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Chapter 6 <br />

RFC<br />

6.4.2.3.3 Ports<br />

RFC Title<br />

3264 An offer/answer model with SDP<br />

3311 Session Initiation Protocol (SIP) UPDATE Method<br />

3323 Privacy mechanism for SIP<br />

3324 Short term req for network asserted id<br />

3325 Private extensions to SIP for asserted id<br />

3326 The Reason Header for the Session Initiation Protocol Protocol (SIP)<br />

3398 ISDN user part to SIP mapping (only QSIG)<br />

3515 SIP Refer method (only for Private SIP Trunking, see Private SIP Trunking<br />

- Services - Transfer )<br />

3550 Common Gateway Interface for SIP<br />

3551 IP Payload Compression Using ITU-T V.44 Packet Method<br />

3555 Mime type registration of RTP payload format<br />

3581 Symetric response routing<br />

3725 3rd party call control (1st scenario only) (except Flow IV)<br />

3840 User Agent Capabilities in Session Initiation Protocol (SIP)<br />

3891 Replaces header (only for Private SIP Trunking, see Private SIP Trunking<br />

- Services - Transfer )<br />

3892 Referred-By mechanism (only for Private SIP Trunking, see Private SIP<br />

Trunking - Services - Transfer )<br />

3966 Tel URI for Telephone Numbers<br />

4028 Session timer<br />

4244 An Extension to the Session Initiation Protocol (SIP) for Request History<br />

Information<br />

4497 Interworking between SIP and QSIG<br />

4566 SDP<br />

4733 RTP payload for DTMF digits<br />

4916 Connected Identity in the SIP ( not compliant with section 4.5)<br />

5009 Private Header (P-Header) Extension to the Session Initiation Protocol<br />

(SIP) for Authorization of Early Media (partially supported)<br />

5626 Managing Client-Initiated Connections in the Session Initiation Protocol<br />

(SIP) (partially supported)<br />

T.38 Annex D Group 3 fax over IP<br />

UDP is used as transport protocol to carry signalling (SIP), and media (RTP, Fax T.38)<br />

protocols.<br />

Note 1:<br />

The Alcatel-Lucent OmniPCX Office Communication Server supports SIP over UDP protocol only.<br />

However, as requested by RFC 3261 section 18.1.1, the PCX switches to TCP if a SIP request message<br />

to be transmitted is within 200 bytes of the path MTU (Maximum Transmission Unit), or if it is larger than<br />

6-42

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