User's Manual

Avaya Branch Gateway Manager 10.0 Page 253
15-601011 Issue 29r (Friday, November 02, 2012)B5800 Branch Gateway
Configuration Settings: Line
5.4.12.2 VoIP
Usability
· Mergeable: These settings are not mergeable. Changes to these settings will require a reboot of the system.
Configuration Settings
· Gateway IP Address: Default = Blank
Enter the IP address of the gateway device at the remote end. This address must not be shared by any other IP line
(H.323, SIP, SES or IP DECT).
· 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 U-Law, 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 systems with
IP500 VCM, IP500 VCM V2 or IP500 Combo cards but it not used by default.
The codecs available to be used are set through the System Codec list (System | System Codec ). The
drop-down selector 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.
· Call Initiation Timeout: Default = 4 seconds. Range = 1 to 99 seconds.
This option sets how long the system should wait for a response to its attempt to initiate a call before following the
alternate routes set in an ARS form.
· 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.
· Media Security: Default = Disable. Release 6.x+.
Secure RTP (SRTP) can be used between IP devices to add additional security. These setting control whether SRTP is
used for this device and the setting used for the SRTP. For further details of SRTP refer to Secure VoIP (SRTP) .
Note that changing the media security setting of a device may cause it to re-register and may end any call currently in
progress on that device.
· 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.
· Prefer – Media security is preferred. Attempt to use secure media first, and if unsuccessful fall back to
non-secure media. This option is not available for H.323 extensions.
· System Default – Use the system-wide default setting specified in the system Media Security (SIP) field (
System | Telephony | Telephony ).
· Advanced
You are able to select this button when the Media Security field is set to Enforce or Prefer. When selected, the
Advanced Media Security Options are displayed.
· 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 = RTCP
This setting allows selection of which parts of the media session should be protected using authentication. The
default is authenticate just the RTCP stream (call control signals).
· Replay Protection:
Displays the options for the SRTP window size. Currently not adjustable.
· SRTP Window Size: Default = 64.
· Crypto Suites: Default = SRTP_AES_CM_128_SHA1_80 and SRTP_AES_CM_128_SHA1_32.
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