Ip Single Line Telephone (Sip) - NEC Univerge SV8100 Features And Specifications Manual

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UNIVERGE SV8100

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 RFC 3842, refer to a

later implementation of SIP and may be supported by the UNIVERGE SV8100. 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.

Each PZ-32IPLA, PZ-64IPLA, or PZ-128IPLA application can support up to 16 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 PZ-( )IPLA DSP Resource. This is a required component of SIP

implementation in the SV8100.

The UNIVERGE SV8100 CD-CP00-US 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 PZ-( )IPLA.

SIP IP Stations use the PZ-( )IPLA. The IPLA controls and interprets RTP messaging from the SIP IP

Phone to the UNIVERGE SV8100 CD-CP00-US.

The IPLA 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:

-Law – Highest Bandwidth

G.729 – Mid-Range Bandwidth

The minimum bandwidth requirements for each voice call is listed in the following table. This includes all

the overhead of VoIP communication, including signaling).

When an IP Soft Phone is connected, set Time Between Packets to 100ms.

IP Single Line Telephone (SIP)

Packetization

Default Jitter

Buffer Delay