Ip Single Line Telephone (Sip) - NEC UNIVERGE SV9100 Features And Specifications

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IP Single Line Telephone (SIP)

SIP (Session Initiation Protocol) is used for Voice over Internet Protocol. It is defined by the IETF

(Internet Engineering Task Force) RFC3261. Other RFC designations, such as RFC3842, refer to a

later implementation of SIP and may be supported by the UNIVERGE SV9100. Commonly called SIP

Station, this feature is used for IP Stations using SIP.

SIP analyzes requests from clients and retrieves responses from servers, then sets call parameters at

either end of the communication, handles call transfer, and terminates. Typically, such features,

including but not limited to Voice over IP services, are available from an SIP service provider.

Depending on licensing, each GPZ-IPLE application can support a maximum of 256 TDM Talk paths.

This total may be shared among SIP Stations or SIP Trunks. Registered SIP Stations and/or SIP Trunks

require a one-to-one relation with the GPZ-IPLE DSP Resource. This is a required component of SIP

implementation in the SV9100.

The UNIVERGE SV9100 GCD-CP10 contains a regular TCP/RTP/IP stack that can handle real-time

media, support industry standard SIP (RFC 3261) communication on the WAN side, and interface with

the GPZ-IPLE.

SIP IP Stations use the GPZ-IPLE. The GPZ-IPLE controls and interprets RTP messaging from the SIP

IP Phone to the UNIVERGE SV9100 GCD-CP10.

The GPZ-IPLE supports only those Codecs that are considered to provide toll-quality equivalent speech

path. The following voice compression methods are supported for the IP Station SIP feature:

G.711 -Law – Highest Bandwidth

G.722 – Wideband

G.729 – Mid-Range Bandwidth

For the minimum bandwidth requirements for each voice call refer to Minimum Bandwidth

Requirements. This includes all the overhead of VoIP communication, including signaling).