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NEC Univerge SV8100 - IP Single Line Telephone (SIP)

NEC Univerge SV8100
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UNIVERGE SV8100 Issue 1.0
SV8100 Features and Specifications Manual 2 - 617
IP Single Line Telephone (SIP)
Description
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) Session Initiation Protocol.
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 Station or SIP Trunks. Registered SIP Stations and/or SIP Trunks require a
one-to-one relation with the PZ-( )ILPA DSP Resource. This is a required component of SIP
implementation in the SV8100.
The UNIVERGE SV8100
CD-CP00-AU 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 utilize the PZ-( )ILPA. The IPLA controls and interprets RTP messaging from the SIP IP
Phone to the UNIVERGE SV8100 CD-CP00-AU.
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:
G.711 uLaw – 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).
Codec
Transmit
Data Rate
Receive
Data Rate
Time
Between
Packets
Packetization
Delay
Default Jitter
Buffer Delay
Theoretical
Maximum
MOS
G711u Law 90 Kbps 90 Kbps 20 ms
1.5 ms 2 datagrams
(40 ms)
4.4
G729 34 Kbps 34 Kbps 20 ms
15.0 ms 2 datagrams
(40 ms)
4.07
When an IP Soft Phone is connected, set Time Between Packets to 100 ms.

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