What are RFCs? HDX, SLICE 2100, SLICE IP TRANSip TRANSip Optional

What are RFCs?

Specifications in the IP World are called RFCs. Technically, this stands for “Requests for Comment”. RFCs go through a formal process of review, leading up to publication as a standard by the Internet Engineering Task Force (IETF). Interestingly, there are no “revisions” to RFCs. If a new RFC supercedes an old one, it is simply given an new RFC number.

There is no “requirement” that all equipment conform to all RFCs. Designers, manufacturers, and publishers of IP equipment and software pick and choose the RFCs they believe are relevant to their product and their customers.

The elements forming TRANSip conform to relevant RFCs. Some of these RFCs are supported by all HDX and SLICE 2100 units running V3.1 software or higher. Other RFCs are only supported if the HDX or SLICE 2100 is equipped with TRANSip.

Other organizations which publish specifications or standards which are used by the HDX, SLICE 2100, SLICE IP, SLICE IP Micro and TRANSip include the International Telecommunications Union (ITU), European Telecommunications Standards Institute (ETSI) and the American National Standards Institute (ANSI).

HDX, SLICE 2100 & SLICE IP

The following standards are supported by the HDX and SLICE 2100 running V3.1 or higher software (whether or not they are equipped with TRANSip), as well as SLICE IP running V4.0
or higher software:

General Specifications

IPv4

IP addressing

IPv6 is described by many separate RFCs.
(RFC 2460- general, RFC4291- addressing)

IP addressing

DNS (RFC1034, RFC1035)

Domain Name System

DHCP (RFC1533, RFC1534, RFC2131, RFC2132)

Dynamic Host Control Prot

NTP (RFC1305)

Network Time Protocol

TRANSip

The following RFCs and codecs are supported by all HDX, SLICE 2100,  SLICE IP and SLICE IP Micro models that are equipped with TRANSip:

RTP-Real Time Transport Protocol-Audio Streaming

RFC3550

Main RTP Spec

RFC3551

Additional RTP spec

RFC3389

Comfort Noise

List of Supported Codecs

ITU-T G.711 A & Mu plus appendix I and appendix II

Pulse Code Modulation (PCM) of Voice Frequencies

ITU-T G.723.1 H&L

RFC 3951, Internet Low Bit Rate Codec (iLBC)

ITU-T G.726

16, 24, 32, 40 kbps Adaptive Differential Pulse Code Modulation (ADPCM)

ITU-T G.729 A & B

Coding of Speech at 8 kbps using Conjugate-Structure Algebraic-Code-Excited Linear Prediction (CS-ACELP)

IETF RFC4040

RTP Payload Format for 64 kbps Transparent Call

SIP Call Management

IETF Draft - SIPPING 19

Session Initiation Protocol Service Examples, draft-ietf-sipping-service-examples-15

RFC2327

SDP: Session Description Protocol

RFC2543

Session Initiation Protocol (SIP)

RFC2833

RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals. Because VoIP encoding methods are optimized specifically for voice, they do not transport DTMF digits or telephony call progress tones. Therefore, these tones must be detected and/or regenerated at both ends of the connection.

RFC2976

SIP INFO Method

RFC3261

Session Initiation Protocol (SIP)

RFC3265

SIP Specific Event Notification

RFC3326

Reason Header Field for SIP

RFC3420

Internet Media Type message/sipfrag

RFC3428

SIP Extension for Instant Messaging

RFC3515

SIP Refer Method

RFC3581

An extension to the SIP for Symmetric Response Routing

RFC3891

SIP “Replaces” Header

RFC3892

SIP Referred-By Mechanism

RFC4028

Session Timers in SIP

RFC4235

An INVITE-Initiated Dialog Event Package for SIP

TRANSip Optional

The following ITU specs are supported as an option in HDX and SLICE 2100 equipped with TRANSip and when the option is purchased:

ITU T.38

Fax over IP (FoIP)

Because VoIP encoding methods are optimized specifically for voice, they do not transport modem tones designed to transport faxes.

For a traditional fax to communicate with a fax connected to the data port of a VoIP phone, or a fax designed specifically to behave as a VoIP phone, it is necessary for a Media Services Circuit with a TRANSip FoIP engine to convert between the two protocols.

ITU V.150.1

Modem over IP (MoIP)

Because VoIP encoding methods are optimized specifically for voice, they do not transport modem tones.

It might seem strange that there is even a requirement for VoIP to transport modem tones, considering that the use of VoIP presumes the availability of a direct IP connection which can transport data. However, existing modems need to be supported as they will be functional for the foreseeable feature.

 
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