125
Manual
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3 Way Conferencing
This feature enables the IAD endpoint to initiate 3 party conferencing. It is provided SIP and
can be achieved through three hook flash events. However for SIP, it can only be enabled if the
underlying DSP has audio mixing support. By default, this feature is not enabled in the system
file settings. To enable it, uncomment the “Set option_voice_dsp_mixing” string in the system
file.
SIP Features
SIP Features supported
SIP is implemented according to RFC3261. The list below provides a summary of the main
features:
• Act as a UAC (User Agent Client) and UAS (User Agent Server).
• Bi-directional authentication between UAC and UAS.
• Authentication between UAC and Proxy server.
• Simple call establishment and termination.
• Caller ID display (Type 1 and Type 2). The phone number and Caller name is configured in the
SIP node.
• Silence Suppression, Echo Cancellation and Digit Map are configured in the SIP node.
• IPv4 Addressing only.
• SIP DNS (as defined by RFC3263).
• SIP information from DHCP
• SIP QoS support
SIP Features not supported
As SIP is an emerging standard, it is also worth noting the features which are not supported in
the current IAD implementation of SIP.
• Dynamic mapping of users to endpoints.
• Support for SIP-B (SIP for Business phones).
• Act as a Proxy Registration Server or Redirection server.
In particular, there are a number of RFCS which outline extensions to SIP which are not
supported in the current IAD implementation of SIP:
• SIP INFO method (as defined by RFC 2976).
• Reliability of provisional responses (as defined by RFC 3262).
• Specific event notification (as defined by RFC 3265).
• The SIP Update method (as defined by RFC 3311).
• The ‘Reason’ extension header field (as defined by RFC 3326).
• The ‘Path’ extension header field (as defined by RFC 3327).
• The Refer method to enable call forwarding and call transfer (as described in RFC 3515) to
enable call forwarding and call transfer.
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