Chapter 10 Ip Single Line Telephone; Section 1 Introduction - NEC Sl2100 Networking Manual

IP Single Line Telephone
S 1 INTRODUCTION
ECTION
Session Initiation Protocol (SIP) Station feature provides Voice over Internet Protocol (VoIP) for IP
stations. This feature is defined by the Internet Engineering Task Force (IETF) RFC3261.
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, Voice over IP services
are available from an SIP service provider.
With the VoIPDB up to 128 TDM talk paths are supported. 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
VoIPDB DSP Resource. This is a required component of SIP implementation in the NEC SL2100. The
NEC SL2100 VoIPDB contains a regular TCP/RTP/IP stack that can handle real-time media and
supports industry standard SIP (RFC3261) communication on the WAN side.
For this feature, the VoIPDB is installed and assigned. The VoIPDB supports IP signaling for up to 128
(SIP Trunks and/or SIP Stations) and reduces the maximum capacity of system stations and/or Trunks
in accordance with the number of registered SIP Stations.
The NEC SL2100 supports the following 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.729. Low bandwidth requirement used on most Wide Area Network links.
• G.711. μLaw – High bandwidth requirement usually used on Local Area Networks.
• G.722 This CODEC is useful in fixed network, Voice over IP applications, where the required
bandwidth is typically not prohibitive.
• G.726 is an ITU-T ADPCM speech-coded standard covering the transmission of voice at rates of
16-, 24-, 32-, and 40Kbps.
The SIP Station feature set supports the HOLD and TRF features based on RFC draft.
• Draft-IETF-sipping-service-examples-09.txt.
• Section 2.5 Draft-ietf-sipping-service-examples- (Transfer - Attended) 15.txt
• IETF RFC is defined as: Internet Engineering Task Force (RFC) Request for Comments.
• The SIP Station feature set supports the Message Waiting Indication (MWI) based on RFC3842.
• SIP INFO works independent from other DTMF methods such as RFC2833. This means SIP
Terminals should send DTMF information by a single method, otherwise the system will receive both
separately causing double digits.
• When PRG 15-05-49 is set to 2: Allowed while RTP is not available, SIP INFO will be received while
RTP is not established. In-band method such as RFC2833 will be used once voice path is
established.
• When PRG 15-05-49 is set to 1: Allowed any time, SIP INFO will be received whenever they arrive.
NAT Mode for SIP Phone
• NAT mode for SIP Phone which can not use SIP P2P mode and standard SIP video call feature
uses P2P mode cannot establish in same system.
• When PRG 10-33-05 NAT mode for SIP phone is set to 1 - Enable, P2P mode for SIP Phone
becomes always Off regardless PRG 15-05-50 setting.
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