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SIP Extension VOIP

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

These settings are shown for SIP IP extensions.

These settings are not mergeable. Changes to these settings will require a reboot of the system.

Field Description
IP Address Default = 0.0.0.0

The IP address of the phone. The default setting accepts connection from any address. If an address is entered, registration is only accepted from a device with that address.

Codec Selection Default = System Default

This field defines the codec or codecs offered during call setup.

The available codecs in default preference order are: G.711 A-Law, G.711 ULAW, G.729 and G.723.1. Note that the default order for G.711 codecs will vary to match the system's default companding setting. G.723.1 is not supported on Linux based systems.

The G.722 64K codec is also supported on IP500/IP500 V2 systems with IP500 VCM, IP500 VCM V2 or IP500 Combo cards.  For Server Edition it is supported on Primary Server,  Secondary Serverand Expansion System (L) systems and on Expansion System (V2) systems fitted with IP500 VCM, IP500 VCM V2 or IP500 Combo.

The codecs available to be used are set through the System Codec list (System | System Codec). The options are:

  • System Default: This is the default setting. When selected, the codec list below show matches the codecs set in the system wide Default Selection list (System | Codecs).

  • Custom: This option allows specific configuration of the codec preferences to be different from the system Default Selection list. When Custom is selected, the list can be used to select which codecs are in the Unused list and in the Selected list and to change the order of the selected codecs.

Fax Transport Support: Default = Off.

This option is only available if Re-Invite Supported is selected. When enabled, the system performs fax tone detection on calls routed via the line and, if fax tone is detected, renegotiates the call codec as configured below. The SIP line provider must support the selected fax method and Re-Invite. The system must have available VCM resources using an IP500 VCM, IP500 VCM V2 or IP500 Combo base card.

For systems in a network, fax relay is supported for fax calls between the systems.

The options are:

  • None Select this option if fax is not supported by the line provider.

  • G.711   G.711 is used for the sending and receiving of faxes.

  • T38   T38 is used for the sending and receiving of faxes. This option is not supported by Linux based systems.

  • T38 Fallback   When you enable this option, T38 is used for sending and receiving faxes on a SIP line. If the called destination does not support T38, the system will send a re-invite to change the transport method to G.711. This option is not supported on Linux based systems.

TDM | IP Gain Default = Default (0dB). Range = -31dB to +31dB.

Allows adjustment of the gain on audio from the system TDM interface to the IP connection. This field is not shown on Linux based platforms.

IP | TDM Gain Default = Default (0dB). Range = -31dB to +31dB. Allows adjustment of the gain on audio from the IP connection to the system TDM interface. This field is not shown on Linux based platforms.
DTMF Support Default = RFC2833.

This setting is used to select the method by which DTMF key presses are signalled to the remote end. The supported options are In Band, RFC2833 or Info.

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 whether SRTP is used for this extension and the settings used for the SRTP. The options are:

  • Same As System: Use the same settings as the system setting configured on the System | VoIP Security tab.

  • Disable: Media security is not required. All media sessions (audio, video, and data) will be enforced to use RTP only.

  • Enforce: Media security is required. All media sessions (audio, video, and data) will be enforced to use SRTP only.

    warningWarning

    Selecting Enforce on a line or extension that does not support media security will result in media setup failures.

  • Best Effort: Media security is preferred. Attempt to use secure media first and if unsuccessful, fall back to non-secure media.

Advanced Media Security Options Not displayed if Media Security is set to Disabled. The options are:
  • Same as System: Use the same setting as the system setting configured on the System | VoIP Security tab.

  • Encryptions: Default = RTP This setting allows selection of which parts of a media session should be protected using encryption. The default is to encrypt just the RTP stream (the speech).

  • Authentication: Default = RTP and RTCP This setting allows selection of which parts of the media session should be protected using authentication.

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

  • Crypto Suites: Default = SRTP_AES_CM_128_SHA1_80. There is also the option to select SRTP_AES_CM_128_SHA1_32.  

VoIP Silence Suppression Default = Off

When selected, this option will detect periods of silence on any call over the line and will not send any data during those silent periods. This feature is not used on IP lines using G.711 between systems. On trunk's between networked systems, the same setting should be set at both ends

Local Hold Music Default = Off.
Allow Direct Media Path Default = On.

This settings controls whether IP calls must be routed via the system or can be routed alternately if possible within the network structure

If enabled, IP calls can take routes other than through the system. This removes the need for a voice compression channel. Both ends of the calls must support Direct Media and be using the same protocol (H.323 or SIP). Enabling this option may cause some vendors problems with changing the media path mid call.

If disabled or not supported at on one end of the call, the call is routed via the system. RTP relay support allows calls between devices using the same audio codec to not require a voice compression channel.

RE-Invite Supported Default = On.

When enabled, Re-Invite can be used during a session to change the characteristics of the session. 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.

Codec Lockdown Default = Off.

Supports RFC 3264 Section 10.2 when RE-Invite Supported is enabled. In response to a SIP offer with a list of codecs supported, some SIP user agents supply a SDP answer that also lists multiple codecs. This means that the user agent may switch to any of those codecs during the session without further negotiation. The system does not support multiple concurrent codecs for a session, so loss of speech path will occur if the codec is changed during the session. If codec lockdown is enabled, when the system receives an SDP answer with more than one codec from the list of offered codecs, it sends an extra re-INVITE using just a single codec from the list and resubmits a new SDP offer with just the single chosen codec.

Reserve License Default = None. Each Avaya IP phones requires an Avaya IP Endpoint license. Each non-Avaya IP phones requires an 3rd Party IP Endpoint license. Normally these licenses are issued in the order that devices register. This option allows this extension to be pre-licensed before the device has registered. This helps prevent a previously licensed phone becoming unlicensed following a system restart if unlicensed devices are also present. The options are:
  • Reserve Avaya IP Endpoint License

  • Reserve 3rd Party IP Endpoint License

  • Both

  • None

Note the following:

  • 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 and cannot be changed.