None.

This setting applies to 3rd party standard SIP extensions. The options are:

  • RFC 5373: Add an RFC 5373 auto answer header to the INVITE.

  • answer-after: Add answer-after header.

  • device auto answers: IP Office relies on the phone to auto answer calls.

Media Security

Default = Same as System.

These settings control how the extension uses 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.