Overview Chapter F: Session Initiation Protocol
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release, thus users should not place 911 calls from any device that they believe is a SIP
extension.
F.1.1 Supported RFCs
ShoreTel 8 supports the following RFCs:
•1889 - Transport Protocol for RTP Applications
•2806 - URL for Telephone Calls
•2327 – Session Description Protocol (SDP)
•2396 - URI (Uniform Resource Identifiers)
•2833 - DTMF
•2976 - SIP Info
•3261 - SIP (Session Initiation Protocol)
•3361 - DHCP (for IPv4)
•3515 - SIP Refer Method
•3891 - SIP Replaces Header
•3892 - SIP Referred-by Mechanism
•3966 - URI for Telephone Calls
F.1.2 General SIP Comments
F.1.2.1 Conferencing
•Ports for MakeMe conferences must be available on the initiating side of a 3-way
conference call involving a SIP end-point.
•MakeMe conference ports are needed even for 3-way conference. Note that
configuration of any MakeMe conferencing support in Director requires a
minimum of 3 available conference ports.
•An individual SIP trunk must be provisioned for each call to the SIP device
(including conference-in or transferred calls). Thus, static SIP trunks must be
provisioned with additional trunks in line with the highest anticipated number of
such calls. Similarly, dynamic SIP trunks also require that additional individual
dynamic SIP trunks are provisioned to handle calls that are placed on hold or for
conference-in calls.
F. 1 . 2 . 2 D T M F
•With G.729, ShoreTel both sends and receives DTMF out of band per RFC 2833.
•With G.711, ShoreTel will only receive DTMF per RFC 2833. Not all ShoreTel
endpoints will send DTMF with G.711. For example, switches may not but
ShoreTel IP phones will.
•ShoreTel IP phones support in-band G711 DTMF signaling. However, out-of-band
DTMF is required for a SIP device to send DTMF to ShoreTel's voicemail or auto
attendant. SIP INFO or DTMF per RFC 2833 can be used.
•ShoreTel can be configured to use the SIP INFO function for DTMF signaling for
environments where out-of-band DTMF is needed but in which RFC 2833 is not
applicable. Note that SIP tie trunks must use SIP INFO and cannot use RFC 2833
DTMF Relay.
