NEC UNIVERGE SV9100 Manual page 5

Part 1: VoIP Reference Manual 5
Session Initiation Protocol (SIP)
SIP is a proposed standard, developed by the Internet Engineering Task Force (IETF) for setting up
sessions between one or more clients. It is currently the leading signalling protocol for Voice over IP,
gradually replacing H.323 in this role. SIP has been designed to be future proof and is very flexible.
H.323
H.323 is a recommendation from the International Telecommunication Union (ITU-T) that defines the
protocols to provide audio-visual communication sessions on any packet network. It is a part of the
H.32x series of protocols which also address communications over ISDN and PSTN. This
signalling method is closely related to ISDN and is inflexible and difficult to implement new features.
Proprietary
Most telephone system manufacturers have developed their own protocols to transmit their own
proprietary features between telephone systems. This method usually provides a very feature-rich
and integrated VoIP network, but is limited to connecting systems from the same manufacturer.
Usually, if one or more of the Telephone Systems (or IP Telephones) is supplied by another
manufacturer it is necessary to use a standards- based protocol (i.e. SIP or H.323)
Please refer to the Compatibility section below
1.4 CODECs?
The term CODEC (COder/DECoder) describes the technology of encoding and decoding a signal.
Within VoIP this specifically refers to the algorithm used to convert speech from the PBX to data for
transmission on an IP network.
The CODEC's supported by the SV9100 are: -
G.711
This is the ITU-T recommendation for coding of speech at 64kbps using PCM (pulse code
modulation). This CODEC is often described as uncompressed as it uses the same sampling rate as
the traditional telephony (TDM). G.711 has a MOS score of 4.4 but uses a large amount of bandwidth
for transmission. This CODEC is not commonly used due to the amount of bandwidth required,
although it can be acceptable in LAN environment (i.e. System IP Phones connected over a
100Mbps LAN).
G.722
This ITU-T standard codec provides 7 kHz wideband audio at data rates from 48 to 64 kbps. This is
useful in a fixed network Voice over IP applications, where the required bandwidth is typically not
prohibitive, and offers a significant improvement in speech quality over older narrowband codec's
such as G.711, without an excessive increase in implementation complexity.
G.726
This ITU-T standard codec provides transmission of voice at rates of 16, 24, 32, and 40 kbps. The
most commonly used mode is 32 kbps, which doubles the usable network capacity by using half the
rate of G.711. It is primarily used on international trunks in the PSTN and also in the DECT wireless
standard.
The SV9100 uses the 32 kbps version.
iLBC
The iLBC codec is an algorithm that compresses each basic frame (20 ms or 30 ms) of 8000 Hz, 16-
bit sampled input speech, into output frames with rate of 400 bits for 30 ms basic frame size and 304
bits for 20 ms basic frame size. This codec is suitable for real time communications such as,