HDX•C, SLICE 2100, SLICE IP TRANSip TRANSip Optional
The following standards are supported by the HDX•C and SLICE 2100 running V3.0 or higher software, as well as SLICE IP and SLICE IP Micro running V4.0 or higher software. Asterisks denote features only supported in software V4.0 or higher.
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IP addressing |
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IPv6 is described by many separate RFCs. |
IP addressing |
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Domain Name System |
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DHCP (RFC1533, RFC1534, RFC2131, RFC2132) |
Dynamic Host Control Prot |
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NTP (RFC1305) |
Network Time Protocol |
The following ITU specifications are optional (when purchased) in HDX•C and SLICE 2100 equipped with TRANSip:
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ITU T.38 |
Fax over IP (FoIP) 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. |
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ITU V.150.1 |
Modem over IP (MoIP) Because VoIP encoding methods are optimized specifically for voice, they do not transport modem tone. 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. |
The following RFCs and codecs are supported by all HDX•C, SLICE 2100, SLICE IP and SLICE IP Micro models that are equipped with TRANSip:
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Main RTP Spec |
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Additional RTP spec |
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Comfort Noise |
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ITU-T G.711 A & Mu plus appendix I and appendix II |
Pulse Code Modulation (PCM) of Voice Frequencies |
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ITU-T G.723.1 H&L |
RFC 3951, Internet Low Bit Rate Codec (iLBC) |
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ITU-T G.726 |
16, 24, 32, 40 kbps Adaptive Differential Pulse Code Modulation (ADPCM) |
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ITU-T G.729 A & B |
Coding of Speech at 8 kbps using Conjugate-Structure Algebraic-Code-Excited Linear Prediction (CS-ACELP) |
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IETF RFC4040 |
RTP Payload Format for 64 kbps Transparent Call |
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IETF Draft - SIPPING 19* |
Session Initiation Protocol Service Examples, draft-ietf-sipping-service-examples-15 |
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RFC2327 |
SDP: Session Description Protocol |
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RFC2543 |
Session Initiation Protocol (SIP) |
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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. |
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RFC2976 |
SIP INFO Method |
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RFC3261 |
Session Initiation Protocol (SIP) |
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RFC3265 |
SIP Specific Event Notification |
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RFC3326 |
Reason Header Field for SIP |
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RFC3420* |
Internet Media Type message/sipfrag |
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RFC3428 |
SIP Extension for Instant Messaging |
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RFC3515* |
SIP Refer Method |
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RFC3581 |
An extension to the SIP for Symmetric Response Routing |
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RFC3891 |
SIP “Replaces” Header |
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RFC3892* |
SIP Referred-By Mechanism |
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RFC4028 |
Session Timers in SIP |
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RFC4235* |
An INVITE-Initiated Dialog Event Package for SIP |