IP Single Line Terminal (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 RFC3842, refer to a
later implementation of SIP and may be supported by the SL2100. Commonly called SIP Station, this
feature is used for IP Stations using SIP.
Maximum capacity of 112 SIP stations are supported.
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.
The VoIPDB application can support up to 112 TDM Talk paths. This total may be shared among SIP
Stations or SIP Trunks. This is a required component of SIP implementation in the SL2100.
The SL2100 CPU 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 VoIPDB.
SIP IP Stations use the VoIPDB. The VoIPDB controls and interprets RTP messaging from the SIP IP
Terminal to the SL2100 CPU.
The VoIPDB 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 A-law - Highest Bandwidth
• G.722 - Wideband
• G.729 - Mid-Range Bandwidth
For the minimum bandwidth requirements for each voice call refer to
. This includes all the overhead of VoIP communication, including
signaling).
Table 1-37 Minimum Bandwidth Requirements
Codec
Transmit Da-
ta Rate
Receive Data
Rate
Time Be-
tween Pack-
ets
Packetization
Delay
Default Jitter
Buffer Delay
Theoretical
Maximum
MOS
G.711 A-law
90 Kbps
90 Kbps
20ms
*1
1.5 ms
2 datagrams
(40ms)
4.4
G.722
64 Kbps
64 Kbps
20ms
*1
1.0 ms
2 datagrams
(40ms)
4.5
G.729
34 Kbps
34 Kbps
20ms
*1
15.0 ms
2 datagrams
(40ms)
4.07
*1. When an IP Soft Phone is connected, set Time Between Packets to 100ms.
• The VoIPDB is an end-point on the IP network from the network administration perspective.
• The CPU uses SIP Protocol to provide telephony services between remote stations through the IP
Network. This is an IETF/ITU standards-based protocol.
• Speech-connection audio quality depends greatly on the available bandwidth between the stations
in the data network. Because Internet is an uncontrolled data network compared to an Intranet,
using this application in Intranet WAN environment with known (or controlled and assured) Quality of
Service (QoS) is highly recommended.
ISSUE 1.0
SL2100
Features and Specifications Manual
1-493
I
Summary of Contents for UNIVERGE SL2100
Page 1: ...Features and Specifications Manual GVT 010794 401 00 AU ISSUE 1 0 May 2017 ...
Page 14: ...MEMO SL2100 ISSUE 1 0 R 4 Regulatory ...
Page 313: ...Operation None ISSUE 1 0 SL2100 Features and Specifications Manual 1 299 D ...
Page 412: ...Operation None SL2100 ISSUE 1 0 1 398 Howler Tone Service H ...
Page 572: ...LCR Dial LCR Dial Editing SL2100 ISSUE 1 0 1 558 LCR Least Cost Routing L ...
Page 573: ...LCR Cost Center Code ISSUE 1 0 SL2100 Features and Specifications Manual 1 559 L ...
Page 846: ...2 Press Hold key and talk with the party SL2100 ISSUE 1 0 1 832 Tone Override T ...
Page 878: ...Operation None SL2100 ISSUE 1 0 1 864 Universal Slots U ...
Page 946: ...MEMO SL2100 ISSUE 1 0 1 932 Warning Tone for Long Conversation W ...
Page 976: ...MEMO SL2100 ISSUE 1 0 3 6 Features Availability by Software Revision ...
Page 977: ...MEMO ISSUE 1 0 SL2100 Features and Specifications Manual 3 7 ...
Page 978: ...Features and Specifications Manual NEC Corporation ISSUE 1 0 ...